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870 Phones wont dial out on my Sip Trunks


AG1
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I cannot get my 870's to dial out on my Nextiva SIP trunks



The 820s work fine so it has to be an 870 firmware issue....does it not? If so does the beta version 8.7.4.8 fix this problem?



I have tried every possible combination of "1" or "Area code" , "country code"in the general settings and I have even tried some wildcard stuff in the dial plan, but like I said the 820s work fine its only on our touchscreen 870s




Is anyone else having this problem with 870's or am I just the lucky one again?

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Yes PnP

 

Here is something I noticed last night, if I dial a number from the buttons the calls never go through and a message comes up saying the call cannot be completed as dialed BUT if I select the number on the touchscreen dialed numbers screen the calls go through because even though a 9 is dialed (selecting my Nextiva trunks) the calls go through on my Sangoma Netborder express card

 

2013/08/20 07:25:51A Conference Room (2203@192.168.100.151) 97015720767 00:07 Nextiva 2013/08/20 07:25:11A Conference Room (2203@192.168.100.151) 94062094291 00:02 Nextiva 2013/08/20 07:24:36A Conference Room (2203@192.168.100.151) 94062094291 00:26 Netborder Express 2013/08/20 07:24:12A Conference Room (2203@192.168.100.151) 97015720767 00:10 Netborder Express 2013/08/20 07:23:33A Conference Room (2203@192.168.100.151) 94783057 00:03 Nextiva 2013/08/20 07:22:55A Conference Room (2203@192.168.100.151) 94783057 Nextiva 2013/08/20 07:21:38A Conference Room (2203@192.168.100.151) 94783057 00:10 Netborder Express

 

 

 

 

[5] 2013/08/20 07:24:17: 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-abtasztybeiq;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6
To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd
Call-ID: b86e1352c5a3-ux2gbyham388
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=zl1csrsm>;reg-id=1
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/08/20 07:24:17: Packet authenticated by transport layer [8] 2013/08/20 07:24:27: Last message repeated 20 times [5] 2013/08/20 07:24:27: 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-hohlier5joxz;rport
From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6
To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd
Call-ID: b86e1352c5a3-ux2gbyham388
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:2203@192.168.100.93:3372;transport=tls;line=zl1csrsm>;reg-id=1
User-Agent: snom870/8.7.3.19
RTP-RxStat: Total_Rx_Pkts=730,Rx_Pkts=724,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=721,Tx_Pkts=721,Remote_Tx_Pkts=2
Proxy-Require: buttons-snom870
Content-Length: 0
[8] 2013/08/20 07:24:27: Packet authenticated by transport layer [5] 2013/08/20 07:24:27: 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-hohlier5joxz;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=hpp68z45m6
To: <sip:97015720767@192.168.100.151;user=phone>;tag=1aea6c0ddd
Call-ID: b86e1352c5a3-ux2gbyham388
CSeq: 3 BYE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[7] 2013/08/20 07:24:27: 1aad7a3e@pbx: Media-aware pass-through mode [8] 2013/08/20 07:24:27: Clearing call port 298, SIP call id b86e1352c5a3-ux2gbyham388 [8] 2013/08/20 07:24:27: Call port 299: state code from 200 to 486 [8] 2013/08/20 07:24:27: Remove leg 4390: Call port 298, SIP call id b86e1352c5a3-ux2gbyham388 [5] 2013/08/20 07:24:27: 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-71d541664ae945d23b8a7c845ced56ea;rport
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=366627966
To: <sip:97015720767@192.168.100.151;user=phone>;tag=ds-2c2e9ec6-e413e3b4
Call-ID: 1aad7a3e@pbx
CSeq: 21846 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
[8] 2013/08/20 07:24:27: Hangup: Call 298 not found [8] 2013/08/20 07:24:27: Last message repeated 2 times [5] 2013/08/20 07:24:27: 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-71d541664ae945d23b8a7c845ced56ea;rport=5060
From: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>;tag=366627966
To: <sip:97015720767@192.168.100.151;user=phone>;tag=ds-2c2e9ec6-e413e3b4
Call-ID: 1aad7a3e@pbx
CSeq: 21846 BYE
Content-Length: 0

 

 

[5] 2013/08/20 07:26:06: 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-a6oxayp0t2cs;rport=3372
From: "Conference Room" <sip:2203@192.168.100.151>;tag=uhjcug5i0f
To: <sip:97015720767@192.168.100.151;user=phone>;tag=46efdabc08
Call-ID: 246f13529cad-iwcc2s3ayfqn
CSeq: 3 BYE
Contact: <sip:2203@192.168.100.151:5061;transport=tls>
User-Agent: snomONE/5.0.10
Content-Length: 0
[7] 2013/08/20 07:26:06: d6463b23@pbx: Media-aware pass-through mode [8] 2013/08/20 07:26:06: Clearing call port 304, SIP call id 246f13529cad-iwcc2s3ayfqn [8] 2013/08/20 07:26:06: Remove leg 4396: Call port 304, SIP call id 246f13529cad-iwcc2s3ayfqn [8] 2013/08/20 07:26:06: Call port 305: state code from 200 to 486 [8] 2013/08/20 07:26:06: Hangup: Call 304 not found [8] 2013/08/20 07:26:06: Last message repeated 2 times [5] 2013/08/20 07:26:06: SIP Tx udp:208.73.146.95:5060: BYE sip:7015720767@208.73.146.95:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.100.151:5060;branch=z9hG4bK-07bd5d0424065697730836ee33cc851e;rport
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1878955888
To: <sip:7015720767@208.73.146.95>;tag=3585994152-502418
Call-ID: d6463b23@pbx
CSeq: 31552 BYE
Max-Forwards: 70
Contact: <sip:14062094291@192.168.100.151:5060;transport=udp>
Remote-Party-ID: "Conference Room" <sip:4064888066@192.168.100.151;user=phone>
Content-Length: 0
[5] 2013/08/20 07:26:06: 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-07bd5d0424065697730836ee33cc851e;rport=13291
From: "Nextiva" <sip:14062094291@208.73.146.95>;tag=1878955888
To: <sip:7015720767@208.73.146.95>;tag=3585994152-502418
Call-ID: d6463b23@pbx
CSeq: 31552 BYE
Content-Length: 0
[7] 2013/08/20 07:26:06: Call d6463b23@pbx: Clear last request [5] 2013/08/20 07:26:06: BYE Response: Terminate d6463b23@pbx [8] 2013/08/20 07:26:06: Clearing call port 305, SIP call id d6463b23@pbx [8] 2013/08/20 07:26:06: Remove leg 4397: Call port 305, SIP call id d6463b23@pbx [8] 2013/08/20 07:26:07: Packet authenticated by transport layer [8] 2013/08/20 07:26:23: Last message repeated 19 times [8] 2013/08/20 07:26:23: Trunk 4: Preparing for re-registration [8] 2013/08/20 07:26:23: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/20 07:26:23: Trunk 4: setup callback to send re-registration after 37 seconds [8] 2013/08/20 07:26:23: Packet authenticated by transport layer [8] 2013/08/20 07:27:00: Last message repeated 60 times [8] 2013/08/20 07:27:00: Trunk 4: Preparing for re-registration [8] 2013/08/20 07:27:00: Trunk Nextiva: Sending registration to 208.73.146.95 [8] 2013/08/20 07:27:00: Trunk 4: setup callback to send re-registration after 38 seconds [8] 2013/08/20 07:27:01: Packet authenticated by transport layer [8] 2013/08/20 07:27:15: Last message repeated 21 times [5] 2013/08/20 07:27:15: Identify trunk (IP address/port and domain match) 3 [8] 2013/08/20 07:27:16: Packet authenticated by transport layer [8] 2013/08/20 07:27:35: Last message repeated 30 times [8] 2013/08/20 07:27:35: Could not find a trunk (3 trunks)
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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
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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/2.0

Via: 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

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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?

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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?

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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:

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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
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