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hang up detects a ring


andrewgroup
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my turn on odd issue. Cisco 2400 IAD with pots lines into latest 410 all firmware releases, when the unit hangs up it immediately detects a false ring andt then times out since not real call was made. I suspect the IAD is doing this. The provider has 200 of these deployed with same FXS configs. They made some changes in voltage settings but same, tried 3 different 410s. Guess I'll get my scope out.

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Seems this Cisco IAD 2400 is unique. When you hang up a call, with a linemans test set too, the Cisco is generating some tones, rather than a clean hangup, and must be causing the CS410 to answer the line again.

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Seems this Cisco IAD 2400 is unique. When you hang up a call, with a linemans test set too, the Cisco is generating some tones, rather than a clean hangup, and must be causing the CS410 to answer the line again.

 

Hi All,

 

Lines hangging fxo ports, I just had caller id turned on on my 2 lines, the 2 days we went with no caller ID the world was great, now I am having some new weirdness, I have not caught the time when the line hangs but it does not happen all the time, It is just that I have a port in use and no calls are being made,,I have noticed now that calls comming from some different area codes will cause both lines to ring..and I can answer the call and the other line will keep ringing but no one there, also, the opposite will happen, and it connects the second line..anyways when I get a chance I will log it and study them to see what the pstn port is doing..

 

:P

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Hi All,

 

Lines hangging fxo ports, I just had caller id turned on on my 2 lines, the 2 days we went with no caller ID the world was great, now I am having some new weirdness, I have not caught the time when the line hangs but it does not happen all the time, It is just that I have a port in use and no calls are being made,,I have noticed now that calls comming from some different area codes will cause both lines to ring..and I can answer the call and the other line will keep ringing but no one there, also, the opposite will happen, and it connects the second line..anyways when I get a chance I will log it and study them to see what the pstn port is doing..

 

:P

 

I spent a little time with the box tonight, here is a log of me hanging up a call right as a second call comes in, just logging the pstn port, I can see the call ringing the second PSTN port, but the phones never ring...I get the caller id info, but no call...the first call was disconnected from the other end by myself, I had called myself with my cell phone..

 

5] 2008/03/25 21:02:14: PSTN: Tone Detection set to 64

[3] 2008/03/25 21:02:47: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:02:51: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/03/25 21:02:51: PSTN: Received on 1 Caller-ID 3256900821

[5] 2008/03/25 21:02:51: PSTN: Received on 1 Name JACKS DEBORAH

[5] 2008/03/25 21:02:51: PSTN: Received LoopInterrupt (remote hung up) signal on channel 0

[3] 2008/03/25 21:02:51: PSTN: Channel 0: Hangup

[5] 2008/03/25 21:02:51: PSTN: Channel 0 goes onhook

[5] 2008/03/25 21:02:51: PSTN: enable_callerid 0

[3] 2008/03/25 21:02:51: PSTN: Channel 0 going to GO_ONHOOK

[5] 2008/03/25 21:02:51: PSTN: Response code: 200

[3] 2008/03/25 21:02:52: PSTN: Channel 0 going to IDLE

[3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:02:54: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:02:54: PSTN: Ringing, but last invite = 1

[3] 2008/03/25 21:02:58: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:03:00: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:03:00: PSTN: Ringing, but last invite = 1

[3] 2008/03/25 21:03:04: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:03:06: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:03:06: PSTN: Ringing, but last invite = 1

[3] 2008/03/25 21:03:10: PSTN: Channel 1 going to NO_RING

[5] 2008/03/25 21:03:11: PSTN: Tone 16 detected on 65

[5] 2008/03/25 21:03:14: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:14: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:15: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:16: PSTN: Timeout without ring on 1, going to idle

[3] 2008/03/25 21:03:16: PSTN: Channel 1 going to IDLE

[5] 2008/03/25 21:03:16: PSTN: Response code: 481

[5] 2008/03/25 21:03:19: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:20: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:24: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:25: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:29: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:30: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:34: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:35: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:39: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:40: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:44: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:45: PSTN: Tone 255 detected on 65

[5] 2008/03/25 21:03:49: PSTN: Tone 34 detected on 65

[5] 2008/03/25 21:03:50: PSTN: Tone 255 detected on 65

 

 

 

 

 

 

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[3] 2008/03/25 21:02:47: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:02:51: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/03/25 21:02:51: PSTN: Received on 1 Caller-ID 3256900821

[5] 2008/03/25 21:02:51: PSTN: Received on 1 Name JACKS DEBORAH

[3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING

 

What surprises me here is that channel 1 obviously receives the caller-ID while in the "RING" state... Is that possible? I thought the caller-ID is sent in the pause between the first and the second ring?

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Just yesterday we had a client 410 running 2448 appear to have 2 lines hung and the SLA lines on all phones were lit. We call the main line in the 4 line hunt group and were able to reach 2 of the 4 lines, and call three simply rang. This would indicate the port wasn't really busied by the PBX, but it had it registered as in use. Of course this was a production system 150 miles away and a reset was done.

 

Is logging the PSTN events likely to capture the tale-tell signs of what may cause this, if it occurs again?

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Well the PSTN gateway is really a seperate subsystem. Just like you connect an external PSTN gateway. The communication runs only via SIP.

 

If a CO-line on the PBX gets stuck, then that means that the gateway did not send a BYE. So if you do logging, make sure that you are watching the SIP traffic on the IP address 127.0.0.1.

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Well the PSTN gateway is really a seperate subsystem. Just like you connect an external PSTN gateway. The communication runs only via SIP.

 

If a CO-line on the PBX gets stuck, then that means that the gateway did not send a BYE. So if you do logging, make sure that you are watching the SIP traffic on the IP address 127.0.0.1.

 

Seems then we have no way of determining the failure, a MAX call length along with max call recording might clear hung ports. (3 hours)

 

Watching the SIP traffic with what? All we have installed is a 410 plus 4 phones, connected to 4 pots lines and a POE switch.

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What surprises me here is that channel 1 obviously receives the caller-ID while in the "RING" state... Is that possible? I thought the caller-ID is sent in the pause between the first and the second ring?

 

Well you would think.. :blink: maybe the processor catches the caller ID right after the ring , you see the no ring after the caller id, (i actually get the phone number in the email trace, it sends a missed called and leaves it in the phone also) but this is what bothering me, is I can see the fxo port flashing, but the phones don't ring, you see the port flashing while this happening

 

[3] 2008/03/25 21:02:52: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:02:54: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:02:54: PSTN: Ringing, but last invite = 1

[3] 2008/03/25 21:02:58: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:03:00: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:03:00: PSTN: Ringing, but last invite = 1

[3] 2008/03/25 21:03:04: PSTN: Channel 1 going to NO_RING

[3] 2008/03/25 21:03:06: PSTN: Channel 1 going to RING

[5] 2008/03/25 21:03:06: PSTN: Ringing, but last invite = 1

 

I was just wondering what the "last invite =1" thing is? :huh:

 

Also where can I find some info on the response and tone codes? :P

thanks

 

russelln

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Well that would mean that the INVITE is already out.

 

There is a new version that has more logging in this area: http://www.pbxnsip.com/protect/update-2898.tgz. Maybe you have the chance to give that image a shot.

 

thanks, loaded new version, here is a log of one call causing 2 channels to ring, this is just logged from the pstn side, if you want me capture one with everything let me know..I will try to grab it..

 

without caller id it worked great....

 

 

[3] 2008/03/26 10:51:55: PSTN: Channel 0 going to RING

[5] 2008/03/26 10:51:59: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/03/26 10:51:59: PSTN: Received on 0: Caller-ID 3255131178

[5] 2008/03/26 10:51:59: PSTN: Received on 0: Name THE SOUND SHOP

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 53590

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 53591

[5] 2008/03/26 10:51:59: Identify trunk (IP address/port and domain match) 5

[5] 2008/03/26 10:51:59: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/03/26 10:51:59: PSTN: Received on 1: Caller-ID 3255131178

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 62300

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 62301

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 53548

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 53549

[5] 2008/03/26 10:51:59: PSTN: Received on 1: Name THE SOUND SHOP

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 56692

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 56693

[5] 2008/03/26 10:51:59: Identify trunk (IP address/port and domain match) 5

[5] 2008/03/26 10:51:59: PSTN: Response code: 100

[5] 2008/03/26 10:51:59: PSTN: Response code: 183

[5] 2008/03/26 10:51:59: PSTN: Response code: 100

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 63712

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 63713

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 51518

[7] 2008/03/26 10:51:59: UDP: Opening socket on port 51519

[5] 2008/03/26 10:51:59: PSTN: Response code: 183

[5] 2008/03/26 10:52:00: Last message repeated 3 times

[3] 2008/03/26 10:52:00: PSTN: Channel 0 going to NO_RING

[5] 2008/03/26 10:52:00: PSTN: Response code: 183

[5] 2008/03/26 10:52:02: Last message repeated 2 times

[3] 2008/03/26 10:52:02: PSTN: Channel 0 going to RING

[5] 2008/03/26 10:52:02: PSTN: Ringing, but last invite = 1

[5] 2008/03/26 10:52:02: PSTN: Response code: 183

[5] 2008/03/26 10:52:06: Last message repeated 3 times

[3] 2008/03/26 10:52:06: PSTN: Channel 0 going to NO_RING

[5] 2008/03/26 10:52:06: PSTN: Response code: 183

[5] 2008/03/26 10:52:07: PSTN: Response code: 200

[5] 2008/03/26 10:52:07: PSTN: RTP destination=100007f

[5] 2008/03/26 10:52:07: PSTN: RTP destination=53590

[5] 2008/03/26 10:52:07: PSTN: RTP OOB codec=101

[6] 2008/03/26 10:52:07: PSTN: Start call on 0

[5] 2008/03/26 10:52:07: PSTN: Channel 0 goes offhook

[3] 2008/03/26 10:52:07: PSTN: Channel 0 going to TALKING

[5] 2008/03/26 10:52:07: PSTN: Country Code set to 64

[5] 2008/03/26 10:52:07: PSTN: Tone Detection set to 0

[7] 2008/03/26 10:52:14: UDP: Opening socket on port 57756

[7] 2008/03/26 10:52:14: UDP: Opening socket on port 57757

[5] 2008/03/26 10:52:14: PSTN: Response code: 183

[5] 2008/03/26 10:52:24: Last message repeated 6 times

[5] 2008/03/26 10:52:24: PSTN: Received BYE message on channel 0

[3] 2008/03/26 10:52:24: PSTN: Channel 0: Hangup

[5] 2008/03/26 10:52:24: PSTN: Channel 0 goes onhook

[5] 2008/03/26 10:52:24: PSTN: enable_callerid 0

[3] 2008/03/26 10:52:24: PSTN: Channel 0 going to GO_ONHOOK

[3] 2008/03/26 10:52:25: PSTN: Channel 0 going to IDLE

[5] 2008/03/26 10:52:27: PSTN: Response code: 200

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  • 2 weeks later...
Hmmm. Can you turn SIP logging on for the address 127.0.0.1? It would be good to see the INVITE coming from the PSTN gateway.

 

If you can reproduce the problem, chances are good that we can solve the riddle...

 

OK, got some time this morning here is the first log, call comes in on second fxo port, system rings extension, answer extension, no one at other end, line keeps ringing...answer call on backup phone...I have on system..

 

[9] 2008/04/10 09:11:44: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-0j4n97biwih4;rport

From: "41" <sip:41@localhost>;tag=8d3szdg5q4

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14209 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:11:44: Packet authenticated by transport layer

[9] 2008/04/10 09:11:44: Resolve 234: tls 192.168.1.102 2358

[9] 2008/04/10 09:11:44: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-0j4n97biwih4;rport=2358

From: "41" <sip:41@localhost>;tag=8d3szdg5q4

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14209 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=182

Contact: <http://192.168.1.102:80>;expires=182

Contact: <https://192.168.1.102:443>;expires=182

Content-Length: 0

 

 

[3] 2008/04/10 09:11:46: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:11:49: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[9] 2008/04/10 09:11:49: PSTN: Caller-ID: Received unknown tag: 01 08 30 34 31 30 30 39 71 c5 12 a4 68 93 a6 4d 4d 9c 33 31 31 37 38 07 0f 54 48 45 20 53 4f 55 4e 44 20 53 c8 3d 41

[9] 2008/04/10 09:11:49: SIP Rx udp:127.0.0.1:5062:

INVITE sip:3256725804@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>

Call-ID: 3f9a62d7@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 INVITE

Content-Type: application/sdp

Content-Length: 137

 

v=0

o=root 0 0 IN IP4 1.1.1.2

s=-

c=IN IP4 1.1.1.2

t=0 0

m=audio 2062 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

 

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 53392

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 53393

[5] 2008/04/10 09:11:49: Identify trunk (IP address/port and domain match) 5

[9] 2008/04/10 09:11:49: Resolve 235: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: Resolve 235: a udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: Resolve 235: udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 2008/04/10 09:11:49: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[7] 2008/04/10 09:11:49: Set packet length to 20

[6] 2008/04/10 09:11:49: Sending RTP for 3f9a62d7@fxo#28741e4a25 to 1.1.1.2:2062

[5] 2008/04/10 09:11:49: PSTN: Response code: 100

[5] 2008/04/10 09:11:49: Trunk PSTN1 sends call to 72

[8] 2008/04/10 09:11:49: Play audio_moh/noise.wav

[7] 2008/04/10 09:11:49: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:11:49: Hunt group 72 called 2 registrations

[5] 2008/04/10 09:11:49: PSTN: Received on 1: Caller-ID 3255131178

[7] 2008/04/10 09:11:49: Set packet length to 20

[9] 2008/04/10 09:11:49: Resolve 236: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: Resolve 236: a udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: Resolve 236: udp 127.0.0.1 5062

[9] 2008/04/10 09:11:49: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 56164

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 56165

[9] 2008/04/10 09:11:49: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label

[9] 2008/04/10 09:11:49: Resolve 237: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:11:49: Resolve 237: a tls 192.168.1.101 2614

[9] 2008/04/10 09:11:49: Resolve 237: tls 192.168.1.101 2614

[9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>

Call-ID: f7b7fdc2@pbx

CSeq: 27585 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 1711529881 1711529881 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 56164 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7PwZcsJzAakf+GjMlE7/1g4IsOOEaMLJKYZNjrdR

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 53014

[7] 2008/04/10 09:11:49: UDP: Opening socket on port 53015

[9] 2008/04/10 09:11:49: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label

[9] 2008/04/10 09:11:49: Resolve 238: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:11:49: Resolve 238: a tls 192.168.1.102 2358

[9] 2008/04/10 09:11:49: Resolve 238: tls 192.168.1.102 2358

[9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.102:2358:

INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 INVITE

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 384

 

v=0

o=- 607823118 607823118 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 53014 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:IaCicJ94fqcoYSuRI/S6qUFJYJJwA9+xGO9mLqg5

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[5] 2008/04/10 09:11:49: PSTN: Received on 1: Name THE SOUND SHOP

[8] 2008/04/10 09:11:49: PSTN: Received Caller-ID on channel 1, but already sent INVITE

[5] 2008/04/10 09:11:49: PSTN: Response code: 183

[9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

Call-ID: f7b7fdc2@pbx

CSeq: 27585 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:11:49: Resolve 239: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:11:49: Resolve 239: a tls 192.168.1.101 2614

[9] 2008/04/10 09:11:49: Resolve 239: tls 192.168.1.101 2614

[9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.101:2614:

PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-51248bc93881184e8df761caa954c81c;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

Call-ID: f7b7fdc2@pbx

CSeq: 27586 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

RAck: 1 27585 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:11:49: Play audio_en/ringback.wav

[9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:11:49: Resolve 240: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:11:49: Resolve 240: a tls 192.168.1.102 2358

[9] 2008/04/10 09:11:49: Resolve 240: tls 192.168.1.102 2358

[9] 2008/04/10 09:11:49: SIP Tx tls:192.168.1.102:2358:

PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-44400d04a1fdd63afa2589dcd4f0f273;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13846 PRACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

RAck: 1 13845 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:11:49: Play audio_en/ringback.wav

[9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-51248bc93881184e8df761caa954c81c;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

Call-ID: f7b7fdc2@pbx

CSeq: 27586 PRACK

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:11:49: Call f7b7fdc2@pbx#986462258: Clear last request

[9] 2008/04/10 09:11:49: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-44400d04a1fdd63afa2589dcd4f0f273;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13846 PRACK

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:11:49: Call 2f9a5c8b@pbx#1946255428: Clear last request

[9] 2008/04/10 09:11:50: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:50: PSTN: Response code: 183

[3] 2008/04/10 09:11:51: PSTN: Channel 1 going to NO_RING

[9] 2008/04/10 09:11:51: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:51: PSTN: Response code: 183

[3] 2008/04/10 09:11:52: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:11:52: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:11:52: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-cc71df1ba31471f6fbbb0389c307fd35;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

Call-ID: f7b7fdc2@pbx

CSeq: 27585 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Type: application/sdp

Content-Length: 463

 

v=0

o=root 1435449537 1435449538 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 52728 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4L+puWgqA3EYj56N47G+I0A/8cIGl8ZkqvFD0jO2

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 52728

a=sendrecv

 

[7] 2008/04/10 09:11:52: Call f7b7fdc2@pbx#986462258: Clear last INVITE

[6] 2008/04/10 09:11:52: Sending RTP for f7b7fdc2@pbx#986462258 to 192.168.1.101:52728

[9] 2008/04/10 09:11:52: Resolve 241: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:11:52: Resolve 241: a tls 192.168.1.101 2614

[9] 2008/04/10 09:11:52: Resolve 241: tls 192.168.1.101 2614

[9] 2008/04/10 09:11:52: SIP Tx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c07f3f53965b11f12d9c1f4ac143483d;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

To: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

Call-ID: f7b7fdc2@pbx

CSeq: 27585 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[7] 2008/04/10 09:11:52: Determine pass-through mode after receiving response

[9] 2008/04/10 09:11:53: Resolve 242: tls 192.168.1.102 2358

[9] 2008/04/10 09:11:53: SIP Tx tls:192.168.1.102:2358:

CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 CANCEL

Max-Forwards: 70

Reason: SIP;cause=200;text="Call completed elsewhere"

Content-Length: 0

 

 

[9] 2008/04/10 09:11:53: Resolve 243: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:11:53: Resolve 243: a udp 127.0.0.1 5062

[9] 2008/04/10 09:11:53: Resolve 243: udp 127.0.0.1 5062

[9] 2008/04/10 09:11:53: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:53: PSTN: Response code: 200

[9] 2008/04/10 09:11:53: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 CANCEL

Content-Length: 0

 

 

[7] 2008/04/10 09:11:53: Call 2f9a5c8b@pbx#1946255428: Clear last request

[9] 2008/04/10 09:11:53: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:11:53: Call 2f9a5c8b@pbx#1946255428: Clear last INVITE

[9] 2008/04/10 09:11:53: Resolve 244: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:11:53: Resolve 244: a tls 192.168.1.102 2358

[9] 2008/04/10 09:11:53: Resolve 244: tls 192.168.1.102 2358

[9] 2008/04/10 09:11:53: SIP Tx tls:192.168.1.102:2358:

ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0b25baeffb1448798c8837752400d61f;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1946255428

To: <sip:3256725804@localhost;user=phone>;tag=wobtjxxe3t

Call-ID: 2f9a5c8b@pbx

CSeq: 13845 ACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[5] 2008/04/10 09:11:53: INVITE Response: Terminate 2f9a5c8b@pbx

[7] 2008/04/10 09:11:53: Other Ports: 2

[7] 2008/04/10 09:11:53: Call Port: 3f9a62d7@fxo#28741e4a25

[7] 2008/04/10 09:11:53: Call Port: f7b7fdc2@pbx#986462258

[9] 2008/04/10 09:11:53: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:53: PSTN: Response code: 200

[9] 2008/04/10 09:11:54: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:54: PSTN: Response code: 200

[9] 2008/04/10 09:11:56: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:11:56: PSTN: Response code: 200

[3] 2008/04/10 09:11:57: PSTN: Channel 1 going to NO_RING

[3] 2008/04/10 09:11:58: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:11:58: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:12:00: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:12:00: PSTN: Response code: 200

[9] 2008/04/10 09:12:01: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-qqrcaqbwme2v;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13652 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:12:01: Packet authenticated by transport layer

[9] 2008/04/10 09:12:01: Resolve 245: tls 192.168.1.101 2614

[9] 2008/04/10 09:12:01: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-qqrcaqbwme2v;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13652 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 179

Content-Length: 0

 

 

[3] 2008/04/10 09:12:03: PSTN: Channel 1 going to NO_RING

[9] 2008/04/10 09:12:03: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-1jx7405cbw2m;rport

From: "40" <sip:40@localhost>;tag=lrws3sy5w8

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14204 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:12:03: Packet authenticated by transport layer

[9] 2008/04/10 09:12:03: Resolve 246: tls 192.168.1.101 2614

[9] 2008/04/10 09:12:03: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-1jx7405cbw2m;rport=2614

From: "40" <sip:40@localhost>;tag=lrws3sy5w8

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14204 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=182

Contact: <http://192.168.1.101:80>;expires=182

Contact: <https://192.168.1.101:443>;expires=182

Content-Length: 0

 

 

[3] 2008/04/10 09:12:04: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:12:04: PSTN: Ringing, but last invite = 1

[3] 2008/04/10 09:12:05: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:12:07: SIP Rx tls:192.168.1.101:2614:

BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-ehgsa1xb4ej8;rport

From: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

Call-ID: f7b7fdc2@pbx

CSeq: 1 BYE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=17,Rx_Pkts=17,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=712,Tx_Pkts=712,Remote_Tx_Pkts=0

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:12:07: Resolve 247: tls 192.168.1.101 2614

[9] 2008/04/10 09:12:07: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-ehgsa1xb4ej8;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=5dp6j7fb67

To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=986462258

Call-ID: f7b7fdc2@pbx

CSeq: 1 BYE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

User-Agent: pbxnsip-PBX/3.0.0.2899

RTP-RxStat: Dur=18,Pkt=716,Oct=126016,Underun=1422

RTP-TxStat: Dur=14,Pkt=17,Oct=2992

Content-Length: 0

 

 

[7] 2008/04/10 09:12:07: Other Ports: 1

[7] 2008/04/10 09:12:07: Call Port: 3f9a62d7@fxo#28741e4a25

[9] 2008/04/10 09:12:07: Resolve 248: url sip:127.0.0.1:5062

[9] 2008/04/10 09:12:07: Resolve 248: udp 127.0.0.1 5062

[9] 2008/04/10 09:12:07: SIP Tx udp:127.0.0.1:5062:

BYE sip:127.0.0.1:5062 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-438247cad4e43aea9daa827aa236b1dd;rport

From: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

Call-ID: 3f9a62d7@fxo

CSeq: 23963 BYE

Max-Forwards: 70

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

RTP-RxStat: Dur=18,Pkt=0,Oct=0,Underun=6

RTP-TxStat: Dur=14,Pkt=882,Oct=151704

Content-Length: 0

 

 

[8] 2008/04/10 09:12:07: DNS: Add dns_a smtp.bizmail.mail.yahoo4.akadns.net 68.142.200.11 (ttl=140)

[9] 2008/04/10 09:12:07: SIP Rx udp:127.0.0.1:5062:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-438247cad4e43aea9daa827aa236b1dd;rport

From: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

To: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

Call-ID: 3f9a62d7@fxo

CSeq: 23963 BYE

Content-Length: 0

 

 

[7] 2008/04/10 09:12:07: Call 3f9a62d7@fxo#28741e4a25: Clear last request

[5] 2008/04/10 09:12:07: BYE Response: Terminate 3f9a62d7@fxo

[8] 2008/04/10 09:12:07: SMTP: Connect to 68.142.200.11:25

[8] 2008/04/10 09:12:07: SMTP: Received 220 smtp100.biz.mail.mud.yahoo.com ESMTP

 

[8] 2008/04/10 09:12:08: SMTP: Received 250-smtp100.biz.mail.mud.yahoo.com

250-AUTH LOGIN PLAIN XYMCOOKIE

250-PIPELINING

250 8BITMIME

 

[8] 2008/04/10 09:12:08: SMTP: Received 334 VXNlcm5hbWU6

 

[8] 2008/04/10 09:12:08: SMTP: Received 334 UGFzc3dvcmQ6

 

[3] 2008/04/10 09:12:08: PSTN: Channel 1 going to NO_RING

[8] 2008/04/10 09:12:08: SMTP: Received 235 ok, go ahead (#2.0.0)

 

[8] 2008/04/10 09:12:08: SMTP: Received 250 ok

 

[8] 2008/04/10 09:12:08: Last message repeated 2 times

[8] 2008/04/10 09:12:08: SMTP: Received 354 go ahead

 

[9] 2008/04/10 09:12:08: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>;tag=28741e4a25

Call-ID: 3f9a62d7@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1380716876 1380716876 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 53392 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:12:08: PSTN: Response code: 200

[8] 2008/04/10 09:12:09: SMTP: Received 250 ok 1207836728 qp 18839

 

[8] 2008/04/10 09:12:09: SMTP: Received 221 smtp100.biz.mail.mud.yahoo.com

 

[8] 2008/04/10 09:12:09: Sucessfully sent email to <russell@thesoundshop-llc.com>

[3] 2008/04/10 09:12:09: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:12:14: SIP Rx udp:127.0.0.1:5062:

CANCEL sip:3256723475@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1804289383

To: <sip:3256723475@localhost;user=phone>

Call-ID: 6931fac9@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 CANCEL

Content-Length: 0

 

 

[9] 2008/04/10 09:12:14: Resolve 249: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:12:14: Resolve 249: a udp 127.0.0.1 5062

[9] 2008/04/10 09:12:14: Resolve 249: udp 127.0.0.1 5062

[9] 2008/04/10 09:12:14: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1804289383

To: <sip:3256723475@localhost;user=phone>

Call-ID: 6931fac9@fxo

CSeq: 1 CANCEL

Content-Length: 0

 

 

[5] 2008/04/10 09:12:14: PSTN: Timeout without ring on 1, going to idle

[3] 2008/04/10 09:12:14: PSTN: Channel 1 going to IDLE

[5] 2008/04/10 09:12:14: PSTN: Response code: 481

[5] 2008/04/10 09:12:27: PSTN: Tone 34 detected on 1

[5] 2008/04/10 09:12:27: PSTN: Tone 255 detected on 1

[5] 2008/04/10 09:12:32: PSTN: Tone 34 detected on 1

[5] 2008/04/10 09:12:32: PSTN: Tone 255 detected on 1

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Allrighty, next call comes in on first FXO port, extension rings, answer extension,no one at other end, analog line keeps ringing, answer call on analog phone, but I get caller id on SNOM phone.

 

[5] 2008/04/10 09:13:05: PSTN: Received on 0: Name ABILEN IND SCH

[8] 2008/04/10 09:13:05: PSTN: Received Caller-ID on channel 0, but already sent INVITE

[5] 2008/04/10 09:13:05: PSTN: Response code: 100

[5] 2008/04/10 09:13:05: PSTN: Response code: 183

[9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29409 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:13:05: Resolve 254: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:13:05: Resolve 254: a tls 192.168.1.101 2614

[9] 2008/04/10 09:13:05: Resolve 254: tls 192.168.1.101 2614

[9] 2008/04/10 09:13:05: SIP Tx tls:192.168.1.101:2614:

PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2a83d7a5343447c2ba9c9f230e923d20;rport

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29410 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

RAck: 1 29409 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:13:05: Play audio_en/ringback.wav

[9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-5ded06c142d0cd523a5e87c446470647;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

Call-ID: 5f095176@pbx

CSeq: 30726 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:13:05: Resolve 255: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:13:05: Resolve 255: a tls 192.168.1.102 2358

[9] 2008/04/10 09:13:05: Resolve 255: tls 192.168.1.102 2358

[9] 2008/04/10 09:13:05: SIP Tx tls:192.168.1.102:2358:

PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-bf69f588cbba0daaa1a352ca0e0a68ba;rport

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

Call-ID: 5f095176@pbx

CSeq: 30727 PRACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

RAck: 1 30726 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:13:05: Play audio_en/ringback.wav

[9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2a83d7a5343447c2ba9c9f230e923d20;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29410 PRACK

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:13:05: Call a4449186@pbx#217507709: Clear last request

[9] 2008/04/10 09:13:05: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-bf69f588cbba0daaa1a352ca0e0a68ba;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

Call-ID: 5f095176@pbx

CSeq: 30727 PRACK

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:13:05: Call 5f095176@pbx#1195090579: Clear last request

[9] 2008/04/10 09:13:06: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:06: PSTN: Response code: 183

[3] 2008/04/10 09:13:06: PSTN: Channel 0 going to NO_RING

[9] 2008/04/10 09:13:07: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:07: PSTN: Response code: 183

[3] 2008/04/10 09:13:08: PSTN: Channel 0 going to RING

[5] 2008/04/10 09:13:08: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:13:08: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-5ded06c142d0cd523a5e87c446470647;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

Call-ID: 5f095176@pbx

CSeq: 30726 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

User-Agent: snom300/7.1.30

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Type: application/sdp

Content-Length: 463

 

v=0

o=root 1135634967 1135634968 IN IP4 192.168.1.102

s=call

c=IN IP4 192.168.1.102

t=0 0

m=audio 50046 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:4PlRLbprmuw+qpsW9CkbT2QqYXuCwIiXZ6f9Olzd

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.102 50046

a=sendrecv

 

[7] 2008/04/10 09:13:08: Call 5f095176@pbx#1195090579: Clear last INVITE

[6] 2008/04/10 09:13:08: Sending RTP for 5f095176@pbx#1195090579 to 192.168.1.102:50046

[9] 2008/04/10 09:13:08: Resolve 256: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:13:08: Resolve 256: a tls 192.168.1.102 2358

[9] 2008/04/10 09:13:08: Resolve 256: tls 192.168.1.102 2358

[9] 2008/04/10 09:13:08: SIP Tx tls:192.168.1.102:2358:

ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c9784da620ae5c4c196f13aa887103f9;rport

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

To: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

Call-ID: 5f095176@pbx

CSeq: 30726 ACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[7] 2008/04/10 09:13:08: Determine pass-through mode after receiving response

[9] 2008/04/10 09:13:08: Resolve 257: tls 192.168.1.101 2614

[9] 2008/04/10 09:13:08: SIP Tx tls:192.168.1.101:2614:

CANCEL sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>

Call-ID: a4449186@pbx

CSeq: 29409 CANCEL

Max-Forwards: 70

Reason: SIP;cause=200;text="Call completed elsewhere"

Content-Length: 0

 

 

[9] 2008/04/10 09:13:08: Resolve 258: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:13:08: Resolve 258: a udp 127.0.0.1 5062

[9] 2008/04/10 09:13:08: Resolve 258: udp 127.0.0.1 5062

[9] 2008/04/10 09:13:08: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:08: PSTN: Response code: 200

[9] 2008/04/10 09:13:09: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29409 CANCEL

Content-Length: 0

 

 

[7] 2008/04/10 09:13:09: Call a4449186@pbx#217507709: Clear last request

[9] 2008/04/10 09:13:09: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport=5061

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29409 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:13:09: Call a4449186@pbx#217507709: Clear last INVITE

[9] 2008/04/10 09:13:09: Resolve 259: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:13:09: Resolve 259: a tls 192.168.1.101 2614

[9] 2008/04/10 09:13:09: Resolve 259: tls 192.168.1.101 2614

[9] 2008/04/10 09:13:09: SIP Tx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-a6612d6557f6f20952121e2346f72850;rport

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=217507709

To: <sip:3256723475@localhost;user=phone>;tag=ooxg71bo4d

Call-ID: a4449186@pbx

CSeq: 29409 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[3] 2008/04/10 09:13:09: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[5] 2008/04/10 09:13:09: INVITE Response: Terminate a4449186@pbx

[7] 2008/04/10 09:13:09: Other Ports: 2

[7] 2008/04/10 09:13:09: Call Port: 5f095176@pbx#1195090579

[7] 2008/04/10 09:13:09: Call Port: 83a5d578@fxo#c8db8ae26e

[9] 2008/04/10 09:13:09: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:09: PSTN: Response code: 200

[9] 2008/04/10 09:13:10: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:10: PSTN: Response code: 200

[9] 2008/04/10 09:13:12: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:12: PSTN: Response code: 200

[3] 2008/04/10 09:13:13: PSTN: Channel 0 going to NO_RING

[3] 2008/04/10 09:13:14: PSTN: Channel 0 going to RING

[5] 2008/04/10 09:13:14: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:13:15: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-iu3rw72wtsv7;rport

From: "41" <sip:41@localhost>;tag=6tk3q6i8fp

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14210 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:13:15: Packet authenticated by transport layer

[9] 2008/04/10 09:13:15: Resolve 260: tls 192.168.1.102 2358

[9] 2008/04/10 09:13:15: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-iu3rw72wtsv7;rport=2358

From: "41" <sip:41@localhost>;tag=6tk3q6i8fp

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14210 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180

Contact: <http://192.168.1.102:80>;expires=180

Contact: <https://192.168.1.102:443>;expires=180

Content-Length: 0

 

 

[9] 2008/04/10 09:13:16: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:16: PSTN: Response code: 200

[9] 2008/04/10 09:13:17: SIP Rx tls:192.168.1.102:2358:

BYE sip:41@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-f2webjjtdm4t;rport

From: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

Call-ID: 5f095176@pbx

CSeq: 1 BYE

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=11,Rx_Pkts=11,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=430,Tx_Pkts=430,Remote_Tx_Pkts=0

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:13:17: Resolve 261: tls 192.168.1.102 2358

[9] 2008/04/10 09:13:17: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-f2webjjtdm4t;rport=2358

From: <sip:3256723475@localhost;user=phone>;tag=s2wpjj2zgg

To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1195090579

Call-ID: 5f095176@pbx

CSeq: 1 BYE

Contact: <sip:41@192.168.1.100:5061;transport=tls>

User-Agent: pbxnsip-PBX/3.0.0.2899

RTP-RxStat: Dur=12,Pkt=437,Oct=76912,Underun=884

RTP-TxStat: Dur=9,Pkt=11,Oct=1936

Content-Length: 0

 

 

[7] 2008/04/10 09:13:17: Other Ports: 1

[7] 2008/04/10 09:13:17: Call Port: 83a5d578@fxo#c8db8ae26e

[8] 2008/04/10 09:13:17: SMTP: Connect to 68.142.200.11:25

[9] 2008/04/10 09:13:17: Resolve 262: url sip:127.0.0.1:5062

[9] 2008/04/10 09:13:17: Resolve 262: udp 127.0.0.1 5062

[9] 2008/04/10 09:13:17: SIP Tx udp:127.0.0.1:5062:

BYE sip:127.0.0.1:5062 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-352849f79ced1ffa776bcbafa30d129a;rport

From: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

Call-ID: 83a5d578@fxo

CSeq: 24806 BYE

Max-Forwards: 70

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

RTP-RxStat: Dur=12,Pkt=0,Oct=0,Underun=10

RTP-TxStat: Dur=9,Pkt=615,Oct=105780

Content-Length: 0

 

 

[9] 2008/04/10 09:13:17: SIP Rx udp:127.0.0.1:5062:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-352849f79ced1ffa776bcbafa30d129a;rport

From: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

To: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

Call-ID: 83a5d578@fxo

CSeq: 24806 BYE

Content-Length: 0

 

 

[7] 2008/04/10 09:13:17: Call 83a5d578@fxo#c8db8ae26e: Clear last request

[5] 2008/04/10 09:13:17: BYE Response: Terminate 83a5d578@fxo

[8] 2008/04/10 09:13:18: SMTP: Received 220 smtp100.biz.mail.mud.yahoo.com ESMTP

 

[8] 2008/04/10 09:13:18: SMTP: Received 250-smtp100.biz.mail.mud.yahoo.com

250-AUTH LOGIN PLAIN XYMCOOKIE

250-PIPELINING

250 8BITMIME

 

[8] 2008/04/10 09:13:18: SMTP: Received 334 VXNlcm5hbWU6

 

[8] 2008/04/10 09:13:18: SMTP: Received 334 UGFzc3dvcmQ6

 

[8] 2008/04/10 09:13:18: SMTP: Received 235 ok, go ahead (#2.0.0)

 

[8] 2008/04/10 09:13:18: SMTP: Received 250 ok

 

[8] 2008/04/10 09:13:18: Last message repeated 2 times

[8] 2008/04/10 09:13:18: SMTP: Received 354 go ahead

 

[3] 2008/04/10 09:13:19: PSTN: Channel 0 going to NO_RING

[8] 2008/04/10 09:13:19: SMTP: Received 250 ok 1207836798 qp 19516

 

[8] 2008/04/10 09:13:19: SMTP: Received 221 smtp100.biz.mail.mud.yahoo.com

 

[8] 2008/04/10 09:13:19: Sucessfully sent email to <russell@thesoundshop-llc.com>

[3] 2008/04/10 09:13:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:13:24: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>;tag=c8db8ae26e

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 1076076106 1076076106 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 62278 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:13:24: PSTN: Response code: 200

[9] 2008/04/10 09:13:25: SIP Rx udp:127.0.0.1:5062:

CANCEL sip:3256725804@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>

Call-ID: 3f9a62d7@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 CANCEL

Content-Length: 0

 

 

[9] 2008/04/10 09:13:25: Resolve 263: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:13:25: Resolve 263: a udp 127.0.0.1 5062

[9] 2008/04/10 09:13:25: Resolve 263: udp 127.0.0.1 5062

[9] 2008/04/10 09:13:25: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=233665123

To: <sip:3256725804@localhost;user=phone>

Call-ID: 3f9a62d7@fxo

CSeq: 1 CANCEL

Content-Length: 0

 

 

[5] 2008/04/10 09:13:25: PSTN: Timeout without ring on 0, going to idle

[3] 2008/04/10 09:13:25: PSTN: Channel 0 going to IDLE

[5] 2008/04/10 09:13:25: PSTN: Response code: 481

[9] 2008/04/10 09:13:30: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-p04p5na21bhq;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13653 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:13:30: Packet authenticated by transport layer

[9] 2008/04/10 09:13:30: Resolve 264: tls 192.168.1.101 2614

[9] 2008/04/10 09:13:30: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-p04p5na21bhq;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13653 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 181

Content-Length: 0

 

 

[9] 2008/04/10 09:13:34: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9lc6y7ibd7v2;rport

From: "40" <sip:40@localhost>;tag=wlaz8y5hn3

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14205 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:13:34: Packet authenticated by transport layer

[9] 2008/04/10 09:13:34: Resolve 265: tls 192.168.1.101 2614

[9] 2008/04/10 09:13:34: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9lc6y7ibd7v2;rport=2614

From: "40" <sip:40@localhost>;tag=wlaz8y5hn3

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14205 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=181

Contact: <http://192.168.1.101:80>;expires=181

Contact: <https://192.168.1.101:443>;expires=181

Content-Length: 0

 

 

[3] 2008/04/10 09:14:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[8] 2008/04/10 09:14:19: DNS: dns_cname smtp.bizmail.yahoo.com expired

[8] 2008/04/10 09:14:27: DNS: dns_a smtp.bizmail.mail.yahoo4.akadns.net expired

[9] 2008/04/10 09:14:45: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-2jfbug3q53c1;rport

From: "41" <sip:41@localhost>;tag=dvtmvo1rcr

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14211 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:14:45: Packet authenticated by transport layer

[9] 2008/04/10 09:14:45: Resolve 266: tls 192.168.1.102 2358

[9] 2008/04/10 09:14:45: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-2jfbug3q53c1;rport=2358

From: "41" <sip:41@localhost>;tag=dvtmvo1rcr

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14211 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=179

Contact: <http://192.168.1.102:80>;expires=179

Contact: <https://192.168.1.102:443>;expires=179

Content-Length: 0

 

 

[9] 2008/04/10 09:15:00: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-6i9c74bg97n7;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13654 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:15:00: Packet authenticated by transport layer

[9] 2008/04/10 09:15:00: Resolve 267: tls 192.168.1.101 2614

[9] 2008/04/10 09:15:00: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-6i9c74bg97n7;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13654 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 178

Content-Length: 0

 

 

[9] 2008/04/10 09:15:05: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-96iva23npt5i;rport

From: "40" <sip:40@localhost>;tag=y1r4lu2urn

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14206 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:15:05: Packet authenticated by transport layer

[9] 2008/04/10 09:15:05: Resolve 268: tls 192.168.1.101 2614

[9] 2008/04/10 09:15:05: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-96iva23npt5i;rport=2614

From: "40" <sip:40@localhost>;tag=y1r4lu2urn

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14206 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=179

Contact: <http://192.168.1.101:80>;expires=179

Contact: <https://192.168.1.101:443>;expires=179

Content-Length: 0

 

 

[3] 2008/04/10 09:15:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:16:15: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-w66jgcmtd3gi;rport

From: "41" <sip:41@localhost>;tag=azhcqtj84e

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14212 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:16:15: Packet authenticated by transport layer

[9] 2008/04/10 09:16:15: Resolve 269: tls 192.168.1.102 2358

[9] 2008/04/10 09:16:15: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-w66jgcmtd3gi;rport=2358

From: "41" <sip:41@localhost>;tag=azhcqtj84e

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14212 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=181

Contact: <http://192.168.1.102:80>;expires=181

Contact: <https://192.168.1.102:443>;expires=181

Content-Length: 0

 

 

[3] 2008/04/10 09:16:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:16:29: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-26t04w8d2q9c;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13655 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:16:29: Packet authenticated by transport layer

[9] 2008/04/10 09:16:29: Resolve 270: tls 192.168.1.101 2614

[9] 2008/04/10 09:16:29: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-26t04w8d2q9c;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13655 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 179

Content-Length: 0

 

 

[9] 2008/04/10 09:16:34: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fy6nyde880e1;rport

From: "40" <sip:40@localhost>;tag=ocf1tlbn9m

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14207 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:16:34: Packet authenticated by transport layer

[9] 2008/04/10 09:16:34: Resolve 271: tls 192.168.1.101 2614

[9] 2008/04/10 09:16:34: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fy6nyde880e1;rport=2614

From: "40" <sip:40@localhost>;tag=ocf1tlbn9m

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14207 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=180

Contact: <http://192.168.1.101:80>;expires=180

Contact: <https://192.168.1.101:443>;expires=180

Content-Length: 0

 

 

[3] 2008/04/10 09:17:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:17:45: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-lz58s2vskjak;rport

From: "41" <sip:41@localhost>;tag=khr4qcyiyl

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14213 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:17:45: Packet authenticated by transport layer

[9] 2008/04/10 09:17:45: Resolve 272: tls 192.168.1.102 2358

[9] 2008/04/10 09:17:45: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-lz58s2vskjak;rport=2358

From: "41" <sip:41@localhost>;tag=khr4qcyiyl

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14213 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180

Contact: <http://192.168.1.102:80>;expires=180

Contact: <https://192.168.1.102:443>;expires=180

Content-Length: 0

 

 

[9] 2008/04/10 09:17:59: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-xv30apapngx6;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13656 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:17:59: Packet authenticated by transport layer

[9] 2008/04/10 09:17:59: Resolve 273: tls 192.168.1.101 2614

[9] 2008/04/10 09:17:59: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-xv30apapngx6;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13656 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 181

Content-Length: 0

 

 

[9] 2008/04/10 09:18:04: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9c7slmor056o;rport

From: "40" <sip:40@localhost>;tag=y7yyjg703x

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14208 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:18:04: Packet authenticated by transport layer

[9] 2008/04/10 09:18:04: Resolve 274: tls 192.168.1.101 2614

[9] 2008/04/10 09:18:04: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-9c7slmor056o;rport=2614

From: "40" <sip:40@localhost>;tag=y7yyjg703x

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14208 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=178

Contact: <http://192.168.1.101:80>;expires=178

Contact: <https://192.168.1.101:443>;expires=178

Content-Length: 0

 

 

[3] 2008/04/10 09:18:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:19:15: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-a3n6w796cqnc;rport

From: "41" <sip:41@localhost>;tag=0ziehgzq3g

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14214 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:19:15: Packet authenticated by transport layer

[9] 2008/04/10 09:19:15: Resolve 275: tls 192.168.1.102 2358

[9] 2008/04/10 09:19:15: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-a3n6w796cqnc;rport=2358

From: "41" <sip:41@localhost>;tag=0ziehgzq3g

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14214 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=180

Contact: <http://192.168.1.102:80>;expires=180

Contact: <https://192.168.1.102:443>;expires=180

Content-Length: 0

 

 

[3] 2008/04/10 09:19:19: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

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Ok so the next call comes in ok, and then after that call, I start calling line one, and the next 3 times I try the system it works fine, then I try line 2 again and the phones do not ring at all, but the weird thing I see in this log is the caller ID message..

 

I was calling from my cell phone but I see a call I received earlier in the caller ID..

 

[3] 2008/04/10 09:43:19: PSTN: Channel 1 going to RING

[9] 2008/04/10 09:43:20: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-tiae9flqybwy;rport

From: "41" <sip:41@localhost>;tag=6h8z2ywbws

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14230 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:43:20: Packet authenticated by transport layer

[9] 2008/04/10 09:43:20: Resolve 397: tls 192.168.1.102 2358

[9] 2008/04/10 09:43:20: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-tiae9flqybwy;rport=2358

From: "41" <sip:41@localhost>;tag=6h8z2ywbws

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14230 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=182

Contact: <http://192.168.1.102:80>;expires=182

Contact: <https://192.168.1.102:443>;expires=182

Content-Length: 0

 

 

[5] 2008/04/10 09:43:22: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/04/10 09:43:22: PSTN: Received on 1: Caller-ID 3255131178

[5] 2008/04/10 09:43:22: PSTN: Received on 1: Name THE SOUND SHOP

[8] 2008/04/10 09:43:22: PSTN: Received Caller-ID on channel 1, but already sent INVITE

[3] 2008/04/10 09:43:23: PSTN: Channel 1 going to NO_RING

[3] 2008/04/10 09:43:25: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:43:25: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:43:30: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-c4rumwbxh7tm;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13673 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:43:30: Packet authenticated by transport layer

[3] 2008/04/10 09:43:30: PSTN: Channel 1 going to NO_RING

[9] 2008/04/10 09:43:30: Resolve 398: tls 192.168.1.101 2614

[9] 2008/04/10 09:43:30: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-c4rumwbxh7tm;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13673 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 180

Content-Length: 0

 

 

[3] 2008/04/10 09:43:31: PSTN: Channel 1 going to RING

[5] 2008/04/10 09:43:31: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:43:32: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-pfh0d9vyfsmq;rport

From: "40" <sip:40@localhost>;tag=jpos857gfh

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14225 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:43:32: Packet authenticated by transport layer

[9] 2008/04/10 09:43:32: Resolve 399: tls 192.168.1.101 2614

[9] 2008/04/10 09:43:32: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-pfh0d9vyfsmq;rport=2614

From: "40" <sip:40@localhost>;tag=jpos857gfh

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14225 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=182

Contact: <http://192.168.1.101:80>;expires=182

Contact: <https://192.168.1.101:443>;expires=182

Content-Length: 0

 

 

[3] 2008/04/10 09:43:36: PSTN: Channel 1 going to NO_RING

[9] 2008/04/10 09:43:42: SIP Rx udp:127.0.0.1:5062:

CANCEL sip:3256723475@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>

Call-ID: 83a5d578@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 CANCEL

Content-Length: 0

 

 

[9] 2008/04/10 09:43:42: Resolve 400: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:43:42: Resolve 400: a udp 127.0.0.1 5062

[9] 2008/04/10 09:43:42: Resolve 400: udp 127.0.0.1 5062

[9] 2008/04/10 09:43:42: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 481 Call/Transaction Does Not Exist

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "ABILEN IND SCH " <sip:3256771444@localhost;user=phone>;tag=1059961393

To: <sip:3256723475@localhost;user=phone>

Call-ID: 83a5d578@fxo

CSeq: 1 CANCEL

Content-Length: 0

 

 

[5] 2008/04/10 09:43:42: PSTN: Timeout without ring on 1, going to idle

[3] 2008/04/10 09:43:42: PSTN: Channel 1 going to IDLE

[5] 2008/04/10 09:43:42: PSTN: Response code: 481

[3] 2008/04/10 09:43:56: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

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Ok now for the fun one, this one I can reproduce regulary, what I have to do is call line 2, the first call will not answer..as I have shown, then call line 2 again, that call will answer and seems to act ok, then call line 1 again, what happens is the extensions ring showing 2 lines at once, you pick up to answer the system answers on the second line, no one is there, you can put it on hold and hit the first line button that wil put the call on hold and then you can unhold it and get the call...the calling party hears music on hold first and then the hears the called party...pretty neat little game to play when this happens...

 

seach for occurrences of "received on" within the log to see...

 

[3] 2008/04/10 09:50:35: PSTN: Channel 0 going to RING

[5] 2008/04/10 09:50:38: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[5] 2008/04/10 09:50:38: PSTN: Received on 0: Caller-ID 3255131178

[5] 2008/04/10 09:50:38: PSTN: Received on 0: Name THE SOUND SHOP

[9] 2008/04/10 09:50:38: SIP Rx udp:127.0.0.1:5062:

INVITE sip:3256725804@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>

Call-ID: baaab916@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 INVITE

Content-Type: application/sdp

Content-Length: 137

 

v=0

o=root 0 0 IN IP4 1.1.1.2

s=-

c=IN IP4 1.1.1.2

t=0 0

m=audio 2078 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 55018

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 55019

[5] 2008/04/10 09:50:38: Identify trunk (IP address/port and domain match) 5

[9] 2008/04/10 09:50:38: Resolve 425: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 425: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 425: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 2008/04/10 09:50:38: PSTN: eVAPI_CALLER_ID_DETECTED_EVENT

[7] 2008/04/10 09:50:38: Set packet length to 20

[6] 2008/04/10 09:50:38: Sending RTP for baaab916@fxo#c00cbed448 to 1.1.1.2:2078

[5] 2008/04/10 09:50:38: Trunk PSTN1 sends call to 72

[8] 2008/04/10 09:50:38: Play audio_moh/noise.wav

[7] 2008/04/10 09:50:38: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:38: Hunt group 72 called 2 registrations

[5] 2008/04/10 09:50:38: PSTN: Received on 1: Caller-ID 3255131178

[7] 2008/04/10 09:50:38: Set packet length to 20

[9] 2008/04/10 09:50:38: Resolve 426: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 426: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 426: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 62690

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 62691

[9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label

[9] 2008/04/10 09:50:38: Resolve 427: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:38: Resolve 427: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: Resolve 427: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>

Call-ID: be4800cd@pbx

CSeq: 523 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 1814072997 1814072997 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 62690 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:27Kzdm31wQsZGrs5Zita0i0FmlEETAAOWJjtOnpf

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 60046

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 60047

[9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label

[9] 2008/04/10 09:50:38: Resolve 428: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:38: Resolve 428: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: Resolve 428: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358:

INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>

Call-ID: c2b83f2d@pbx

CSeq: 972 INVITE

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 1111676072 1111676072 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 60046 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:RtoOcTTgnzl68D3H8EsfiDhZA5c5WanMJShUs7L1

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[5] 2008/04/10 09:50:38: PSTN: Received on 1: Name THE SOUND SHOP

[9] 2008/04/10 09:50:38: SIP Rx udp:127.0.0.1:5062:

INVITE sip:3256723475@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>

Call-ID: cc458e19@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 INVITE

Content-Type: application/sdp

Content-Length: 137

 

v=0

o=root 0 0 IN IP4 1.1.1.2

s=-

c=IN IP4 1.1.1.2

t=0 0

m=audio 2080 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 55816

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 55817

[5] 2008/04/10 09:50:38: Identify trunk (IP address/port and domain match) 5

[9] 2008/04/10 09:50:38: Resolve 429: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 429: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 429: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Content-Length: 0

 

 

[7] 2008/04/10 09:50:38: Set packet length to 20

[6] 2008/04/10 09:50:38: Sending RTP for cc458e19@fxo#9f853f62b9 to 1.1.1.2:2080

[5] 2008/04/10 09:50:38: Trunk PSTN1 sends call to 72

[5] 2008/04/10 09:50:38: PSTN: Response code: 100

[5] 2008/04/10 09:50:38: PSTN: Response code: 183

[8] 2008/04/10 09:50:38: Play audio_moh/noise.wav

[7] 2008/04/10 09:50:38: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:38: Hunt group 72 called 2 registrations

[7] 2008/04/10 09:50:38: Set packet length to 20

[9] 2008/04/10 09:50:38: Resolve 430: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 430: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: Resolve 430: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:38: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 57200

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 57201

[9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.101:2614;transport=tls because of flow-label

[9] 2008/04/10 09:50:38: Resolve 431: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:38: Resolve 431: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: Resolve 431: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>

Call-ID: ce792cdd@pbx

CSeq: 9718 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 386

 

v=0

o=- 2036358899 2036358899 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 57200 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:SFJvIVfVvQWsVrf5g6YT7Z0hvPDWo5WOmVuxCkSK

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 50794

[7] 2008/04/10 09:50:38: UDP: Opening socket on port 50795

[9] 2008/04/10 09:50:38: Using outbound proxy sip:192.168.1.102:2358;transport=tls because of flow-label

[9] 2008/04/10 09:50:38: Resolve 432: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:38: Resolve 432: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: Resolve 432: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358:

INVITE sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>

Call-ID: a7ef3565@pbx

CSeq: 941 INVITE

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 384

 

v=0

o=- 150428609 150428609 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 50794 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8edFl/syNBzvJPOSuoErFTI1Wb0z5Zqgk87i/gfQ

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

Call-ID: be4800cd@pbx

CSeq: 523 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:50:38: Resolve 433: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:38: Resolve 433: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: Resolve 433: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614:

PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-72c1970c907da13fa76269fb3e59c8cd;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

Call-ID: be4800cd@pbx

CSeq: 524 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

RAck: 1 523 INVITE

Content-Length: 0

 

 

[5] 2008/04/10 09:50:38: PSTN: Response code: 100

[8] 2008/04/10 09:50:38: Play audio_en/ringback.wav

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 972 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:50:38: Resolve 434: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:38: Resolve 434: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: Resolve 434: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358:

PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-80fa95266c2da1afd45ffd6c51e9ebb3;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 973 PRACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

RAck: 1 972 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:50:38: Play audio_en/ringback.wav

[5] 2008/04/10 09:50:38: PSTN: Response code: 183

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

Call-ID: ce792cdd@pbx

CSeq: 9718 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:50:38: Resolve 435: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:38: Resolve 435: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: Resolve 435: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.101:2614:

PRACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-42c3ee21b2b22d876cfb3d1fd4fac0ea;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

Call-ID: ce792cdd@pbx

CSeq: 9719 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

RAck: 1 9718 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:50:38: Play audio_en/ringback.wav

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 941 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Require: 100rel

RSeq: 1

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Content-Length: 0

 

 

[9] 2008/04/10 09:50:38: Resolve 436: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:38: Resolve 436: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: Resolve 436: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:38: SIP Tx tls:192.168.1.102:2358:

PRACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2ba2e9f29dcce53650d8d5f6ac5d1f23;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 942 PRACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

RAck: 1 941 INVITE

Content-Length: 0

 

 

[8] 2008/04/10 09:50:38: Play audio_en/ringback.wav

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-72c1970c907da13fa76269fb3e59c8cd;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

Call-ID: be4800cd@pbx

CSeq: 524 PRACK

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:38: Call be4800cd@pbx#1213919521: Clear last request

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-42c3ee21b2b22d876cfb3d1fd4fac0ea;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

Call-ID: ce792cdd@pbx

CSeq: 9719 PRACK

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:38: Call ce792cdd@pbx#1658042394: Clear last request

[9] 2008/04/10 09:50:38: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-80fa95266c2da1afd45ffd6c51e9ebb3;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 973 PRACK

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:38: Call c2b83f2d@pbx#829959946: Clear last request

[9] 2008/04/10 09:50:39: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-2ba2e9f29dcce53650d8d5f6ac5d1f23;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 942 PRACK

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:39: Call a7ef3565@pbx#184923319: Clear last request

[9] 2008/04/10 09:50:39: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:39: PSTN: Response code: 183

[9] 2008/04/10 09:50:39: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:39: PSTN: Response code: 183

[3] 2008/04/10 09:50:39: PSTN: Channel 0 going to NO_RING

[9] 2008/04/10 09:50:40: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:40: PSTN: Response code: 183

[9] 2008/04/10 09:50:40: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:40: PSTN: Response code: 183

[3] 2008/04/10 09:50:41: PSTN: Channel 0 going to RING

[5] 2008/04/10 09:50:41: PSTN: Ringing, but last invite = 1

[9] 2008/04/10 09:50:42: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:42: PSTN: Response code: 183

[9] 2008/04/10 09:50:42: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:42: PSTN: Response code: 183

[7] 2008/04/10 09:50:43: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:43: Hunt group 72 called 0 registrations

[7] 2008/04/10 09:50:43: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:43: Hunt group 72 called 0 registrations

[9] 2008/04/10 09:50:46: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:46: PSTN: Response code: 183

[3] 2008/04/10 09:50:46: PSTN: Channel 0 going to NO_RING

[9] 2008/04/10 09:50:46: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:46: PSTN: Response code: 183

[3] 2008/04/10 09:50:47: PSTN: Channel 0 going to RING

[5] 2008/04/10 09:50:47: PSTN: Ringing, but last invite = 1

[7] 2008/04/10 09:50:48: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:48: Hunt group 72 called 0 registrations

[7] 2008/04/10 09:50:48: Hunt Group 72: Moving to next stage

[7] 2008/04/10 09:50:48: Hunt group 72 called 0 registrations

[9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f9daac678e35eccefae7af0c85ba2101;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

Call-ID: be4800cd@pbx

CSeq: 523 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Type: application/sdp

Content-Length: 459

 

v=0

o=root 72619833 72619834 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 63850 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850

a=sendrecv

 

[7] 2008/04/10 09:50:48: Call be4800cd@pbx#1213919521: Clear last INVITE

[6] 2008/04/10 09:50:48: Sending RTP for be4800cd@pbx#1213919521 to 192.168.1.101:63850

[9] 2008/04/10 09:50:48: Resolve 437: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:48: Resolve 437: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:48: Resolve 437: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-f1eef136ef0e7efa6a434109f75c828d;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

To: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

Call-ID: be4800cd@pbx

CSeq: 523 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[7] 2008/04/10 09:50:48: Determine pass-through mode after receiving response

[9] 2008/04/10 09:50:48: Resolve 438: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.102:2358:

CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>

Call-ID: c2b83f2d@pbx

CSeq: 972 CANCEL

Max-Forwards: 70

Reason: SIP;cause=200;text="Call completed elsewhere"

Content-Length: 0

 

 

[9] 2008/04/10 09:50:48: Resolve 439: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:48: Resolve 439: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:48: Resolve 439: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:48: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

Call-ID: baaab916@fxo

CSeq: 1 INVITE

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 206

 

v=0

o=- 963854412 963854412 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55018 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:48: PSTN: Response code: 200

[9] 2008/04/10 09:50:48: SIP Rx udp:127.0.0.1:5062:

ACK sip:3256725804@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

To: <sip:3256725804@localhost;user=phone>

Call-ID: baaab916@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 ACK

Content-Length: 0

 

 

[5] 2008/04/10 09:50:48: PSTN: RTP destination=100007f

[5] 2008/04/10 09:50:48: PSTN: RTP destination=55018

[9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 972 CANCEL

Content-Length: 0

 

 

[7] 2008/04/10 09:50:48: Call c2b83f2d@pbx#829959946: Clear last request

[9] 2008/04/10 09:50:48: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 972 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:48: Call c2b83f2d@pbx#829959946: Clear last INVITE

[9] 2008/04/10 09:50:48: Resolve 440: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:48: Resolve 440: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:48: Resolve 440: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:48: SIP Tx tls:192.168.1.102:2358:

ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-c55a23ffcba2c8431c6995527c6e074e;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=829959946

To: <sip:3256725804@localhost;user=phone>;tag=t153uxg8ia

Call-ID: c2b83f2d@pbx

CSeq: 972 ACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[5] 2008/04/10 09:50:48: INVITE Response: Terminate c2b83f2d@pbx

[7] 2008/04/10 09:50:48: Other Ports: 5

[7] 2008/04/10 09:50:48: Call Port: a7ef3565@pbx#184923319

[7] 2008/04/10 09:50:48: Call Port: baaab916@fxo#c00cbed448

[7] 2008/04/10 09:50:48: Call Port: be4800cd@pbx#1213919521

[7] 2008/04/10 09:50:48: Call Port: cc458e19@fxo#9f853f62b9

[7] 2008/04/10 09:50:48: Call Port: ce792cdd@pbx#1658042394

[5] 2008/04/10 09:50:48: PSTN: RTP OOB codec=101

[6] 2008/04/10 09:50:48: PSTN: Start call on 0

[5] 2008/04/10 09:50:48: PSTN: Channel 0 goes offhook

[3] 2008/04/10 09:50:48: PSTN: Channel 0 going to TALKING

[5] 2008/04/10 09:50:48: PSTN: Country Code set to 64

[5] 2008/04/10 09:50:48: PSTN: Tone Detection set to 0

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-7d3wqn17yyn0;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 521

 

v=0

o=root 72619833 72619835 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850

a=sendonly

 

[7] 2008/04/10 09:50:52: Set packet length to 20

[9] 2008/04/10 09:50:52: Resolve 441: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-7d3wqn17yyn0;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 1 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 1814072997 1814072997 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 62690 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:27Kzdm31wQsZGrs5Zita0i0FmlEETAAOWJjtOnpf

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=recvonly

 

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-795ba654881a546cd196f5b7166c0662;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

Call-ID: ce792cdd@pbx

CSeq: 9718 INVITE

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Content-Type: application/sdp

Content-Length: 463

 

v=0

o=root 1101773985 1101773986 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 54456 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456

a=sendrecv

 

[7] 2008/04/10 09:50:52: Call ce792cdd@pbx#1658042394: Clear last INVITE

[6] 2008/04/10 09:50:52: Sending RTP for ce792cdd@pbx#1658042394 to 192.168.1.101:54456

[9] 2008/04/10 09:50:52: Resolve 442: url sip:192.168.1.101:2614;transport=tls

[9] 2008/04/10 09:50:52: Resolve 442: a tls 192.168.1.101 2614

[9] 2008/04/10 09:50:52: Resolve 442: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-0be30940ce01eeca9c4803e1291e794f;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

To: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

Call-ID: ce792cdd@pbx

CSeq: 9718 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[7] 2008/04/10 09:50:52: Determine pass-through mode after receiving response

[9] 2008/04/10 09:50:52: Resolve 443: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358:

CANCEL sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>

Call-ID: a7ef3565@pbx

CSeq: 941 CANCEL

Max-Forwards: 70

Reason: SIP;cause=200;text="Call completed elsewhere"

Content-Length: 0

 

 

[9] 2008/04/10 09:50:52: Resolve 444: aaaa udp 127.0.0.1 5062

[9] 2008/04/10 09:50:52: Resolve 444: a udp 127.0.0.1 5062

[9] 2008/04/10 09:50:52: Resolve 444: udp 127.0.0.1 5062

[9] 2008/04/10 09:50:52: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fwh4dc3rv8kq;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[5] 2008/04/10 09:50:52: PSTN: Response code: 200

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 941 CANCEL

Content-Length: 0

 

 

[7] 2008/04/10 09:50:52: Call a7ef3565@pbx#184923319: Clear last request

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport=5061

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 941 INVITE

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1

Content-Length: 0

 

 

[7] 2008/04/10 09:50:52: Call a7ef3565@pbx#184923319: Clear last INVITE

[9] 2008/04/10 09:50:52: Resolve 445: url sip:192.168.1.102:2358;transport=tls

[9] 2008/04/10 09:50:52: Resolve 445: a tls 192.168.1.102 2358

[9] 2008/04/10 09:50:52: Resolve 445: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358:

ACK sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-78fdc559082cc3d548b51b53d5840b87;rport

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=184923319

To: <sip:3256723475@localhost;user=phone>;tag=l81zzl3z1b

Call-ID: a7ef3565@pbx

CSeq: 941 ACK

Max-Forwards: 70

Contact: <sip:41@192.168.1.100:5061;transport=tls>

Content-Length: 0

 

 

[5] 2008/04/10 09:50:52: INVITE Response: Terminate a7ef3565@pbx

[7] 2008/04/10 09:50:52: Other Ports: 4

[7] 2008/04/10 09:50:52: Call Port: baaab916@fxo#c00cbed448

[7] 2008/04/10 09:50:52: Call Port: be4800cd@pbx#1213919521

[7] 2008/04/10 09:50:52: Call Port: cc458e19@fxo#9f853f62b9

[7] 2008/04/10 09:50:52: Call Port: ce792cdd@pbx#1658042394

[9] 2008/04/10 09:50:52: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:52: PSTN: Response code: 200

[9] 2008/04/10 09:50:52: SIP Rx tls:192.168.1.102:2358:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-wn0vp59hpqv9;rport

From: "41" <sip:41@localhost>;tag=gmz36br37m

To: "41" <sip:41@localhost>

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14235 REGISTER

Max-Forwards: 70

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:ec9ea11d-0cc3-4c4e-b306-6953a0386af5>"

Contact: <http://192.168.1.102:80>

Contact: <https://192.168.1.102:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.102

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:50:52: Packet authenticated by transport layer

[9] 2008/04/10 09:50:52: Resolve 446: tls 192.168.1.102 2358

[9] 2008/04/10 09:50:52: SIP Tx tls:192.168.1.102:2358:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.102:2060;branch=z9hG4bK-wn0vp59hpqv9;rport=2358

From: "41" <sip:41@localhost>;tag=gmz36br37m

To: "41" <sip:41@localhost>;tag=b4f17a5688

Call-ID: 3c26701aaf59-kw566buonz1c

CSeq: 14235 REGISTER

Contact: <sip:41@192.168.1.102:2060;transport=tls;line=94aa7mxt>;expires=178

Contact: <http://192.168.1.102:80>;expires=178

Contact: <https://192.168.1.102:443>;expires=178

Content-Length: 0

 

 

[9] 2008/04/10 09:50:53: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:53: PSTN: Response code: 200

[9] 2008/04/10 09:50:55: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:55: PSTN: Response code: 200

[9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dqgcmhhzt4ao;rport

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 525

 

v=0

o=root 1101773985 1101773987 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 54456 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456

a=sendonly

 

[7] 2008/04/10 09:50:59: Set packet length to 20

[9] 2008/04/10 09:50:59: Resolve 447: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:59: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dqgcmhhzt4ao;rport=2614

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 1 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 2036358899 2036358899 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 57200 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:SFJvIVfVvQWsVrf5g6YT7Z0hvPDWo5WOmVuxCkSK

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=recvonly

 

[9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-wuzuqb8xkevm;rport

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:50:59: SIP Rx tls:192.168.1.101:2614:

SUBSCRIBE sip:40@localhost;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-89yq4h62rmed;rport

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13678 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:50:59: Packet authenticated by transport layer

[9] 2008/04/10 09:50:59: Resolve 448: tls 192.168.1.101 2614

[9] 2008/04/10 09:50:59: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-89yq4h62rmed;rport=2614

From: <sip:40@localhost>;tag=vgds1qnpk0

To: <sip:40@localhost;user=phone>;tag=9d4243196b

Call-ID: 3c2674105300-yc4ja694jzkk

CSeq: 13678 SUBSCRIBE

Contact: <sip:192.168.1.100:5061;transport=tls>

Expires: 182

Content-Length: 0

 

 

[9] 2008/04/10 09:50:59: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:50:59: PSTN: Response code: 200

[9] 2008/04/10 09:51:04: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-0ox4u2ne519o;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 521

 

v=0

o=root 72619833 72619836 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850

a=sendrecv

 

[7] 2008/04/10 09:51:04: Set packet length to 20

[9] 2008/04/10 09:51:04: Resolve 449: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:04: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-0ox4u2ne519o;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 2 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 1814072997 1814072998 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 62690 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uKJD47d2GRAzTPXN7IibzoainkabVkoqCYHu2afW

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[9] 2008/04/10 09:51:05: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-617u8w78jtdr;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:51:06: SIP Rx tls:192.168.1.101:2614:

REGISTER sip:localhost SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-d4l0jblczcy7;rport

From: "40" <sip:40@localhost>;tag=aje04aahuz

To: "40" <sip:40@localhost>

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14230 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:822ed566-4660-4bfb-90c3-2b5411c69eb0>"

Contact: <http://192.168.1.101:80>

Contact: <https://192.168.1.101:443>

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Proxy-Require: buttons

Expires: 3600

Content-Length: 0

 

 

[8] 2008/04/10 09:51:06: Packet authenticated by transport layer

[9] 2008/04/10 09:51:06: Resolve 450: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:06: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-d4l0jblczcy7;rport=2614

From: "40" <sip:40@localhost>;tag=aje04aahuz

To: "40" <sip:40@localhost>;tag=c02c2dcda3

Call-ID: 3c26741018ce-d9hh4ncip2ol

CSeq: 14230 REGISTER

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;expires=180

Contact: <http://192.168.1.101:80>;expires=180

Contact: <https://192.168.1.101:443>;expires=180

Content-Length: 0

 

 

[5] 2008/04/10 09:51:06: PSTN: Busy Tone detected on 0

[9] 2008/04/10 09:51:07: SIP Tr udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

To: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

Call-ID: cc458e19@fxo

CSeq: 1 INVITE

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 208

 

v=0

o=- 2066418937 2066418937 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 55816 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[5] 2008/04/10 09:51:07: PSTN: Response code: 200

[9] 2008/04/10 09:51:12: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-84aeuxcd83pr;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 3 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 521

 

v=0

o=root 72619833 72619837 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850

a=sendonly

 

[7] 2008/04/10 09:51:12: Set packet length to 20

[9] 2008/04/10 09:51:12: Resolve 451: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:12: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-84aeuxcd83pr;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 3 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 1814072997 1814072998 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 62690 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uKJD47d2GRAzTPXN7IibzoainkabVkoqCYHu2afW

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=recvonly

 

[9] 2008/04/10 09:51:12: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-g5kd9vd2hgs8;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 3 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:51:19: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-rcnakugffl5x;rport

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 525

 

v=0

o=root 1101773985 1101773988 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 54456 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:6EntKzyU6HUFgzR0HQ1lQeVZ/vormeVvpmowjswY

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 54456

a=sendrecv

 

[7] 2008/04/10 09:51:19: Set packet length to 20

[9] 2008/04/10 09:51:19: Resolve 452: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:19: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-rcnakugffl5x;rport=2614

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 2 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 2036358899 2036358900 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 57200 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X3jfQy9VXD+IqDRVlL3wYkOSAIktV9RYYFVGOf0c

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[9] 2008/04/10 09:51:19: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-2to6qqqswo3y;rport

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[3] 2008/04/10 09:51:20: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:51:31: SIP Rx tls:192.168.1.101:2614:

BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fk7owztky7tb;rport

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=19,Rx_Pkts=12,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=968,Tx_Pkts=615,Remote_Tx_Pkts=0

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:51:31: Resolve 453: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:31: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-fk7owztky7tb;rport=2614

From: <sip:3256723475@localhost;user=phone>;tag=k67nty7ini

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1658042394

Call-ID: ce792cdd@pbx

CSeq: 3 BYE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

User-Agent: pbxnsip-PBX/3.0.0.2899

RTP-RxStat: Dur=53,Pkt=971,Oct=170896,Underun=1946

RTP-TxStat: Dur=40,Pkt=29,Oct=3556

Content-Length: 0

 

 

[7] 2008/04/10 09:51:31: Other Ports: 3

[7] 2008/04/10 09:51:31: Call Port: baaab916@fxo#c00cbed448

[7] 2008/04/10 09:51:31: Call Port: be4800cd@pbx#1213919521

[7] 2008/04/10 09:51:31: Call Port: cc458e19@fxo#9f853f62b9

[8] 2008/04/10 09:51:31: SMTP: Connect to 68.142.200.11:25

[9] 2008/04/10 09:51:31: Resolve 454: url sip:127.0.0.1:5062

[9] 2008/04/10 09:51:31: Resolve 454: udp 127.0.0.1 5062

[9] 2008/04/10 09:51:31: SIP Tx udp:127.0.0.1:5062:

BYE sip:127.0.0.1:5062 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-29d2771eb98b126bb76f455f73d2498d;rport

From: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

Call-ID: cc458e19@fxo

CSeq: 16490 BYE

Max-Forwards: 70

Contact: <sip:3256723475@127.0.0.1:5060;transport=udp>

RTP-RxStat: Dur=53,Pkt=0,Oct=0,Underun=12

RTP-TxStat: Dur=40,Pkt=2628,Oct=452016

Content-Length: 0

 

 

[9] 2008/04/10 09:51:31: SIP Rx udp:127.0.0.1:5062:

SIP/2.0 404 Not found

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-29d2771eb98b126bb76f455f73d2498d;rport

From: <sip:3256723475@localhost;user=phone>;tag=9f853f62b9

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1330573317

Call-ID: cc458e19@fxo

CSeq: 16490 BYE

Content-Length: 0

 

 

[7] 2008/04/10 09:51:31: Call cc458e19@fxo#9f853f62b9: Clear last request

[5] 2008/04/10 09:51:31: BYE Response: Terminate cc458e19@fxo

[7] 2008/04/10 09:51:31: Other Ports: 2

[7] 2008/04/10 09:51:31: Call Port: baaab916@fxo#c00cbed448

[7] 2008/04/10 09:51:31: Call Port: be4800cd@pbx#1213919521

[8] 2008/04/10 09:51:31: SMTP: Received 220 smtp104.biz.mail.mud.yahoo.com ESMTP

 

[8] 2008/04/10 09:51:31: SMTP: Received 250-smtp104.biz.mail.mud.yahoo.com

250-AUTH LOGIN PLAIN XYMCOOKIE

250-PIPELINING

250 8BITMIME

 

[8] 2008/04/10 09:51:32: SMTP: Received 334 VXNlcm5hbWU6

 

[8] 2008/04/10 09:51:32: SMTP: Received 334 UGFzc3dvcmQ6

 

[8] 2008/04/10 09:51:32: SMTP: Received 235 ok, go ahead (#2.0.0)

 

[8] 2008/04/10 09:51:32: SMTP: Received 250 ok

 

[8] 2008/04/10 09:51:32: Last message repeated 2 times

[8] 2008/04/10 09:51:32: SMTP: Received 354 go ahead

 

[8] 2008/04/10 09:51:33: SMTP: Received 250 ok 1207839092 qp 10292

 

[8] 2008/04/10 09:51:33: SMTP: Received 221 smtp104.biz.mail.mud.yahoo.com

 

[8] 2008/04/10 09:51:33: Sucessfully sent email to <russell@thesoundshop-llc.com>

[3] 2008/04/10 09:51:33: SMTP: Cannot resolve smtp.mail.thesoundshop-llc.com

[9] 2008/04/10 09:51:34: SIP Rx tls:192.168.1.101:2614:

INVITE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dzad3hlzhe4z;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 4 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom300/7.1.30

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 521

 

v=0

o=root 72619833 72619838 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 63850 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iQOaQK1ugUoXci63fXVnXrqLQvOnvTlwm79vmtjx

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=encryption:optional

a=alt:1 0.9 : user 9kksj== 192.168.1.101 63850

a=sendrecv

 

[7] 2008/04/10 09:51:34: Set packet length to 20

[9] 2008/04/10 09:51:34: Resolve 455: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:34: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-dzad3hlzhe4z;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 4 INVITE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.0.2899

Content-Type: application/sdp

Content-Length: 398

 

v=0

o=- 1814072997 1814072999 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 62690 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+wKRozbmN/PWmTdomsCW8ZTuUeo0i+dQYEPkFOOn

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[9] 2008/04/10 09:51:34: SIP Rx tls:192.168.1.101:2614:

ACK sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-mhb8iebg71pe;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 4 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

Proxy-Require: buttons

Content-Length: 0

 

 

[5] 2008/04/10 09:51:42: PSTN: Busy Tone detected on 0

[9] 2008/04/10 09:51:43: SIP Rx tls:192.168.1.101:2614:

BYE sip:40@192.168.1.100:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-5fu8xaouui7x;rport

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 5 BYE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2060;transport=tls;line=t4hgou5n>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=1015,Rx_Pkts=469,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=1016,Tx_Pkts=471,Remote_Tx_Pkts=0

Proxy-Require: buttons

Content-Length: 0

 

 

[9] 2008/04/10 09:51:43: Resolve 456: tls 192.168.1.101 2614

[9] 2008/04/10 09:51:43: SIP Tx tls:192.168.1.101:2614:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2060;branch=z9hG4bK-5fu8xaouui7x;rport=2614

From: <sip:3256725804@localhost;user=phone>;tag=ve112iifw2

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1213919521

Call-ID: be4800cd@pbx

CSeq: 5 BYE

Contact: <sip:40@192.168.1.100:5061;transport=tls>

User-Agent: pbxnsip-PBX/3.0.0.2899

RTP-RxStat: Dur=65,Pkt=1016,Oct=178816,Underun=6

RTP-TxStat: Dur=55,Pkt=1043,Oct=180816

Content-Length: 0

 

 

[7] 2008/04/10 09:51:43: Other Ports: 1

[7] 2008/04/10 09:51:43: Call Port: baaab916@fxo#c00cbed448

[8] 2008/04/10 09:51:44: SMTP: Connect to 68.142.200.11:25

[9] 2008/04/10 09:51:44: Resolve 457: url sip:127.0.0.1:5062

[9] 2008/04/10 09:51:44: Resolve 457: udp 127.0.0.1 5062

[9] 2008/04/10 09:51:44: SIP Tx udp:127.0.0.1:5062:

BYE sip:127.0.0.1:5062 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-073f078b6f7a519cac4f081ed18ed0ec;rport

From: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

Call-ID: baaab916@fxo

CSeq: 24189 BYE

Max-Forwards: 70

Contact: <sip:3256725804@127.0.0.1:5060;transport=udp>

RTP-RxStat: Dur=66,Pkt=2754,Oct=473688,Underun=40

RTP-TxStat: Dur=55,Pkt=3233,Oct=556076

Content-Length: 0

 

 

[5] 2008/04/10 09:51:44: PSTN: Received BYE message on channel 0

[9] 2008/04/10 09:51:44: SIP Rx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-073f078b6f7a519cac4f081ed18ed0ec;rport

From: <sip:3256725804@localhost;user=phone>;tag=c00cbed448

To: "THE SOUND SHOP " <sip:3255131178@localhost;user=phone>;tag=1956297539

Call-ID: baaab916@fxo

CSeq: 24189 BYE

Content-Length: 0

 

 

[7] 2008/04/10 09:51:44: Call baaab916@fxo#c00cbed448: Clear last request

[5] 2008/04/10 09:51:44: BYE Response: Terminate baaab916@fxo

[3] 2008/04/10 09:51:44: PSTN: Channel 0: Hangup

[5] 2008/04/10 09:51:44: PSTN: Channel 0 goes onhook

[5] 2008/04/10 09:51:44: PSTN: enable_callerid 0

[3] 2008/04/10 09:51:44: PSTN: Channel 0 going to GO_ONHOOK

[8] 2008/04/10 09:51:44: SMTP: Received 220 smtp108.biz.mail.mud.yahoo.com ESMTP

 

[8] 2008/04/10 09:51:44: SMTP: Received 250-smtp108.biz.mail.mud.yahoo.com

250-AUTH LOGIN PLAIN XYMCOOKIE

250-PIPELINING

250 8BITMIME

 

[8] 2008/04/10 09:51:44: SMTP: Received 334 VXNlcm5hbWU6

 

[8] 2008/04/10 09:51:44: SMTP: Received 334 UGFzc3dvcmQ6

 

[8] 2008/04/10 09:51:44: SMTP: Received 235 ok, go ahead (#2.0.0)

 

[8] 2008/04/10 09:51:44: SMTP: Received 250 ok

 

[8] 2008/04/10 09:51:44: Last message repeated 2 times

[8] 2008/04/10 09:51:44: SMTP: Received 354 go ahead

 

[3] 2008/04/10 09:51:45: PSTN: Channel 0 going to IDLE

[8] 2008/04/10 09:51:45: SMTP: Received 250 ok 1207839104 qp 3575

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