AG1
-
Posts
75 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by AG1
-
-
I started this thread almost 2 weeks ago, and the company I ordered these units from (Telephony Ware) will not answer their phone or assist in getting these replaced. I have numerous phone calls into them as well as emails and NOTHING.
LOTS of NOTHING going on around Snom it seems, which is why I am migrating away from their phones and my next move today is to start researching new PBXs. Gotta get something I can get help on since I am not a programmer like most of you guys.
-
None of that makes any sense.
If you click on the link you provided it takes me to yet another firmware update that I can place in the update software window, click on load and NOTHING.
Then if I click the "recovery update description" it tells me how to do it on a phone with buttons. My Snom PA1 has no buttons so once again another couple hours lost while going in a circle.
-
Bump
-
Thanks for the input on apps Stefano and Carlos
I tried out the Bria app and it worked great, but now i will check out the 3CX one as well.
Thanks again for the feedback
-
They looked factory sealed and the MAC addresses were all close to the same. Like I said a couple of them worked flawlessly and the other 8 suck. Still nothing from Snom on emails I sent or posts on here.
I still have the versions I can at least talk to with versions 8.4.23 and one other version and those PA1s will not let me upgrade the firmware, you click load and the screen refreshes and nothing.
Very frustrating when things are put into the market place and fixed afterwards.
-
Does anyone know of an app or Snom download that I could install on an iPad to make it an extension of my Snom One PBX?
Basically it would be a wireless device I could pair a blue tooth headset to?
Thanks
-
I ordered 10 of the Snom PA1 units for 2 separate paging projects and I have spent the last 18 hours wondering why I bought this product.
3 of them are completely unresponsive, I cannot reset them using any of the steps on the snom wiki or other internet sights, they just bricked out of the box.
2 of them came up and worked like they should have, and I easily upgraded the firm ware to the 8.7.3.25 version they work just fine and I never had a problem with them.
The other 5 are in various states of disarray. Most of them have different versions of firmware, some get an IP through DHCP some dont, some let me access the web interface some dont. On the couple that I can actually use the web interface they will not let me upgrade the firmware. I have loaded the 8.7.3.25bin file into the firmware upgrade location and clicked on load but nothing happens. I have tried internet explorer, chrome and firefox browsers, tried doing the factory default, tried configuring by just using a PoE switch not connected to my network. I would be surprised if anyone can come up with a scenario that I havent tried and they just will not work.
Any suggestions before I mail them back and get new ones?
This should have taken about an hour to set up half of these, instead its going on 2 work days and I still only have 2 working units. Is this normal with this product?
-
Thanks, I got it working the way I want and now the 870s work as well.
-
Yes I have 4 CO lines configured on the phone, I deleted them on one of the phones and now the phone will call out on the Nextiva trunk.
Now I have a question about BLF, the reason I had co lines configured was so the operator and 4 other people could see lines ringing into the building and answer them even though they are not in the hunt group.
If I use BLF to configure a button to monitor incoming calls would I put "3000" in the parameters box since that is my hunt group?
If not how would I make 4 buttons you can monitor and answer on the virtual buttons?
-
INVITE sip:914062094291@192.168.100.151;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1
X-Serialnumber: 0004134150E1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom870/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons-snom870
Content-Type: application/sdp
Content-Length: 517
v=0
o=root 974376235 974376235 IN IP4 192.168.100.93
s=call
c=IN IP4 192.168.100.93
t=0 0
m=audio 55204 RTP/AVP 9 0 8 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:sX/0lap2QdSW3yfqJoTZSpNIji/HGcd9jq4vyQfE
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv [8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:11: Allocating for call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:11: Could not find a trunk (3 trunks) [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 1 INVITE
Content-Length: 0
[8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Sending RTP for 57ca55528438-mg2rh2cjxzgr to 192.168.100.93:55204, codec not set yet [8] 2013/10/09 15:22:11: Incoming call: Request URI sip:914062094291@192.168.100.151;user=phone, To is <sip:914062094291@192.168.100.151;user=phone> [8] 2013/10/09 15:22:11: Call from an user 2203 [8] 2013/10/09 15:22:11: To is <sip:914062094291@192.168.100.151;user=phone>, user 0, domain 2 [8] 2013/10/09 15:22:11: From user 2203 [8] 2013/10/09 15:22:11: Set the To domain based on From user 2203@192.168.100.151 [8] 2013/10/09 15:22:11: Call state for call object 337: idle [7] 2013/10/09 15:22:11: Call port 389: Set codecs to "" preference count 3 [7] 2013/10/09 15:22:11: Skipping pattern match because CO-line is not available for trunk Nextiva [5] 2013/10/09 15:22:11: Dialplan "Standard Plan": Match 914062094291@192.168.100.151 to sip:914062094291@192.168.100.151:5066;user=phone on trunk Netborder Express [5] 2013/10/09 15:22:11: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Allocating for call port 390, SIP call id 50aa13fa@pbx [7] 2013/10/09 15:22:11: Call port 390: Set codecs to "0" preference count 2 [5] 2013/10/09 15:22:11: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Call port 390: state code from 0 to 100 [5] 2013/10/09 15:22:11: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:11: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:11: Play audio_moh/noise.wav, caching true [5] 2013/10/09 15:22:11: SIP Tx udp:192.168.100.151:5066: INVITE sip:914062094291@192.168.100.151:5066;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>
Call-ID: 50aa13fa@pbx
CSeq: 29245 INVITE
Max-Forwards: 70
Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Privacy: id
Content-Type: application/sdp
Content-Length: 243
v=0
o=- 154740586 154740586 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 60494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/10/09 15:22:11: SIP Rx udp:192.168.100.151:5066: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29245 INVITE
Content-Length: 0
Server: Netborder Express Gateway/4.3.13
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
[8] 2013/10/09 15:22:11: Call port 389: state code from 0 to 183 [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 6 (mapped to 9) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 3 (mapped to 8) [8] 2013/10/09 15:22:11: Call port 389: Added predefined codec 7 (mapped to 18) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 5 (mapped to 99) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 10 (mapped to 108) [8] 2013/10/09 15:22:11: Call port 389: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:11: Set packet length to 20 [6] 2013/10/09 15:22:11: Call-leg 389: Codec PCMU/8000 is chosen for call id 57ca55528438-mg2rh2cjxzgr [5] 2013/10/09 15:22:11: set codec: codec PCMU/8000 is set to call-leg 389 [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 1 INVITE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 386
v=0
o=- 639170658 639170658 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 49472 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/10/09 15:22:11: SIP Rx tls:192.168.100.93:3372: PRACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/10/09 15:22:11: Packet authenticated by transport layer [5] 2013/10/09 15:22:11: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-az5g906p0x93;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 2 PRACK
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[8] 2013/10/09 15:22:11: Packet authenticated by transport layer [8] 2013/10/09 15:22:12: Last message repeated 3 times [8] 2013/10/09 15:22:12: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:13: SMTP: Received 220 mx.google.com ESMTP v22sm64056817yhn.12 - gsmtp [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [8] 2013/10/09 15:22:13: Packet authenticated by transport layer [8] 2013/10/09 15:22:13: SMTP: Received 250-mx.google.com at your service, [216.228.51.194]
250-SIZE 35882577
250-8BITMIME
250-STARTTLS
250-ENHANCEDSTATUSCODES
250 CHUNKING [8] 2013/10/09 15:22:13: SMTP: Send STARTTLS [8] 2013/10/09 15:22:13: SMTP: Received 220 2.0.0 Ready to start TLS [8] 2013/10/09 15:22:13: SMTP: Send EHLO localhost [4] 2013/10/09 15:22:13: Certificate for Equifax Secure Certificate Authority not available [5] 2013/10/09 15:22:13: Certificate for smtp.gmail.com could not be verified against [8] 2013/10/09 15:22:13: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:13: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:13: Trunk 4: Preparing for re-registration [8] 2013/10/09 15:22:13: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/10/09 15:22:13: Trunk 4: setup callback to send re-registration after 38 seconds [5] 2013/10/09 15:22:13: SMTP: Connection to 74.125.140.108:25 failed [8] 2013/10/09 15:22:14: Packet authenticated by transport layer [8] 2013/10/09 15:22:16: Last message repeated 4 times [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29245 INVITE
Content-Length: 238
Content-Type: application/sdp
Server: Netborder Express Gateway/4.3.13
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
v=0
o=Sangoma-Tech 1381353736 1381353785 IN IP4 192.168.100.151
s=SIP Call
c=IN IP4 192.168.100.151
t=0 0
m=audio 14574 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [6] 2013/10/09 15:22:16: Call-leg 390: Codec PCMU/8000 is chosen for call id 50aa13fa@pbx [6] 2013/10/09 15:22:16: Call-leg 390: Sending RTP for 50aa13fa@pbx to 192.168.100.151:14574, codec PCMU/8000 [5] 2013/10/09 15:22:16: set codec: codec PCMU/8000 is set to call-leg 390 [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call state for call object 337: alerting [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 183 [8] 2013/10/09 15:22:16: Last message repeated 2 times [7] 2013/10/09 15:22:16: 57ca55528438-mg2rh2cjxzgr: RTP pass-through mode [7] 2013/10/09 15:22:16: 50aa13fa@pbx: RTP pass-through mode [5] 2013/10/09 15:22:16: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-735047f4d4c642fb68012d418c450f00;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29245 INVITE
Content-Length: 238
Content-Type: application/sdp
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
Server: Netborder Express Gateway/4.3.13
v=0
o=Sangoma-Tech 1381353736 1381353786 IN IP4 192.168.100.151
s=SIP Call
c=IN IP4 192.168.100.151
t=0 0
m=audio 14574 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv [7] 2013/10/09 15:22:16: Call 50aa13fa@pbx: Clear last INVITE [8] 2013/10/09 15:22:16: Call port 390: Added predefined codec 2 (mapped to 0) [8] 2013/10/09 15:22:16: Call port 390: Added rtpmap codec 1 (mapped to 101) [7] 2013/10/09 15:22:16: Set packet length to 20 [5] 2013/10/09 15:22:16: SIP Tx udp:192.168.100.151:5066: ACK sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-b0002f2d1f9910b246232d9f43d1c0b0;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29245 ACK
Max-Forwards: 70
Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Privacy: id
Content-Length: 0
[7] 2013/10/09 15:22:16: Determine pass-through mode after receiving response [8] 2013/10/09 15:22:16: Call state for call object 337: connected [5] 2013/10/09 15:22:16: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/10/09 15:22:16: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [8] 2013/10/09 15:22:16: Call port 390: state code from 100 to 200 [8] 2013/10/09 15:22:16: Call port 389: state code from 183 to 200 [5] 2013/10/09 15:22:16: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-utv4lgeoggk3;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 1 INVITE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Content-Type: application/sdp
Content-Length: 386
v=0
o=- 639170658 639170658 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 49472 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+mUYvsDtufVA2IS4U3Q3DZB12+ZzLjFbdKTFyIiv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/10/09 15:22:16: SIP Rx tls:192.168.100.93:3372: ACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-3ojncdnhtw1j;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/10/09 15:22:16: Packet authenticated by transport layer [8] 2013/10/09 15:22:18: Last message repeated 6 times [8] 2013/10/09 15:22:18: SMTP: Connect to 74.125.140.108:25 [8] 2013/10/09 15:22:19: SMTP: Received 421 4.4.5 Server busy, try again later. (mx.google.com) s21sm64052493yhk.9 - gsmtp [5] 2013/10/09 15:22:19: SMTP Server returned 421 [8] 2013/10/09 15:22:19: Packet authenticated by transport layer [8] 2013/10/09 15:22:22: Last message repeated 9 times [8] 2013/10/09 15:22:22: rtp_hangup: call port 387, too early to disconnect [8] 2013/10/09 15:22:22: rtp_hangup: call port 388, too early to disconnect [8] 2013/10/09 15:22:23: Packet authenticated by transport layer [8] 2013/10/09 15:22:26: Last message repeated 6 times [8] 2013/10/09 15:22:26: Play audio_en/aa_no_answer.wav space10 audio_en/aa_receive_callback.wav audio_en/aa_leave_message.wav audio_en/aa_offer_cellphone.wav space50, caching false [8] 2013/10/09 15:22:26: Call port 379: state code from 200 to 200 [8] 2013/10/09 15:22:29: Packet authenticated by transport layer [8] 2013/10/09 15:22:35: Last message repeated 12 times [5] 2013/10/09 15:22:35: Identify trunk (IP address/port and domain match) 3 [8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Rx tls:192.168.100.93:3372: BYE sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=27lf1wbz>;reg-id=1
User-Agent: snom870/8.7.3.19
RTP-RxStat: Total_Rx_Pkts=1242,Rx_Pkts=1227,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=1229,Tx_Pkts=1229,Remote_Tx_Pkts=6
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/10/09 15:22:36: Packet authenticated by transport layer [5] 2013/10/09 15:22:36: SIP Tx tls:192.168.100.93:3372: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-u7zgtvycra3f;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=ngf4lmqvqg
To: <sip:914062094291@192.168.100.151;user=phone>;tag=14dc02e81d
Call-ID: 57ca55528438-mg2rh2cjxzgr
CSeq: 3 BYE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[7] 2013/10/09 15:22:36: 50aa13fa@pbx: Media-aware pass-through mode [8] 2013/10/09 15:22:36: Clearing call port 389, SIP call id 57ca55528438-mg2rh2cjxzgr [8] 2013/10/09 15:22:36: Call port 390: state code from 200 to 486 [5] 2013/10/09 15:22:36: SIP Tx udp:192.168.100.151:5066: BYE sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29246 BYE
Max-Forwards: 70
Contact: <sip:anonymous@192.168.100.151:5060;transport=udp>
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Privacy: id
Content-Length: 0
[5] 2013/10/09 15:22:36: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-3f99471547e748b3382e3a011de7b126;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=2073723721
To: <sip:914062094291@192.168.100.151;user=phone>;tag=ds-515242e6-e9eac574
Call-ID: 50aa13fa@pbx
CSeq: 29246 BYE
Content-Length: 0
-
What do you mean send an invite from the pbx?
-
I have a Nextiva SIP trunk, I set my dial plan to dial that trunk (9*) when 9 is pushed then the number
This works on the Snom 821, Snom 720 and Grandstream GXP 2200 I can select the Nextiva trunk to dial out.
I also have a Sangoma NBE gateway that is the default if a 9 is not dialed. You can dial out on the 821,720, 2200 and 870 phones I have in the building.
The 870 phones will not dial out on the Nextiva trunk when you dial 9 then the number. The message that comes back is "you must first dial a 1 when calling this number" 91406209xxxx is what I am dialing
I have one standard dial plan set up in my PBX
101- Nextiva - 9*
102 - NBE - 911
103 - NBE - *
104 - NBE - 411
105 - NBE - 811
106 - NBE - ^([2-9][0-9]{6})@.*
107 - NBE - ^([0-9]{10})@.*
108 - NBE - ^(1[0-9]{10})@.*
All of the phones are assigned to this dial plan but the 870s will not dial out on my Nextiva trunk
Any ideas.......................I am probably just going to sell the 5 - 870 phones and get the GXP2200 instead since this is phone related.
-
Nevermind, when the PBX rebooted it came back up with a number in a field there shouldnt have been a number, I deleted the number now it registers all phones
-
I have rebooted each phone as well as the PBX and each time the PBX reboot only 2 phones register.
Every time it is a different 2 phones.
Any ideas
-
I know it starts with 90xx?
More problems........again
-
And here is my 870 after I factory defaulted the settings and updated the firmware.................still wont dial out
INVITE sip:92094291@192.168.100.151:5066;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-bcc0c30ad0bfe60618252725cb88ac39;rport
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>
Call-ID: 3779be92@pbx
CSeq: 3674 INVITE
Max-Forwards: 70
Contact: <sip:4064888066@192.168.100.151:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 1151548849 1151548849 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 56400 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/27 22:52:00: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:00: Last message repeated 3 times [5] 2013/08/27 22:52:00: SIP Rx udp:192.168.100.151:5066: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-bcc0c30ad0bfe60618252725cb88ac39;rport=5060
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3674 INVITE
Content-Length: 0
Server: Netborder Express Gateway/4.3.13
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
[8] 2013/08/27 22:52:00: Play audio_moh/noise.wav, caching true [8] 2013/08/27 22:52:00: Call port 3: state code from 0 to 183 [8] 2013/08/27 22:52:00: Call port 3: Added predefined codec 6 (mapped to 9) [8] 2013/08/27 22:52:00: Call port 3: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:52:00: Call port 3: Added predefined codec 3 (mapped to 8) [8] 2013/08/27 22:52:00: Call port 3: Added predefined codec 7 (mapped to 18) [8] 2013/08/27 22:52:00: Call port 3: Added rtpmap codec 5 (mapped to 99) [8] 2013/08/27 22:52:00: Call port 3: Added rtpmap codec 10 (mapped to 108) [8] 2013/08/27 22:52:00: Call port 3: Added rtpmap codec 1 (mapped to 101) [7] 2013/08/27 22:52:00: Set packet length to 20 [6] 2013/08/27 22:52:00: Call-leg 3: Codec PCMU/8000 is chosen for call id d6821d52e46c-tbnonianfi7r [5] 2013/08/27 22:52:00: set codec: codec PCMU/8000 is set to call-leg 3 [5] 2013/08/27 22:52:00: SIP Tx tls:192.168.100.93:3357: SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-lsfnbuka903g;rport=3357
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 1 INVITE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 388
v=0
o=- 1824430072 1824430072 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 61302 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:soahnjFGvd/tLKUAMFLy9fNp5YqVa5lHLuR/3yZk
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [6] 2013/08/27 22:52:00: Received bindRequest for user 192.168.100.151\2203 [5] 2013/08/27 22:52:00: SIP Rx tls:192.168.100.93:3357: PRACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-okttveypq9al;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3357;transport=tls;line=q2hecvx1>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/08/27 22:52:00: Packet authenticated by transport layer [5] 2013/08/27 22:52:00: SIP Tx tls:192.168.100.93:3357: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-okttveypq9al;rport=3357
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 2 PRACK
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[8] 2013/08/27 22:52:00: Packet authenticated by transport layer [8] 2013/08/27 22:52:04: Last message repeated 5 times [5] 2013/08/27 22:52:04: SIP Rx udp:192.168.100.151:5066: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-bcc0c30ad0bfe60618252725cb88ac39;rport=5060
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3674 INVITE
Content-Length: 238
Content-Type: application/sdp
Server: Netborder Express Gateway/4.3.13
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
v=0
o=Sangoma-Tech 1377665524 1377665573 IN IP4 192.168.100.151
s=SIP Call
c=IN IP4 192.168.100.151
t=0 0
m=audio 18430 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv [8] 2013/08/27 22:52:04: Call port 4: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:52:04: Call port 4: Added rtpmap codec 1 (mapped to 101) [7] 2013/08/27 22:52:04: Set packet length to 20 [5] 2013/08/27 22:52:04: set codec: codec PCMU/8000 is set to call-leg 4 [6] 2013/08/27 22:52:04: Call-leg 4: Codec PCMU/8000 is chosen for call id 3779be92@pbx [6] 2013/08/27 22:52:04: Call-leg 4: Sending RTP for 3779be92@pbx to 192.168.100.151:18430, codec PCMU/8000 [5] 2013/08/27 22:52:04: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Last message repeated 3 times [8] 2013/08/27 22:52:04: Call state for call object 2852: alerting [5] 2013/08/27 22:52:04: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-bcc0c30ad0bfe60618252725cb88ac39;rport=5060
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3674 INVITE
Content-Length: 238
Content-Type: application/sdp
Contact: <sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp>
Server: Netborder Express Gateway/4.3.13
v=0
o=Sangoma-Tech 1377665524 1377665574 IN IP4 192.168.100.151
s=SIP Call
c=IN IP4 192.168.100.151
t=0 0
m=audio 18430 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv [7] 2013/08/27 22:52:04: Call 3779be92@pbx: Clear last INVITE [8] 2013/08/27 22:52:04: Call port 4: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:52:04: Call port 4: Added rtpmap codec 1 (mapped to 101) [7] 2013/08/27 22:52:04: Set packet length to 20 [5] 2013/08/27 22:52:04: SIP Tx udp:192.168.100.151:5066: ACK sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-140cfac8e3f676984577ddae2f3153eb;rport
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3674 ACK
Max-Forwards: 70
Contact: <sip:4064888066@192.168.100.151:5060;transport=udp>
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Content-Length: 0
[7] 2013/08/27 22:52:04: Determine pass-through mode after receiving response [8] 2013/08/27 22:52:04: Call state for call object 2852: connected [5] 2013/08/27 22:52:04: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:04: Last message repeated 3 times [8] 2013/08/27 22:52:04: Call port 3: state code from 183 to 183 [8] 2013/08/27 22:52:04: Last message repeated 2 times [7] 2013/08/27 22:52:04: d6821d52e46c-tbnonianfi7r: RTP pass-through mode [7] 2013/08/27 22:52:04: 3779be92@pbx: RTP pass-through mode [8] 2013/08/27 22:52:04: Call port 4: state code from 100 to 200 [8] 2013/08/27 22:52:04: Call port 3: state code from 183 to 200 [5] 2013/08/27 22:52:04: SIP Tx tls:192.168.100.93:3357: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-lsfnbuka903g;rport=3357
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 1 INVITE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Content-Type: application/sdp
Content-Length: 388
v=0
o=- 1824430072 1824430072 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 61302 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:soahnjFGvd/tLKUAMFLy9fNp5YqVa5lHLuR/3yZk
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/27 22:52:04: SIP Rx tls:192.168.100.93:3357: ACK sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-o6dmuz7zb7n6;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3357;transport=tls;line=q2hecvx1>;reg-id=1
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/08/27 22:52:04: Packet authenticated by transport layer [8] 2013/08/27 22:52:17: Last message repeated 24 times [5] 2013/08/27 22:52:17: SIP Rx tls:192.168.100.93:3357: BYE sip:2203@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-l7hvskv0o8dj;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3357;transport=tls;line=q2hecvx1>;reg-id=1
User-Agent: snom870/8.7.3.19
RTP-RxStat: Total_Rx_Pkts=834,Rx_Pkts=818,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=820,Tx_Pkts=820,Remote_Tx_Pkts=6
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/08/27 22:52:17: Packet authenticated by transport layer [5] 2013/08/27 22:52:17: SIP Tx tls:192.168.100.93:3357: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-l7hvskv0o8dj;rport=3357
From: "Conference Room" <sip:2203@192.168.100.151>;tag=vlnn8geka9
To: <sip:92094291@192.168.100.151;user=phone>;tag=c18e1b3c21
Call-ID: d6821d52e46c-tbnonianfi7r
CSeq: 3 BYE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[7] 2013/08/27 22:52:17: 3779be92@pbx: Media-aware pass-through mode [8] 2013/08/27 22:52:17: Clearing call port 3, SIP call id d6821d52e46c-tbnonianfi7r [8] 2013/08/27 22:52:17: Call port 4: state code from 200 to 486 [5] 2013/08/27 22:52:17: SIP Tx udp:192.168.100.151:5066: BYE sip:NetborderExpressGateway@192.168.100.151:5066;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-ca1f4ba5d42ad0b8c624d72eaf833428;rport
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3675 BYE
Max-Forwards: 70
Contact: <sip:4064888066@192.168.100.151:5060;transport=udp>
P-Asserted-Identity: "Netborder Express" <sip:192.168.100.151:5066>
Content-Length: 0
[5] 2013/08/27 22:52:17: SIP Rx udp:192.168.100.151:5066: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-ca1f4ba5d42ad0b8c624d72eaf833428;rport=5060
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=1920819945
To: <sip:92094291@192.168.100.151;user=phone>;tag=ds-6d611217-ffdc69a0
Call-ID: 3779be92@pbx
CSeq: 3675 BYE
Content-Length: 0
[7] 2013/08/27 22:52:17: Call 3779be92@pbx: Clear last request [5] 2013/08/27 22:52:17: BYE Response: Terminate 3779be92@pbx [8] 2013/08/27 22:52:17: Clearing call port 4, SIP call id 3779be92@pbx [8] 2013/08/27 22:52:17: Remove leg 7677: Call port 3, SIP call id d6821d52e46c-tbnonianfi7r [8] 2013/08/27 22:52:17: Hangup: Call 3 not found [8] 2013/08/27 22:52:17: Last message repeated 2 times [8] 2013/08/27 22:52:17: Remove leg 7678: Call port 4, SIP call id 3779be92@pbx [5] 2013/08/27 22:52:17: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:17: Last message repeated 3 times [8] 2013/08/27 22:52:17: Packet authenticated by transport layer [8] 2013/08/27 22:52:21: Last message repeated 8 times [8] 2013/08/27 22:52:21: Could not find a trunk (3 trunks) [5] 2013/08/27 22:52:21: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Last message repeated 3 times [5] 2013/08/27 22:52:21: Sending IM from "Lee Candee" <sip:2012@192.168.100.151> to "Lee Candee" <sip:2012@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Stacy Kober" <sip:2006@192.168.100.151> to "Stacy Kober" <sip:2006@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Conference Room" <sip:2203@192.168.100.151> to "Conference Room" <sip:2203@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Lynette Hoon" <sip:2018@192.168.100.151> to "Lynette Hoon" <sip:2018@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Sherri Dardis" <sip:2014@192.168.100.151> to "Sherri Dardis" <sip:2014@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:21: Sending IM from "Brandon Roth" <sip:2021@192.168.100.151> to "Brandon Roth" <sip:2021@192.168.100.151> (1 destinations) [5] 2013/08/27 22:52:22: Last message repeated 3 times [8] 2013/08/27 22:52:22: Packet authenticated by transport layer [8] 2013/08/27 22:52:30: Last message repeated 11 times [5] 2013/08/27 22:52:30: SIP Rx tls:192.168.100.93:3357: INVITE sip:914062094291@192.168.100.151;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.100.93:3357;branch=z9hG4bK-z9b5y24tonei;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=lf3usgd0gk
To: <sip:914062094291@192.168.100.151;user=phone>
Call-ID: f3821d527f58-ufjcqksv0ktm
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3357;transport=tls;line=q2hecvx1>;reg-id=1
X-Serialnumber: 0004134150E1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom870/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons-snom870
Content-Type: application/sdp
Content-Length: 519 -
INVITE sip:92094291@192.168.100.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK1422683647;rport
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 10 INVITE
Contact: "Grandstream 2200" <sip:2215@192.168.100.165:38590>
Max-Forwards: 70
User-Agent: Grandstream GXP2200 1.0.3.6
Privacy: none
P-Preferred-Identity: "Grandstream 2200" <sip:2215@192.168.100.151>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 313
v=0
o=2215 8000 8000 IN IP4 192.168.100.165
s=SIP Call
c=IN IP4 192.168.100.165
t=0 0
m=audio 56736 RTP/AVP 0 8 9 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 [8] 2013/08/27 22:22:40: Allocating for call port 1, SIP call id 453083683-38590-2@BJC.BGI.BAA.BGF [8] 2013/08/27 22:22:40: Could not find a trunk (3 trunks) [5] 2013/08/27 22:22:40: SIP Tx udp:192.168.100.165:38590: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK1422683647;rport=38590
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 10 INVITE
Content-Length: 0
[5] 2013/08/27 22:22:40: SIP Tx udp:192.168.100.165:38590: SIP/2.0 401 Authentication Required
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK1422683647;rport=38590
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 10 INVITE
User-Agent: snomONE/5.0.10
WWW-Authenticate: Digest realm="192.168.100.151",nonce="a2422a81b1f0ee97116921d80bcf2d9c",domain="sip:92094291@192.168.100.151",algorithm=MD5
Content-Length: 0
[5] 2013/08/27 22:22:40: SIP Rx udp:192.168.100.165:38590: ACK sip:92094291@192.168.100.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK1422683647;rport
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 10 ACK
Content-Length: 0
[5] 2013/08/27 22:22:40: SIP Rx udp:192.168.100.165:38590: INVITE sip:92094291@192.168.100.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK227563974;rport
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 11 INVITE
Contact: "Grandstream 2200" <sip:2215@192.168.100.165:38590>
Authorization: Digest username="2215", realm="192.168.100.151", nonce="a2422a81b1f0ee97116921d80bcf2d9c", uri="sip:92094291@192.168.100.151", response="9ed11666eaaba10f5b4eb1e59388ccba", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP2200 1.0.3.6
Privacy: none
P-Preferred-Identity: "Grandstream 2200" <sip:2215@192.168.100.151>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 313
v=0
o=2215 8000 8000 IN IP4 192.168.100.165
s=SIP Call
c=IN IP4 192.168.100.165
t=0 0
m=audio 56736 RTP/AVP 0 8 9 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 [8] 2013/08/27 22:22:40: Tagging request with existing tag [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 3 (mapped to 8) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 6 (mapped to 9) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 7 (mapped to 18) [7] 2013/08/27 22:22:40: Set packet length to 20 [8] 2013/08/27 22:22:40: Call port 1: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/27 22:22:40: Call-leg 1: Sending RTP for 453083683-38590-2@BJC.BGI.BAA.BGF to 192.168.100.165:56736, codec not set yet [8] 2013/08/27 22:22:40: Incoming call: Request URI sip:92094291@192.168.100.151, To is <sip:92094291@192.168.100.151> [8] 2013/08/27 22:22:40: Call from an user 2215 [5] 2013/08/27 22:22:40: SIP Tx udp:192.168.100.165:38590: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK227563974;rport=38590
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 11 INVITE
Content-Length: 0
[8] 2013/08/27 22:22:40: To is <sip:92094291@192.168.100.151>, user 0, domain 2 [8] 2013/08/27 22:22:40: From user 2215 [8] 2013/08/27 22:22:40: Set the To domain based on From user 2215@192.168.100.151 [8] 2013/08/27 22:22:40: Call state for call object 2851: idle [7] 2013/08/27 22:22:40: Call port 1: Set codecs to "" preference count 3 [5] 2013/08/27 22:22:40: Dialplan "Standard Plan": Match 92094291@192.168.100.151 to sip:2094291@208.73.146.95;user=phone on trunk Nextiva [8] 2013/08/27 22:22:40: Allocating for call port 2, SIP call id 5063b48b@pbx [8] 2013/08/27 22:22:40: Play audio_moh/noise.wav, caching true [7] 2013/08/27 22:22:40: Call port 2: Set codecs to "0 18" preference count 3 [8] 2013/08/27 22:22:40: Call port 2: state code from 0 to 100 [5] 2013/08/27 22:22:40: SIP Tx udp:208.73.146.95:5060: INVITE sip:2094291@208.73.146.95;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-59f259ad8a1b7a80bd55788dc92791d6;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5078 INVITE
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Remote-Party-ID: "Grandstream 2200" <sip:4064888066@192.168.100.151;user=phone>
Content-Type: application/sdp
Content-Length: 292
v=0
o=- 1836188768 1836188768 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 58528 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/27 22:22:40: Call port 1: state code from 0 to 183 [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 3 (mapped to 8) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 6 (mapped to 9) [8] 2013/08/27 22:22:40: Call port 1: Added predefined codec 7 (mapped to 18) [7] 2013/08/27 22:22:40: Set packet length to 20 [8] 2013/08/27 22:22:40: Call port 1: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/27 22:22:40: Call-leg 1: Codec PCMU/8000 is chosen for call id 453083683-38590-2@BJC.BGI.BAA.BGF [5] 2013/08/27 22:22:40: set codec: codec PCMU/8000 is set to call-leg 1 [5] 2013/08/27 22:22:40: SIP Tx udp:192.168.100.165:38590: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK227563974;rport=38590
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 11 INVITE
Contact: <sip:2215@192.168.100.151:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Content-Type: application/sdp
Content-Length: 302
v=0
o=- 382789709 382789709 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 50778 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/27 22:22:41: SIP Rx udp:208.73.146.95:5060: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-59f259ad8a1b7a80bd55788dc92791d6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5078 INVITE
[5] 2013/08/27 22:22:41: SIP Rx udp:208.73.146.95:5060: SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-59f259ad8a1b7a80bd55788dc92791d6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5078 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="voip.nextiva.com", nonce="f0969170571ef36b084ade288fb1c3f0"
[8] 2013/08/27 22:22:41: Answer challenge with username 14062094291 [5] 2013/08/27 22:22:41: SIP Tx udp:208.73.146.95:5060: INVITE sip:2094291@208.73.146.95;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-4de04c9d347c51e5d5713a77fc6272b6;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5079 INVITE
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Remote-Party-ID: "Grandstream 2200" <sip:4064888066@192.168.100.151;user=phone>
Proxy-Authorization: Digest realm="voip.nextiva.com",nonce="f0969170571ef36b084ade288fb1c3f0",response="bb7a43737a41ca587d9fc5354c12234c",username="14062094291",uri="sip:2094291@208.73.146.95;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 292
v=0
o=- 1836188768 1836188768 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 58528 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/27 22:22:41: SIP Rx udp:208.73.146.95:5060: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4de04c9d347c51e5d5713a77fc6272b6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5079 INVITE
[8] 2013/08/27 22:22:41: Packet authenticated by transport layer [5] 2013/08/27 22:22:42: SIP Rx udp:208.73.146.95:5060: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4de04c9d347c51e5d5713a77fc6272b6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5079 INVITE
Content-Length: 240
Content-Type: application/sdp
v=0
o=msc3 24252 7300 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 32570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/27 22:22:42: Call port 2: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:22:42: Call port 2: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/27 22:22:42: Call-leg 2: Codec PCMU/8000 is chosen for call id 5063b48b@pbx [6] 2013/08/27 22:22:42: Call-leg 2: Sending RTP for 5063b48b@pbx to 208.73.146.95:32570, codec PCMU/8000 [5] 2013/08/27 22:22:42: set codec: codec PCMU/8000 is set to call-leg 2 [8] 2013/08/27 22:22:42: Call state for call object 2851: alerting [6] 2013/08/27 22:22:42: Trunk Nextiva: Ignoring the SDP due to the trunk setting [8] 2013/08/27 22:22:42: Play audio_en/ringback.wav, caching true [8] 2013/08/27 22:22:42: Call port 1: state code from 183 to 183 [5] 2013/08/27 22:22:42: SIP Rx udp:208.73.146.95:5060: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4de04c9d347c51e5d5713a77fc6272b6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>
Call-ID: 5063b48b@pbx
CSeq: 5079 INVITE
Content-Length: 240
Content-Type: application/sdp
v=0
o=msc3 24252 7300 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 32570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/27 22:22:42: Call port 2: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:22:42: Call port 2: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/27 22:22:42: Trunk Nextiva: Ignoring the SDP due to the trunk setting [5] 2013/08/27 22:22:42: SIP Rx udp:208.73.146.95:5060: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4de04c9d347c51e5d5713a77fc6272b6;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>;tag=3586652800-492860
Call-ID: 5063b48b@pbx
CSeq: 5079 INVITE
Contact: <sip:2094291@208.73.146.95:5060;transport=udp>
Content-Length: 240
Content-Type: application/sdp
v=0
o=msc3 24252 7300 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 32570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [7] 2013/08/27 22:22:42: Call 5063b48b@pbx: Clear last INVITE [8] 2013/08/27 22:22:42: Call port 2: Added predefined codec 2 (mapped to 0) [8] 2013/08/27 22:22:42: Call port 2: Added rtpmap codec 1 (mapped to 101) [5] 2013/08/27 22:22:42: SIP Tx udp:208.73.146.95:5060: ACK sip:2094291@208.73.146.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-164e380a5da82641b1e0a16a28fb0789;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1726530904
To: <sip:2094291@208.73.146.95>;tag=3586652800-492860
Call-ID: 5063b48b@pbx
CSeq: 5079 ACK
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Remote-Party-ID: "Grandstream 2200" <sip:4064888066@192.168.100.151;user=phone>
Content-Length: 0
[7] 2013/08/27 22:22:42: Determine pass-through mode after receiving response [8] 2013/08/27 22:22:42: Call state for call object 2851: connected [8] 2013/08/27 22:22:42: Call port 2: state code from 100 to 200 [8] 2013/08/27 22:22:42: Call port 1: state code from 183 to 200 [5] 2013/08/27 22:22:42: SIP Tx udp:192.168.100.165:38590: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK227563974;rport=38590
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 11 INVITE
Contact: <sip:2215@192.168.100.151:5060>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Content-Type: application/sdp
Content-Length: 302
v=0
o=- 382789709 382789709 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 50778 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [7] 2013/08/27 22:22:42: 453083683-38590-2@BJC.BGI.BAA.BGF: RTP pass-through mode [7] 2013/08/27 22:22:42: 5063b48b@pbx: RTP pass-through mode [5] 2013/08/27 22:22:42: SIP Rx udp:192.168.100.165:38590: ACK sip:2215@192.168.100.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.165:38590;branch=z9hG4bK1272942014;rport
From: "Grandstream 2200" <sip:2215@192.168.100.151>;tag=1790540350
To: <sip:92094291@192.168.100.151>;tag=ab387c5463
Call-ID: 453083683-38590-2@BJC.BGI.BAA.BGF
CSeq: 11 ACK
Contact: <sip:2215@192.168.100.165:38590>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXP2200 1.0.3.6
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
[5] 2013/08/27 22:22:42: Tuning to new SSRC from 192.168.100.165:56736 [8] 2013/08/27 22:22:43: Packet authenticated by transport layer [8] 2013/08/27 22:22:49: Last message repeated 5 times [8] 2013/08/27 22:22:49: Trunk 4: Preparing for re-registration [8] 2013/08/27 22:22:49: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/27 22:22:49: Trunk 4: setup callback to send re-registration after 38 seconds [8] 2013/08/27 22:22:50: Packet authenticated by transport layer [5] 2013/08/27 22:22:51: SIP Rx udp:192.168.100.165:38590:
-
I just added a Grandstream GXP 2200 that now calls out dialing 9 to reach my VoIp trunk (Nextiva)
Do you still think its the trunk when the 820 and now Grandstream Gxp 2200 work on the same dial plan?
Its the 870 phones......why?
-
OK then how do I run a Nextiva trunk and analog lines at the same time if I am not able to select IP or analog?
I dont have the option to run strictly VoIP because our internet sucks. How would I set it up where I just have to dial the "real number"?
This all worked the way I had it set up for about 5 months, then it stopped working with the 870s. I am failing to see how it isnt the 870 phone itself, can you explain why the 820s work flawlessly?
-
I have one standard dial plan both phones are running that dial plan.
I have it set up to dial 9 to use the Nextiva trunk all other calls use NetBorder Express. 870s wont dial out on the Nextiva trunk
-
Here is the 870 log. It says I need to dial a 1 before dialing.....as you can see..........I did.
INVITE sip:914062094291@192.168.100.151;user=phone SIP/2.0Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-0y9dr3iz29r1;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=b9019ejn2f
To: <sip:914062094291@192.168.100.151;user=phone>
Call-ID: 2f6c165214c3-2j0z19hcqcc9
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=zl1csrsm>;reg-id=1
X-Serialnumber: 0004134150E1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom870/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons-snom870
Content-Type: application/sdp
Content-Length: 519
v=0
o=root 1921552477 1921552477 IN IP4 192.168.100.93
s=call
c=IN IP4 192.168.100.93
t=0 0
m=audio 62958 RTP/AVP 9 0 8 99 108 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:uzt4Tjw+9mLEoIVj1o0dejxA1w83WikY3yUvSzYY
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[8] 2013/08/22 13:49:59: Packet authenticated by transport layer [8] 2013/08/22 13:49:59: Allocating for call port 272, SIP call id 2f6c165214c3-2j0z19hcqcc9 [8] 2013/08/22 13:49:59: Could not find a trunk (3 trunks) [5] 2013/08/22 13:49:59: SIP Tx tls:192.168.100.93:3372: SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.100.93:3372;branch=z9hG4bK-0y9dr3iz29r1;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=b9019ejn2f
To: <sip:914062094291@192.168.100.151;user=phone>;tag=9c536ddac2
Call-ID: 2f6c165214c3-2j0z19hcqcc9
CSeq: 1 INVITE
Content-Length: 0
-
It sends the call out on the Nextiva trunk using the 821 phones.
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-94w78l2wmf5w;rport=3238
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 1 INVITE
Contact: <sip:2202@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 388
v=0
o=- 2146324864 2146324864 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 61612 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:L2eH2LImWvqRubppwAhzLE2Hdiymzoq9DxiTr6XM
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/22 13:37:18: SIP Rx tls:192.168.100.111:3238: PRACK sip:2202@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-8s7focb4mdt5;rport
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:2202@192.168.100.111:3238;transport=tls;line=4ofyd87d>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons-snom821
Content-Length: 0
[8] 2013/08/22 13:37:18: Packet authenticated by transport layer [5] 2013/08/22 13:37:18: SIP Tx tls:192.168.100.111:3238: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-8s7focb4mdt5;rport=3238
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 2 PRACK
Contact: <sip:2202@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[6] 2013/08/22 13:37:18: Received bindRequest for user 192.168.100.151\2202 [5] 2013/08/22 13:37:18: SIP Rx udp:208.73.146.95:5060: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-8333a433660cee2b3b67b0dc5b4605a2;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32163 INVITE
[5] 2013/08/22 13:37:18: SIP Rx udp:208.73.146.95:5060: SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-8333a433660cee2b3b67b0dc5b4605a2;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32163 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="voip.nextiva.com", nonce="a41124cc1dd80ffc14006a0de89c94c0"
[8] 2013/08/22 13:37:18: Answer challenge with username 14062094291 [5] 2013/08/22 13:37:18: SIP Tx udp:208.73.146.95:5060: INVITE sip:14062094291@208.73.146.95;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Remote-Party-ID: "Anonymous" <sip:anonymous@anonymous.invalid>
Proxy-Authorization: Digest realm="voip.nextiva.com",nonce="a41124cc1dd80ffc14006a0de89c94c0",response="da9b775ea549d53271ee9cce6521982e",username="14062094291",uri="sip:14062094291@208.73.146.95;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 292
v=0
o=- 1070790178 1070790178 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 51584 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [5] 2013/08/22 13:37:18: SIP Rx udp:208.73.146.95:5060: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
[6] 2013/08/22 13:37:18: Received bindRequest for user 192.168.100.151\2202 [6] 2013/08/22 13:37:18: Last message repeated 3 times [8] 2013/08/22 13:37:18: Packet authenticated by transport layer [8] 2013/08/22 13:37:19: Last message repeated 2 times [5] 2013/08/22 13:37:19: SIP Rx udp:208.73.146.95:5060: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
Content-Length: 241
Content-Type: application/sdp
v=0
o=msc3 17199 19356 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 30742 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/22 13:37:19: Call port 254: Added predefined codec 2 (mapped to 0) [8] 2013/08/22 13:37:19: Call port 254: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/22 13:37:19: Call-leg 254: Codec PCMU/8000 is chosen for call id 6c60efb7@pbx [6] 2013/08/22 13:37:19: Call-leg 254: Sending RTP for 6c60efb7@pbx to 208.73.146.95:30742, codec PCMU/8000 [5] 2013/08/22 13:37:19: set codec: codec PCMU/8000 is set to call-leg 254 [8] 2013/08/22 13:37:19: Call state for call object 2202: alerting [6] 2013/08/22 13:37:19: Trunk Nextiva: Ignoring the SDP due to the trunk setting [8] 2013/08/22 13:37:19: Play audio_en/ringback.wav, caching true [8] 2013/08/22 13:37:19: Call port 253: state code from 183 to 183 [8] 2013/08/22 13:37:19: Packet authenticated by transport layer [8] 2013/08/22 13:37:30: Last message repeated 18 times [5] 2013/08/22 13:37:30: Dictionary: Item dom_logging.htm reg_logging.htm#sysadmin en not found [8] 2013/08/22 13:37:31: Packet authenticated by transport layer [8] 2013/08/22 13:37:39: Last message repeated 16 times [6] 2013/08/22 13:37:39: Websocket GET / HTTP/1.1
Upgrade: websocket
Connection: Upgrade
Host: 192.168.100.151
Origin: http://192.168.100.151
Pragma: no-cache
Cache-Control: no-cache
Sec-WebSocket-Key: 1aIG9lsZwqPRC7L8YD/syg==
Sec-WebSocket-Version: 13
Sec-WebSocket-Extensions: x-webkit-deflate-frame
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/28.0.1500.95 Safari/537.36
Cookie: ui_dom_other=block; session=2ofq3969fk1412usckma; ui_reg_gen=block; ui_reg_sip=block; ui_reg_net=block; ui_reg_sec=block; acct_table#pageNavPos=1
[8] 2013/08/22 13:37:39: Packet authenticated by transport layer [8] 2013/08/22 13:37:41: Last message repeated 2 times [8] 2013/08/22 13:37:41: Trunk 4: Preparing for re-registration [8] 2013/08/22 13:37:41: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/22 13:37:41: Trunk 4: setup callback to send re-registration after 38 seconds [8] 2013/08/22 13:37:42: Packet authenticated by transport layer [8] 2013/08/22 13:37:45: Last message repeated 6 times [5] 2013/08/22 13:37:45: SIP Rx udp:208.73.146.95:5060: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
Content-Length: 241
Content-Type: application/sdp
v=0
o=msc3 17199 19356 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 30742 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/22 13:37:45: Call port 254: Added predefined codec 2 (mapped to 0) [8] 2013/08/22 13:37:45: Call port 254: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/22 13:37:45: Trunk Nextiva: Ignoring the SDP due to the trunk setting [5] 2013/08/22 13:37:45: SIP Rx udp:208.73.146.95:5060: SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
Content-Length: 241
Content-Type: application/sdp
v=0
o=msc3 17199 19356 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 30742 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [8] 2013/08/22 13:37:45: Call port 254: Added predefined codec 2 (mapped to 0) [8] 2013/08/22 13:37:45: Call port 254: Added rtpmap codec 1 (mapped to 101) [6] 2013/08/22 13:37:45: Trunk Nextiva: Ignoring the SDP due to the trunk setting [5] 2013/08/22 13:37:45: SIP Rx udp:208.73.146.95:5060: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-4c4bc8d13221ed72bca239be8ceaa682;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>;tag=3586189249-717981
Call-ID: 6c60efb7@pbx
CSeq: 32164 INVITE
Contact: <sip:14062094291@208.73.146.95:5060;transport=udp>
Content-Length: 241
Content-Type: application/sdp
v=0
o=msc3 17199 19356 IN IP4 208.73.146.95
s=sip call
c=IN IP4 208.73.146.95
t=0 0
m=audio 30742 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [7] 2013/08/22 13:37:45: Call 6c60efb7@pbx: Clear last INVITE [8] 2013/08/22 13:37:45: Call port 254: Added predefined codec 2 (mapped to 0) [8] 2013/08/22 13:37:45: Call port 254: Added rtpmap codec 1 (mapped to 101) [5] 2013/08/22 13:37:45: SIP Tx udp:208.73.146.95:5060: ACK sip:14062094291@208.73.146.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-cc35e0221a92676d73d0cf1a807d22d5;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>;tag=3586189249-717981
Call-ID: 6c60efb7@pbx
CSeq: 32164 ACK
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Remote-Party-ID: "Anonymous" <sip:anonymous@anonymous.invalid>
Content-Length: 0
[7] 2013/08/22 13:37:45: Determine pass-through mode after receiving response [8] 2013/08/22 13:37:45: Call state for call object 2202: connected [8] 2013/08/22 13:37:45: Call port 254: state code from 100 to 200 [8] 2013/08/22 13:37:45: Call port 253: state code from 183 to 200 [5] 2013/08/22 13:37:45: SIP Tx tls:192.168.100.111:3238: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-94w78l2wmf5w;rport=3238
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 1 INVITE
Contact: <sip:2202@192.168.100.151:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/5.0.10
Content-Type: application/sdp
Content-Length: 388
v=0
o=- 2146324864 2146324864 IN IP4 192.168.100.151
s=-
c=IN IP4 192.168.100.151
t=0 0
m=audio 61612 RTP/AVP 0 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:L2eH2LImWvqRubppwAhzLE2Hdiymzoq9DxiTr6XM
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv [7] 2013/08/22 13:37:45: 3669165245ce-5ofu1e8t5lm6: RTP pass-through mode [7] 2013/08/22 13:37:45: 6c60efb7@pbx: RTP pass-through mode [5] 2013/08/22 13:37:45: SIP Rx tls:192.168.100.111:3238: ACK sip:2202@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-zybcpy1bphvt;rport
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2202@192.168.100.111:3238;transport=tls;line=4ofyd87d>;reg-id=1
Proxy-Require: buttons-snom821
Content-Length: 0
[8] 2013/08/22 13:37:45: Packet authenticated by transport layer [8] 2013/08/22 13:37:46: Last message repeated 2 times [5] 2013/08/22 13:37:46: SIP Rx tls:192.168.100.111:3238: BYE sip:2202@192.168.100.151:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-2bcd2xwbtnyb;rport
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:2202@192.168.100.111:3238;transport=tls;line=4ofyd87d>;reg-id=1
User-Agent: snom821/8.7.3.10
RTP-RxStat: Total_Rx_Pkts=1383,Rx_Pkts=1383,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=1386,Tx_Pkts=1386,Remote_Tx_Pkts=0
Proxy-Require: buttons-snom821
Content-Length: 0
[8] 2013/08/22 13:37:46: Packet authenticated by transport layer [5] 2013/08/22 13:37:46: SIP Tx tls:192.168.100.111:3238: SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.111:3238;branch=z9hG4bK-2bcd2xwbtnyb;rport=3238
From: "Mike Schlosser" <sip:2202@192.168.100.151>;tag=movbq97udm
To: <sip:914062094291@192.168.100.151;user=phone>;tag=41bddeb4a9
Call-ID: 3669165245ce-5ofu1e8t5lm6
CSeq: 3 BYE
Contact: <sip:2202@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[7] 2013/08/22 13:37:46: 6c60efb7@pbx: Media-aware pass-through mode [8] 2013/08/22 13:37:46: Clearing call port 253, SIP call id 3669165245ce-5ofu1e8t5lm6 [8] 2013/08/22 13:37:46: Call port 254: state code from 200 to 486 [5] 2013/08/22 13:37:46: SIP Tx udp:208.73.146.95:5060: BYE sip:14062094291@208.73.146.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-7455ab467c0b8eccc76bd6e1d6e6eedc;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>;tag=3586189249-717981
Call-ID: 6c60efb7@pbx
CSeq: 32165 BYE
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Remote-Party-ID: "Anonymous" <sip:anonymous@anonymous.invalid>
Content-Length: 0
[8] 2013/08/22 13:37:46: Remove leg 5879: Call port 253, SIP call id 3669165245ce-5ofu1e8t5lm6 [8] 2013/08/22 13:37:46: Hangup: Call 253 not found [8] 2013/08/22 13:37:46: Last message repeated 2 times [5] 2013/08/22 13:37:46: SIP Rx udp:208.73.146.95:5060: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.151:5060;received=216.228.51.194;branch=z9hG4bK-7455ab467c0b8eccc76bd6e1d6e6eedc;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1481659776
To: <sip:14062094291@208.73.146.95>;tag=3586189249-717981
Call-ID: 6c60efb7@pbx
CSeq: 32165 BYE
Content-Length: 0 -
Will the GXP2200 work with a Snom PBX?
Anyone have this working?
-
So any ideas?
Sell the 870s and get better phones?
Anything?
Ringtones
in General Setup
Posted
Read this and I dont understand the part about creating a folder.
http://wiki.snomone.com/index.php?title=Customized_Ringtones
Where do you create this folder on the pbx (Snom One) or on your computer?
If Snom One, how and where do you create this? Do I need root access?
Figured I would mess around with something else until I can get help for the PA1s that dont work.