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dissconnected calls

russelln

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russelln Newbie

russelln

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I just purchased a pbx cs 410, it is running 2.0.9.2059, i currently have it connected to 2 snom 300 phones running 7.1.30, both phones register and can call the auto attendant, the voice mail, and call each other just fine, and all the functions work, speaker phone, hold, etc..when I call an outside line, the calls terminated as soon as the call starts to ring, when calling into the appliance the extension will ring, the hunt group will ring the extensions, but as soon as the call is answered it will dissconnect the call, I have the devices running through a little linksys 4 port router and my computers are at the end of the phones. I have spent all weekend trying things, I have pushed the software back on the phones, that did not help, I pushed it forward again, in the system log I sometimes get the error line seize not sent, but I get Using codecs pcmu pcma g726-32 gsm telephone-event and then last message sent 150 to 160 times right before the call is terminated...

 

any ideas?

 

thanks :D:blink:

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russelln Newbie

russelln

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Is this one of the white CS410's or a the new black ones? If you sent the log level to 9 and turn on all the logging can you attach that to a post?

It is one of the white ones, it is the unit i received for going to conference..

 

this is log of an phone made from the phone going to an outside line.

 

[9] 2008/01/28 09:44:15: SIP Rx tls:192.168.1.101:2085:

REGISTER sip:192.168.1.100 SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-9oeskr5zhfjz;rport

From: "40" <sip:40@192.168.1.100>;tag=ih8boluj3o

To: "40" <sip:40@192.168.1.100>

Call-ID: 3c2670179315-6efy8i6x6pem

CSeq: 18 REGISTER

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:2ca48eb6-7c90-4c3b-9629-44d87d6ef467>"

User-Agent: snom300/7.1.30

Supported: gruu

Allow-Events: dialog

X-Real-IP: 192.168.1.101

Expires: 3600

Content-Length: 0

 

 

[8] 2008/01/28 09:44:15: Packet authenticated by transport layer

[9] 2008/01/28 09:44:15: SIP Rx tls:192.168.1.101:2085:

SUBSCRIBE sip:40@192.168.1.100;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-074bqznz71rr;rport

From: <sip:40@192.168.1.100>;tag=x8j92jnnj3

To: <sip:40@192.168.1.100;user=phone>;tag=c90edd3e33

Call-ID: 3c2673ccd6bc-dx2ek6btaki0

CSeq: 10 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom300/7.1.30

Expires: 3600

Content-Length: 0

 

 

[8] 2008/01/28 09:44:15: Packet authenticated by transport layer

[9] 2008/01/28 09:44:16: Resolve destination 1399: tls 192.168.1.101 2085

[9] 2008/01/28 09:44:16: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-9oeskr5zhfjz;rport=2085

From: "40" <sip:40@192.168.1.100>;tag=ih8boluj3o

To: "40" <sip:40@192.168.1.100>;tag=3161a91250

Call-ID: 3c2670179315-6efy8i6x6pem

CSeq: 18 REGISTER

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1;q=1.0;+sip.instance="<urn:uuid:2ca48eb6-7c90-4c3b-9629-44d87d6ef467>";expires=180

Content-Length: 0

 

 

[9] 2008/01/28 09:44:16: Resolve destination 1400: tls 192.168.1.101 2085

[9] 2008/01/28 09:44:16: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-074bqznz71rr;rport=2085

From: <sip:40@192.168.1.100>;tag=x8j92jnnj3

To: <sip:40@192.168.1.100;user=phone>;tag=c90edd3e33

Call-ID: 3c2673ccd6bc-dx2ek6btaki0

CSeq: 10 SUBSCRIBE

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

Expires: 180

Content-Length: 0

 

 

[9] 2008/01/28 09:44:47: SIP Rx tls:192.168.1.101:2085:

INVITE sip:6910549@192.168.1.100;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-mag1fld2pbwy;rport

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;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

Content-Type: application/sdp

Content-Length: 523

 

v=0

o=root 346434938 346434938 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 58548 RTP/AVP 0 8 9 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:hjzPjBq0mUBGsBx6W3NhOXt6s36iIS/G+DadDosj

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: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 58548

a=sendrecv

 

[8] 2008/01/28 09:44:47: Packet authenticated by transport layer

[7] 2008/01/28 09:44:47: UDP: Opening socket on port 53798

[7] 2008/01/28 09:44:47: UDP: Opening socket on port 53799

[8] 2008/01/28 09:44:47: Could not find a trunk (1 trunks)

[9] 2008/01/28 09:44:47: Using outbound proxy sip:192.168.1.101:2085;transport=tls because of flow-label

[9] 2008/01/28 09:44:47: Last message repeated 2 times

[9] 2008/01/28 09:44:47: Resolve destination 1401: tls 192.168.1.101 2085

[9] 2008/01/28 09:44:47: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-mag1fld2pbwy;rport=2085

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 1 INVITE

Content-Length: 0

 

 

[6] 2008/01/28 09:45:13: Sending RTP to 192.168.1.101:58548

[5] 2008/01/28 09:45:13: Dialplan: Match 6910549@192.168.1.100 to <sip:6910549@127.0.0.1;user=phone> on trunk PSTN

[7] 2008/01/28 09:45:13: UDP: Opening socket on port 51370

[7] 2008/01/28 09:45:13: UDP: Opening socket on port 51371

[8] 2008/01/28 09:45:13: Play audio_moh/noise.wav

[5] 2008/01/28 09:45:13: Using codecs pcmu pcma g726-32 gsm telephone-event

[9] 2008/01/28 09:45:13: Resolve destination 1402: url sip:127.0.0.1:5062

[9] 2008/01/28 09:45:13: Resolve destination 1402: udp 127.0.0.1 5062

[9] 2008/01/28 09:45:13: SIP Tx udp:127.0.0.1:5062:

INVITE sip:6910549@127.0.0.1;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-a688476afdffb24790f4220e7e86a7ea;rport

From: "Russell " <sip:40@127.0.0.1>;tag=1010686817

To: <sip:6910549@127.0.0.1;user=phone>

Call-ID: 0fcf3ef3@pbx

CSeq: 25334 INVITE

Max-Forwards: 70

Contact: <sip:40@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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 251

 

v=0

o=- 392626792 392626792 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 51370 RTP/AVP 0 8 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[5] 2008/01/28 09:45:13: Using codecs pcmu pcma g726-32 gsm telephone-event

[9] 2008/01/28 09:45:13: Resolve destination 1403: tls 192.168.1.101 2085

[9] 2008/01/28 09:45:13: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 183 Ringing

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-mag1fld2pbwy;rport=2085

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

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/2.0.9.2059

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 343

 

v=0

o=- 218325793 218325793 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 53798 RTP/AVP 0 8 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wWfmcsrIJGoPJKEQQ3PGtAN+vTotZffWGS82QDew

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[9] 2008/01/28 09:45:13: SIP Rx udp:127.0.0.1:5062:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-a688476afdffb24790f4220e7e86a7ea;rport

From: "Russell " <sip:40@127.0.0.1>;tag=1010686817

To: <sip:6910549@127.0.0.1;user=phone>

Call-ID: 0fcf3ef3@pbx

CSeq: 25334 INVITE

Content-Length: 0

 

 

[9] 2008/01/28 09:45:13: SIP Rx tls:192.168.1.101:2085:

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

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-meqxbi3qpnp7;rport

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

RAck: 1 1 INVITE

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

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

Content-Length: 0

 

 

[8] 2008/01/28 09:45:13: Packet authenticated by transport layer

[9] 2008/01/28 09:45:13: Resolve destination 1404: tls 192.168.1.101 2085

[9] 2008/01/28 09:45:13: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-meqxbi3qpnp7;rport=2085

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 2 PRACK

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

User-Agent: pbxnsip-PBX/2.0.9.2059

Content-Length: 0

 

 

[6] 2008/01/28 09:45:13: Sending RTP to 1.1.1.2:2066

[8] 2008/01/28 09:45:13: No codec available for sending

[8] 2008/01/28 09:45:16: Last message repeated 161 times

 

[9] 2008/01/28 09:45:16: 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-a688476afdffb24790f4220e7e86a7ea;rport

From: "Russell " <sip:40@127.0.0.1>;tag=1010686817

To: <sip:6910549@127.0.0.1;user=phone>;tag=1

Call-ID: 0fcf3ef3@pbx

CSeq: 25334 INVITE

Contact: <sip:127.0.0.1:5062>

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 2066 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

 

[7] 2008/01/28 09:45:16: Call 0fcf3ef3@pbx#1010686817: Clear last INVITE

[9] 2008/01/28 09:45:16: Resolve destination 1405: url sip:127.0.0.1:5062

[9] 2008/01/28 09:45:16: Resolve destination 1405: udp 127.0.0.1 5062

[9] 2008/01/28 09:45:16: SIP Tx udp:127.0.0.1:5062:

ACK sip:127.0.0.1:5062 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-56a7871a722ffbb5fe979666f3fe0d28;rport

From: "Russell " <sip:40@127.0.0.1>;tag=1010686817

To: <sip:6910549@127.0.0.1;user=phone>;tag=1

Call-ID: 0fcf3ef3@pbx

CSeq: 25334 ACK

Max-Forwards: 70

Contact: <sip:40@127.0.0.1:5060;transport=udp>

Content-Length: 0

 

 

[9] 2008/01/28 09:45:16: Resolve destination 1406: tls 192.168.1.101 2085

[9] 2008/01/28 09:45:16: SIP Tx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-mag1fld2pbwy;rport=2085

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 343

 

v=0

o=- 218325793 218325793 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 53798 RTP/AVP 0 8 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wWfmcsrIJGoPJKEQQ3PGtAN+vTotZffWGS82QDew

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[7] 2008/01/28 09:45:16: 0fcf3ef3@pbx#1010686817: RTP pass-through mode

 

[7] 2008/01/28 09:45:16: 3c2677535a15-27mtkizzzbyb#397d2a4024: RTP pass-through mode

 

[9] 2008/01/28 09:45:16: SIP Rx tls:192.168.1.101:2085:

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

Via: SIP/2.0/TLS 192.168.1.101:2051;branch=z9hG4bK-7vpodli0fy1v;rport

From: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

To: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

Content-Length: 0

 

 

[8] 2008/01/28 09:45:16: Packet authenticated by transport layer

[9] 2008/01/28 09:45:17: SIP Rx udp:127.0.0.1:5062:

BYE sip:40@127.0.0.1:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: <sip:6910549@127.0.0.1;user=phone>;tag=1

To: "Russell " <sip:40@127.0.0.1>;tag=1010686817

Call-ID: 0fcf3ef3@pbx

Contact: <sip:127.0.0.1:5062>

CSeq: 1 BYE

Content-Length: 0

 

 

[9] 2008/01/28 09:45:17: Resolve destination 1407: a udp 127.0.0.1 5062

[9] 2008/01/28 09:45:17: Resolve destination 1407: udp 127.0.0.1 5062

[9] 2008/01/28 09:45:17: SIP Tx udp:127.0.0.1:5062:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 127.0.0.1:5062

From: <sip:6910549@127.0.0.1;user=phone>;tag=1

To: "Russell " <sip:40@127.0.0.1>;tag=1010686817

Call-ID: 0fcf3ef3@pbx

CSeq: 1 BYE

Contact: <sip:40@127.0.0.1:5060;transport=udp>

User-Agent: pbxnsip-PBX/2.0.9.2059

RTP-RxStat: Dur=4,Pkt=209,Oct=35948,Underun=0

RTP-TxStat: Dur=1,Pkt=49,Oct=8428

Content-Length: 0

 

 

[7] 2008/01/28 09:45:17: Other Ports: 1

[7] 2008/01/28 09:45:17: Call Port: 3c2677535a15-27mtkizzzbyb#397d2a4024

[9] 2008/01/28 09:45:17: Resolve destination 1408: url sip:192.168.1.101:2085;transport=tls

[9] 2008/01/28 09:45:17: Resolve destination 1408: a tls 192.168.1.101 2085

[9] 2008/01/28 09:45:17: Resolve destination 1408: tls 192.168.1.101 2085

[9] 2008/01/28 09:45:17: SIP Tx tls:192.168.1.101:2085:

BYE sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-8a781625dd8643aef9fd42c3b3edeefd;rport

From: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

To: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 12128 BYE

Max-Forwards: 70

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

RTP-RxStat: Dur=30,Pkt=208,Oct=36608,Underun=0

RTP-TxStat: Dur=1,Pkt=213,Oct=37488

Content-Length: 0

 

 

[9] 2008/01/28 09:45:17: SIP Rx tls:192.168.1.101:2085:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-8a781625dd8643aef9fd42c3b3edeefd;rport=5061

From: <sip:6910549@192.168.1.100;user=phone>;tag=397d2a4024

To: "40" <sip:40@192.168.1.100>;tag=cre86y56cp

Call-ID: 3c2677535a15-27mtkizzzbyb

CSeq: 12128 BYE

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=213,Rx_Pkts=213,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=1478,Tx_Pkts=1478,Remote_Tx_Pkts=0

Content-Length: 0

 

 

[7] 2008/01/28 09:45:17: Call 3c2677535a15-27mtkizzzbyb#397d2a4024: Clear last request

[5] 2008/01/28 09:45:17: BYE Response: Terminate 3c2677535a15-27mtkizzzbyb

[8] 2008/01/28 09:45:22: Route: eth0 c0a80100 ffffff00

[8] 2008/01/28 09:45:22: Route: eth1 01010100 ffffff00

[8] 2008/01/28 09:45:22: Default Route uses 192.168.1.100

 

THIS IS THE LOG OF AN OUTSIDE CALL BEING ANSWERED

 

[9] 2008/01/28 09:56:14: SIP Rx udp:127.0.0.1:5062:

INVITE sip:2121212121@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1059961393

To: <sip:2121212121@localhost;user=phone>

Call-ID: 83a5d578@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 2070 RTP/AVP 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

 

[7] 2008/01/28 09:56:14: UDP: Opening socket on port 57956

[7] 2008/01/28 09:56:14: UDP: Opening socket on port 57957

[5] 2008/01/28 09:56:14: Identify trunk (IP address/port and domain match) 1

[9] 2008/01/28 09:56:14: Resolve destination 1432: a udp 127.0.0.1 5062

[9] 2008/01/28 09:56:14: Resolve destination 1432: udp 127.0.0.1 5062

[9] 2008/01/28 09:56:14: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Content-Length: 0

 

 

[6] 2008/01/28 09:56:26: Sending RTP to 1.1.1.2:2070

[5] 2008/01/28 09:56:26: Trunk PSTN sends call to 72

[8] 2008/01/28 09:56:26: Play audio_moh/noise.wav

[7] 2008/01/28 09:56:26: Hunt Group: Moving to next stage

[5] 2008/01/28 09:56:26: Using codecs pcmu telephone-event

[9] 2008/01/28 09:56:26: Resolve destination 1433: a udp 127.0.0.1 5062

[9] 2008/01/28 09:56:26: Resolve destination 1433: udp 127.0.0.1 5062

[9] 2008/01/28 09:56:26: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:40@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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 177

 

v=0

o=- 797383531 797383531 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 57956 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[7] 2008/01/28 09:56:26: UDP: Opening socket on port 61568

[7] 2008/01/28 09:56:26: UDP: Opening socket on port 61569

[9] 2008/01/28 09:56:26: Using outbound proxy sip:192.168.1.101:2085;transport=tls because of flow-label

[5] 2008/01/28 09:56:26: Using codecs pcmu pcma g726-32 gsm telephone-event

[9] 2008/01/28 09:56:26: Resolve destination 1434: url sip:192.168.1.101:2085;transport=tls

[9] 2008/01/28 09:56:26: Resolve destination 1434: a tls 192.168.1.101 2085

[9] 2008/01/28 09:56:26: Resolve destination 1434: tls 192.168.1.101 2085

[9] 2008/01/28 09:56:26: SIP Tx tls:192.168.1.101:2085:

INVITE sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-880d8ed34b7ac0470331457a55f1219b;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>

Call-ID: 55966ee1@pbx

CSeq: 8270 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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 343

 

v=0

o=- 446442818 446442818 IN IP4 192.168.1.100

s=-

c=IN IP4 192.168.1.100

t=0 0

m=audio 61568 RTP/AVP 0 8 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:I1kQVBlECm7dqeXivzgeDopTcxTJqbJ3CyPFi9M+

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[9] 2008/01/28 09:56:26: SIP Rx tls:192.168.1.101:2085:

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-880d8ed34b7ac0470331457a55f1219b;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8270 INVITE

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;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] 2008/01/28 09:56:26: Resolve destination 1435: url sip:192.168.1.101:2085;transport=tls

[9] 2008/01/28 09:56:26: Resolve destination 1435: a tls 192.168.1.101 2085

[9] 2008/01/28 09:56:26: Resolve destination 1435: tls 192.168.1.101 2085

[9] 2008/01/28 09:56:26: SIP Tx tls:192.168.1.101:2085:

PRACK sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-43cfd2027efc3903214de87697c32935;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8271 PRACK

Max-Forwards: 70

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

RAck: 1 8270 INVITE

Content-Length: 0

 

 

[8] 2008/01/28 09:56:26: Play audio_en/ringback.wav

[9] 2008/01/28 09:56:26: Resolve destination 1436: a udp 127.0.0.1 5062

[9] 2008/01/28 09:56:26: Resolve destination 1436: udp 127.0.0.1 5062

[9] 2008/01/28 09:56:26: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:40@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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 177

 

v=0

o=- 797383531 797383531 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 57956 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[9] 2008/01/28 09:56:26: SIP Rx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-43cfd2027efc3903214de87697c32935;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8271 PRACK

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

Content-Length: 0

 

 

[7] 2008/01/28 09:56:26: Call 55966ee1@pbx#1923129968: Clear last request

[9] 2008/01/28 09:56:26: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:40@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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 177

 

v=0

o=- 797383531 797383531 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 57956 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[9] 2008/01/28 09:56:30: Last message repeated 6 times

 

[9] 2008/01/28 09:56:30: SIP Rx tls:192.168.1.101:2085:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-880d8ed34b7ac0470331457a55f1219b;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8270 INVITE

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;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: 437

 

v=0

o=root 898463240 898463241 IN IP4 192.168.1.101

s=call

c=IN IP4 192.168.1.101

t=0 0

m=audio 59022 RTP/AVP 0 8 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KYw+3lgyms9J7h/OzBMxHa5qYZtybVHNPjOSYwRa

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/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 59022

a=sendrecv

 

[7] 2008/01/28 09:56:30: Call 55966ee1@pbx#1923129968: Clear last INVITE

[6] 2008/01/28 09:56:30: Sending RTP to 192.168.1.101:59022

[9] 2008/01/28 09:56:30: Resolve destination 1437: url sip:192.168.1.101:2085;transport=tls

[9] 2008/01/28 09:56:30: Resolve destination 1437: a tls 192.168.1.101 2085

[9] 2008/01/28 09:56:30: Resolve destination 1437: tls 192.168.1.101 2085

[9] 2008/01/28 09:56:30: SIP Tx tls:192.168.1.101:2085:

ACK sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-577e463648e521ae9992f47e5fb97bb8;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8270 ACK

Max-Forwards: 70

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

Content-Length: 0

 

 

[9] 2008/01/28 09:56:30: Resolve destination 1438: a udp 127.0.0.1 5062

[9] 2008/01/28 09:56:30: Resolve destination 1438: udp 127.0.0.1 5062

[9] 2008/01/28 09:56:30: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 1 INVITE

Contact: <sip:40@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/2.0.9.2059

Content-Type: application/sdp

Content-Length: 177

 

v=0

o=- 797383531 797383531 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 57956 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

[7] 2008/01/28 09:56:30: 55966ee1@pbx#1923129968: RTP pass-through mode

 

[7] 2008/01/28 09:56:30: 83a5d578@fxo#35bd34fbf7: RTP pass-through mode

 

[9] 2008/01/28 09:56:30: SIP Rx udp:127.0.0.1:5062:

ACK sip:2121212121@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1059961393

To: <sip:2121212121@localhost;user=phone>

Call-ID: 83a5d578@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 1 ACK

Content-Length: 0

 

 

[9] 2008/01/28 09:56:30: SIP Rx udp:127.0.0.1:5062:

BYE sip:2121212121@localhost;user=phone SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5062

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

Contact: <sip:127.0.0.1:5062>

CSeq: 2 BYE

Content-Length: 0

 

 

[9] 2008/01/28 09:56:30: Resolve destination 1439: a udp 127.0.0.1 5062

[9] 2008/01/28 09:56:30: Resolve destination 1439: udp 127.0.0.1 5062

[9] 2008/01/28 09:56:30: 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=1059961393

To: <sip:2121212121@localhost;user=phone>;tag=35bd34fbf7

Call-ID: 83a5d578@fxo

CSeq: 2 BYE

Contact: <sip:40@127.0.0.1:5060;transport=udp>

User-Agent: pbxnsip-PBX/2.0.9.2059

RTP-RxStat: Dur=17,Pkt=18,Oct=3096,Underun=0

RTP-TxStat: Dur=0,Pkt=242,Oct=41624

Content-Length: 0

 

 

[7] 2008/01/28 09:56:30: Other Ports: 1

[7] 2008/01/28 09:56:30: Call Port: 55966ee1@pbx#1923129968

[9] 2008/01/28 09:56:30: Resolve destination 1440: url sip:192.168.1.101:2085;transport=tls

[9] 2008/01/28 09:56:30: Resolve destination 1440: a tls 192.168.1.101 2085

[9] 2008/01/28 09:56:30: Resolve destination 1440: tls 192.168.1.101 2085

[9] 2008/01/28 09:56:30: SIP Tx tls:192.168.1.101:2085:

BYE sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe SIP/2.0

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-1b49fe166b0ea8ef6955e63f79e5cb3b;rport

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8272 BYE

Max-Forwards: 70

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

RTP-RxStat: Dur=5,Pkt=26,Oct=4576,Underun=0

RTP-TxStat: Dur=0,Pkt=19,Oct=3344

Content-Length: 0

 

 

[9] 2008/01/28 09:56:30: SIP Rx tls:192.168.1.101:2085:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.1.100:5061;branch=z9hG4bK-1b49fe166b0ea8ef6955e63f79e5cb3b;rport=5061

From: "Anonymous" <sip:anonymous@localhost;user=phone>;tag=1923129968

To: <sip:2121212121@localhost;user=phone>;tag=geqdq5whvx

Call-ID: 55966ee1@pbx

CSeq: 8272 BYE

Contact: <sip:40@192.168.1.101:2051;transport=tls;line=p1d2a5oe>;flow-id=1

User-Agent: snom300/7.1.30

RTP-RxStat: Total_Rx_Pkts=19,Rx_Pkts=19,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=28,Tx_Pkts=28,Remote_Tx_Pkts=0

Content-Length: 0

 

 

[7] 2008/01/28 09:56:30: Call 55966ee1@pbx#1923129968: Clear last request

[5] 2008/01/28 09:56:30: BYE Response: Terminate 55966ee1@pbx

 

 

 

 

Copyright © 2005-2007 pbxnsip Inc. All rights reserved. See the license agreement for more information.

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  • 4 weeks later...
russelln Newbie

russelln

Posted
  • Author
Posted

Ok...

now I have a new Black CS410 (with the latest and greatest firmware, just downloaded last night) and it will connect some of time..I have 2 PSTN lines from ATT and one has my DSL on it and the other is used for a fax line,

 

the primary number is filtered, I can call into the box or out from a SNOM 300 phone and I get connected about 50% of the time, I can call the same number and get connected and then call it again and do not get a connection, it will ring the outgoing line at least one time and then disconnect, when I call my cell phone it will show a missed call..when calling into the system it will connect and will do everything you want it to do, :D then you call in again, the phone will light up like it is going to ring and then get a disconnected message on the phone screen.... :huh:

 

I think what is going on, ;) from when the time the system captures the line and the first ring, that the line battery falls far enough that the fxo processor senses it as a disconnect pulse...it always manages to ring one time...the lines from where i work are about a mile long to the phone company, to disconnect a line from my handset, I have to hook switch the line about 3 to 6 seconds sometimes, :(

 

Is there a way to delay the CPC pulse time, I think what is going on is that when the system first captures line and the first ring pulse is sent to the called party and the phone co connects the call to the calling party that the fxo port can sense the drop in voltage and it thinks the call is dissconnected, this would also explain why I never hear busy signals, as the system switches from the called line to the busy tone, the drop in voltage is read as CPC pulse and dissconnects the line, it must take a little longer for MA BELL to switch on the busy signal and :D sometimes switching the ring side fast enough :o to make the call connect..just an idea as to why this is happening.. :D

 

something has to be telling the fxo port that the call is dissconnected, it alway reads a disconnect from the fxo port when it happens...so what is the CPC pulse time on these FXO ports and is there a way to adjust them... :blink:

 

thanks

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andrewgroup Newbie

andrewgroup

Posted
Posted

Tough to stay focused on the message with all the facial expressions. Problems related to POTS lines and software driven systems are common among every maker. We had some troubles a long time ago with Audio Code 4 port model 118 and the local switch company (Sprint/Embarq) made a line adjustment to improve switch detection. I believe it's a Siemens switch and he was an old timer, and said it's only been recently they've seen these problems. I think it would be worthwhile to do some line voltage measurements.

 

The ability to adjust various parameters on the FXO gateway is a must.

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