vicidial.org

by **bossmon** » Tue May 12, 2020 8:16 am

I've followed telnyx's configuration instructions but my calls are not connecting. I know the issue is not with my vicidial system because I was able to get everything working with a different carrier.

Before posting in here, I did try resolving the issue with their chat support but they were unable to figure it out. They did not even see my server on their end.

My system specs:
Vicibox 9.0.2 (express-install)
Version: 2.14-751a
SVN Version: 3241
DB Schema Version: 1595
Build: 200425-0949
Asterisk Version: Asterisk 13.29.2-vici
Phone: Viciphone Webphone

I followed Telnyx's configuration instructions from their support docs here

purchased a number
set up a sip connection
provisioned my number
created an outbound profile

(the chat support agent confirmed this was set up correctly in my dashboard)

Then I added Telnyx as a carrier in my vicidial admin per their instructions:

Carrier ID: TelnyxCarrier
Name: telnyxRegistration
String : leave blank.
Template ID: NONE

Account Entry:

Code: Select all: [telnyx] disallow=all allow=ulaw allow=g729 type=peer insecure=port,invite host=sip.telnyx.com dtmfmode=rfc2833 context=default

Protocol: SIP
Global String:

Code: Select all: Telnyx=SIP/telnyx

Dial Plan:

Code: Select all: exten => _9NXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _9NXXXXXXXXXX,2,Dial(${Telnyx}/${EXTEN:1},60,tTor) exten => _9NXXXXXXXXXX,3,Hangup

When I run 'sip show peers' in Asterisk CLI, I see the the telnyx sip as active and status "OK"

Code: Select all: vicibox999*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 101/101 11.11.111.1 D Yes Yes 51927 OK (397 ms) 102/102 (Unspecified) D Yes Yes 0 UNKNOWN 201/201 (Unspecified) D Yes Yes 0 UNKNOWN gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN telnyx 192.76.120.10 Yes Yes 5060 OK (36 ms) 5 sip peers [Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline]

After some time, asterisk shows this error

Code: Select all: [May 12 08:30:46] ERROR[3322]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 12 08:31:00] ERROR[3322]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 12 08:31:03] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:31:03] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:31:03] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:31:03] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:31:08] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:31:08] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:31:14] ERROR[3322]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 12 08:31:28] ERROR[3322]: chan_sip.c:4271 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

When I log in as an agent I hear "you are the only one in this conference 'bloop' " so no issues there.

this is the asterisk debug logs when I log in to the webphone

Code: Select all: vicibox999*CLI> sip set debug on SIP Debugging re-enabled [May 12 08:35:38] [May 12 08:35:38] <--- SIP read from WS:99.99.999.9:52170 ---> [May 12 08:35:38] REGISTER sip:666.666.666.66 SIP/2.0 [May 12 08:35:38] Via: SIP/2.0/WSS 192.0.2.52;branch=z9hG4bK3599118 [May 12 08:35:38] Max-Forwards: 70 [May 12 08:35:38] To: "101" <sip:101@666.666.666.66> [May 12 08:35:38] From: "101" <sip:101@666.666.666.66>;tag=29ijds9l2b [May 12 08:35:38] Call-ID: e0odkhtfbauu0hnp00f1s3 [May 12 08:35:38] CSeq: 83 REGISTER [May 12 08:35:38] Authorization: Digest algorithm=MD5, username="101", realm="asterisk", nonce="543530e7", uri="sip:666.666.666.66", response="3f023c624d77006a4a3bd020cc450f40" [May 12 08:35:38] Contact: <sip:ij0fpvda@192.0.2.52;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:7074a775-15c3-448d-87af-a22b8e840b5a>";expires=0 [May 12 08:35:38] Supported: path, gruu, outbound [May 12 08:35:38] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:38] Content-Length: 0 [May 12 08:35:38] [May 12 08:35:38] <-------------> [May 12 08:35:38] --- (12 headers 0 lines) --- [May 12 08:35:38] [May 12 08:35:38] <--- Transmitting (NAT) to 99.99.999.9:52170 ---> [May 12 08:35:38] SIP/2.0 401 Unauthorized [May 12 08:35:38] Via: SIP/2.0/WSS 192.0.2.52;branch=z9hG4bK3599118;received=99.99.999.9;rport=52170 [May 12 08:35:38] From: "101" <sip:101@666.666.666.66>;tag=29ijds9l2b [May 12 08:35:38] To: "101" <sip:101@666.666.666.66>;tag=as23160447 [May 12 08:35:38] Call-ID: e0odkhtfbauu0hnp00f1s3 [May 12 08:35:38] CSeq: 83 REGISTER [May 12 08:35:38] Server: Asterisk PBX 13.29.2-vici [May 12 08:35:38] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:35:38] Supported: replaces, timer [May 12 08:35:38] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="381c4a34" [May 12 08:35:38] Content-Length: 0 [May 12 08:35:38] [May 12 08:35:38] [May 12 08:35:38] <------------> [May 12 08:35:38] Scheduling destruction of SIP dialog 'e0odkhtfbauu0hnp00f1s3' in 32000 ms (Method: REGISTER) [May 12 08:35:42] == WebSocket connection from '99.99.999.9:52178' for protocol 'sip' accepted using version '13' [May 12 08:35:42] [May 12 08:35:42] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:42] REGISTER sip:666.666.666.66 SIP/2.0 [May 12 08:35:42] Via: SIP/2.0/WSS 192.0.2.197;branch=z9hG4bK2340421 [May 12 08:35:42] Max-Forwards: 70 [May 12 08:35:42] To: "101" <sip:101@666.666.666.66> [May 12 08:35:42] From: "101" <sip:101@666.666.666.66>;tag=aalvi2n7g3 [May 12 08:35:42] Call-ID: 0hrd94bqt1i17bll60u3jf [May 12 08:35:42] CSeq: 81 REGISTER [May 12 08:35:42] Contact: <sip:v39je08u@192.0.2.197;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:25a0871e-8c2a-432b-887c-3ff9b5aab1a3>";expires=600 [May 12 08:35:42] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 12 08:35:42] Supported: path, gruu, outbound [May 12 08:35:42] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:42] Content-Length: 0 [May 12 08:35:42] [May 12 08:35:42] <-------------> [May 12 08:35:42] --- (12 headers 0 lines) --- [May 12 08:35:42] [May 12 08:35:42] <--- Transmitting (NAT) to 99.99.999.9:52178 ---> [May 12 08:35:42] SIP/2.0 401 Unauthorized [May 12 08:35:42] Via: SIP/2.0/WSS 192.0.2.197;branch=z9hG4bK2340421;received=99.99.999.9;rport=52178 [May 12 08:35:42] From: "101" <sip:101@666.666.666.66>;tag=aalvi2n7g3 [May 12 08:35:42] To: "101" <sip:101@666.666.666.66>;tag=as0509ae51 [May 12 08:35:42] Call-ID: 0hrd94bqt1i17bll60u3jf [May 12 08:35:42] CSeq: 81 REGISTER [May 12 08:35:42] Server: Asterisk PBX 13.29.2-vici [May 12 08:35:42] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:35:42] Supported: replaces, timer [May 12 08:35:42] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="157e3b2a" [May 12 08:35:42] Content-Length: 0 [May 12 08:35:42] [May 12 08:35:42] [May 12 08:35:42] <------------> [May 12 08:35:42] Scheduling destruction of SIP dialog '0hrd94bqt1i17bll60u3jf' in 32000 ms (Method: REGISTER) [May 12 08:35:43] [May 12 08:35:43] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:43] REGISTER sip:666.666.666.66 SIP/2.0 [May 12 08:35:43] Via: SIP/2.0/WSS 192.0.2.197;branch=z9hG4bK6394159 [May 12 08:35:43] Max-Forwards: 70 [May 12 08:35:43] To: "101" <sip:101@666.666.666.66> [May 12 08:35:43] From: "101" <sip:101@666.666.666.66>;tag=aalvi2n7g3 [May 12 08:35:43] Call-ID: 0hrd94bqt1i17bll60u3jf [May 12 08:35:43] CSeq: 82 REGISTER [May 12 08:35:43] Authorization: Digest algorithm=MD5, username="101", realm="asterisk", nonce="157e3b2a", uri="sip:666.666.666.66", response="fea41644686d230cfe77b56bedd778c3" [May 12 08:35:43] Contact: <sip:v39je08u@192.0.2.197;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:25a0871e-8c2a-432b-887c-3ff9b5aab1a3>";expires=600 [May 12 08:35:43] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 12 08:35:43] Supported: path, gruu, outbound [May 12 08:35:43] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:43] Content-Length: 0 [May 12 08:35:43] [May 12 08:35:43] <-------------> [May 12 08:35:43] --- (13 headers 0 lines) --- [May 12 08:35:43] -- Registered SIP '101' at 99.99.999.9:52178 [May 12 08:35:43] Reliably Transmitting (NAT) to 99.99.999.9:52178: [May 12 08:35:43] OPTIONS sip:v39je08u@192.0.2.197;transport=wss SIP/2.0 [May 12 08:35:43] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK2e685260;rport [May 12 08:35:43] Max-Forwards: 70 [May 12 08:35:43] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as30c3f297 [May 12 08:35:43] To: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:43] Contact: <sip:asterisk@666.666.666.66:0;transport=ws> [May 12 08:35:43] Call-ID: 35f2238c2aa9f9fe22925ea5424d46c4@666.666.666.66:0 [May 12 08:35:43] CSeq: 102 OPTIONS [May 12 08:35:43] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:35:43] Date: Tue, 12 May 2020 12:35:43 GMT [May 12 08:35:43] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:35:43] Supported: replaces, timer [May 12 08:35:43] Content-Length: 0 [May 12 08:35:43] [May 12 08:35:43] [May 12 08:35:43] --- [May 12 08:35:43] [May 12 08:35:43] <--- Transmitting (NAT) to 99.99.999.9:52178 ---> [May 12 08:35:43] SIP/2.0 200 OK [May 12 08:35:43] Via: SIP/2.0/WSS 192.0.2.197;branch=z9hG4bK6394159;received=99.99.999.9;rport=52178 [May 12 08:35:43] From: "101" <sip:101@666.666.666.66>;tag=aalvi2n7g3 [May 12 08:35:43] To: "101" <sip:101@666.666.666.66>;tag=as0509ae51 [May 12 08:35:43] Call-ID: 0hrd94bqt1i17bll60u3jf [May 12 08:35:43] CSeq: 82 REGISTER [May 12 08:35:43] Server: Asterisk PBX 13.29.2-vici [May 12 08:35:43] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:35:43] Supported: replaces, timer [May 12 08:35:43] Expires: 600 [May 12 08:35:43] Contact: <sip:v39je08u@192.0.2.197;transport=wss>;expires=600 [May 12 08:35:43] Date: Tue, 12 May 2020 12:35:43 GMT [May 12 08:35:43] Content-Length: 0 [May 12 08:35:43] [May 12 08:35:43] [May 12 08:35:43] <------------> [May 12 08:35:43] Scheduling destruction of SIP dialog '2e0aef8b25218c500d3748e01c754a1a@666.666.666.66:0' in 11648 ms (Method: NOTIFY) [May 12 08:35:43] Reliably Transmitting (NAT) to 99.99.999.9:52178: [May 12 08:35:43] NOTIFY sip:v39je08u@192.0.2.197;transport=wss SIP/2.0 [May 12 08:35:43] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK47dc994f;rport [May 12 08:35:43] Max-Forwards: 70 [May 12 08:35:43] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as2d7e4bef [May 12 08:35:43] To: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:43] Contact: <sip:asterisk@666.666.666.66:0;transport=ws> [May 12 08:35:43] Call-ID: 2e0aef8b25218c500d3748e01c754a1a@666.666.666.66:0 [May 12 08:35:43] CSeq: 102 NOTIFY [May 12 08:35:43] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:35:43] Event: message-summary [May 12 08:35:43] Content-Type: application/simple-message-summary [May 12 08:35:43] Content-Length: 109 [May 12 08:35:43] [May 12 08:35:43] Messages-Waiting: no [May 12 08:35:43] Message-Account: sip:asterisk@666.666.666.66:0;transport=WS [May 12 08:35:43] Voice-Message: 0/0 (0/0) [May 12 08:35:43] [May 12 08:35:43] --- [May 12 08:35:43] Scheduling destruction of SIP dialog '0hrd94bqt1i17bll60u3jf' in 32000 ms (Method: REGISTER) [May 12 08:35:43] [May 12 08:35:43] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:43] SIP/2.0 200 OK [May 12 08:35:43] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK2e685260;rport [May 12 08:35:43] To: <sip:v39je08u@192.0.2.197;transport=wss>;tag=3kgu8t9edm [May 12 08:35:43] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as30c3f297 [May 12 08:35:43] Call-ID: 35f2238c2aa9f9fe22925ea5424d46c4@666.666.666.66:0 [May 12 08:35:43] CSeq: 102 OPTIONS [May 12 08:35:43] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 12 08:35:43] Accept: application/sdp,application/dtmf-relay [May 12 08:35:43] Supported: outbound [May 12 08:35:43] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:43] Content-Length: 0 [May 12 08:35:43] [May 12 08:35:43] <-------------> [May 12 08:35:43] --- (11 headers 0 lines) --- [May 12 08:35:43] [May 12 08:35:43] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:43] SIP/2.0 481 Subscription does not exist [May 12 08:35:43] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK47dc994f;rport [May 12 08:35:43] To: <sip:v39je08u@192.0.2.197;transport=wss>;tag=jgsqo79gq1 [May 12 08:35:43] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as2d7e4bef [May 12 08:35:43] Call-ID: 2e0aef8b25218c500d3748e01c754a1a@666.666.666.66:0 [May 12 08:35:43] CSeq: 102 NOTIFY [May 12 08:35:43] Supported: outbound [May 12 08:35:43] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:43] Content-Length: 0 [May 12 08:35:43] [May 12 08:35:43] <-------------> [May 12 08:35:43] --- (9 headers 0 lines) --- [May 12 08:35:44] Really destroying SIP dialog '35f2238c2aa9f9fe22925ea5424d46c4@666.666.666.66:0' Method: OPTIONS [May 12 08:35:44] Really destroying SIP dialog '2e0aef8b25218c500d3748e01c754a1a@666.666.666.66:0' Method: NOTIFY [May 12 08:35:44] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:35:44] == Using SIP RTP CoS mark 5 [May 12 08:35:44] Audio is at 14882 [May 12 08:35:44] Adding codec ulaw to SDP [May 12 08:35:44] Adding codec gsm to SDP [May 12 08:35:44] Adding non-codec 0x1 (telephone-event) to SDP [May 12 08:35:44] Reliably Transmitting (NAT) to 99.99.999.9:52178: [May 12 08:35:44] INVITE sip:v39je08u@192.0.2.197;transport=wss SIP/2.0 [May 12 08:35:44] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK16ca7026;rport [May 12 08:35:44] Max-Forwards: 70 [May 12 08:35:44] From: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66:0>;tag=as41307394 [May 12 08:35:44] To: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:44] Contact: <sip:5555555555@666.666.666.66:0;transport=ws> [May 12 08:35:44] Call-ID: 7a50ccd135ddc31564980c87570360f4@666.666.666.66:0 [May 12 08:35:44] CSeq: 102 INVITE [May 12 08:35:44] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:35:44] Date: Tue, 12 May 2020 12:35:44 GMT [May 12 08:35:44] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:35:44] Supported: replaces, timer [May 12 08:35:44] Remote-Party-ID: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66>;party=calling;privacy=off;screen=no [May 12 08:35:44] Content-Type: application/sdp [May 12 08:35:44] Content-Length: 682 [May 12 08:35:44] [May 12 08:35:44] v=0 [May 12 08:35:44] o=root 1350343420 1350343420 IN IP4 666.666.666.66 [May 12 08:35:44] s=Asterisk PBX 13.29.2-vici [May 12 08:35:44] c=IN IP4 666.666.666.66 [May 12 08:35:44] t=0 0 [May 12 08:35:44] m=audio 14882 RTP/SAVPF 0 3 101 [May 12 08:35:44] a=rtpmap:0 PCMU/8000 [May 12 08:35:44] a=rtpmap:3 GSM/8000 [May 12 08:35:44] a=rtpmap:101 telephone-event/8000 [May 12 08:35:44] a=fmtp:101 0-16 [May 12 08:35:44] a=ptime:20 [May 12 08:35:44] a=maxptime:150 [May 12 08:35:44] a=ice-ufrag:6bb8ddde2e51d9d0135a70b857533f0d [May 12 08:35:44] a=ice-pwd:0539aa187a44003d612c74ac24079f66 [May 12 08:35:44] a=candidate:H9b8a9d56 1 UDP 2130706431 666.666.666.66 14882 typ host [May 12 08:35:44] a=candidate:H9b8a9d56 2 UDP 2130706430 666.666.666.66 14883 typ host [May 12 08:35:44] a=connection:new [May 12 08:35:44] a=setup:actpass [May 12 08:35:44] a=fingerprint:SHA-256 8C:0C:99:73:ED:A9:FB:50:E0:88:BC:CF:54:2D:BE:72:77:8D:0E:76:EE:8C:37:AA:88:33:D8:54:6E:7B:60:0E [May 12 08:35:44] a=rtcp-mux [May 12 08:35:44] a=sendrecv [May 12 08:35:44] [May 12 08:35:44] --- [May 12 08:35:44] -- Called 101 [May 12 08:35:44] [May 12 08:35:44] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:44] SIP/2.0 100 Trying [May 12 08:35:44] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK16ca7026;rport [May 12 08:35:44] To: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:44] From: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66:0>;tag=as41307394 [May 12 08:35:44] Call-ID: 7a50ccd135ddc31564980c87570360f4@666.666.666.66:0 [May 12 08:35:44] CSeq: 102 INVITE [May 12 08:35:44] Supported: outbound [May 12 08:35:44] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:44] Content-Length: 0 [May 12 08:35:44] [May 12 08:35:44] <-------------> [May 12 08:35:44] --- (9 headers 0 lines) --- [May 12 08:35:44] [May 12 08:35:44] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:44] SIP/2.0 180 Ringing [May 12 08:35:44] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK16ca7026;rport [May 12 08:35:44] To: <sip:v39je08u@192.0.2.197;transport=wss>;tag=p9tbinr4vh [May 12 08:35:44] From: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66:0>;tag=as41307394 [May 12 08:35:44] Call-ID: 7a50ccd135ddc31564980c87570360f4@666.666.666.66:0 [May 12 08:35:44] CSeq: 102 INVITE [May 12 08:35:44] Contact: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:44] Supported: outbound [May 12 08:35:44] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:44] Content-Length: 0 [May 12 08:35:44] [May 12 08:35:44] <-------------> [May 12 08:35:44] --- (10 headers 0 lines) --- [May 12 08:35:44] sip_route_dump: route/path hop: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:44] -- SIP/101-00000002 is ringing [May 12 08:35:44] > 0x5599d8a6a800 -- Strict RTP learning after remote address set to: 99.99.999.9:51923 [May 12 08:35:45] > 0x5599d8a6a800 -- Strict RTP learning after remote address set to: 99.99.999.9:51923 [May 12 08:35:45] [May 12 08:35:45] <--- SIP read from WS:99.99.999.9:52178 ---> [May 12 08:35:45] SIP/2.0 200 OK [May 12 08:35:45] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK16ca7026;rport [May 12 08:35:45] To: <sip:v39je08u@192.0.2.197;transport=wss>;tag=p9tbinr4vh [May 12 08:35:45] From: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66:0>;tag=as41307394 [May 12 08:35:45] Call-ID: 7a50ccd135ddc31564980c87570360f4@666.666.666.66:0 [May 12 08:35:45] CSeq: 102 INVITE [May 12 08:35:45] Contact: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:45] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 12 08:35:45] Supported: outbound [May 12 08:35:45] User-Agent: VICIphone 1.0-rc1 [May 12 08:35:45] Content-Type: application/sdp [May 12 08:35:45] Content-Length: 1428 [May 12 08:35:45] [May 12 08:35:45] v=0 [May 12 08:35:45] o=- 3241321883756551513 2 IN IP4 127.0.0.1 [May 12 08:35:45] s=- [May 12 08:35:45] t=0 0 [May 12 08:35:45] a=msid-semantic: WMS anzCXQOyoJxcKLyqBVFKl7KIpxwqQt7pPeeH [May 12 08:35:45] m=audio 51923 UDP/TLS/RTP/SAVPF 0 101 [May 12 08:35:45] c=IN IP4 99.99.999.9 [May 12 08:35:45] a=rtcp:9 IN IP4 0.0.0.0 [May 12 08:35:45] a=candidate:2999745851 1 udp 2122260223 192.168.56.1 51922 typ host generation 0 network-id 2 [May 12 08:35:45] a=candidate:2437072876 1 udp 2122194687 192.168.1.2 51923 typ host generation 0 network-id 1 network-cost 10 [May 12 08:35:45] a=candidate:941443129 1 udp 1685987071 99.99.999.9 51923 typ srflx raddr 192.168.1.2 rport 51923 generation 0 network-id 1 network-cost 10 [May 12 08:35:45] a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 9 typ host tcptype active generation 0 network-id 2 [May 12 08:35:45] a=candidate:3753982748 1 tcp 1518214911 192.168.1.2 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 12 08:35:45] a=ice-ufrag:uR+F [May 12 08:35:45] a=ice-pwd:NhznkAGp4+OpFr+dCk4FqBS5 [May 12 08:35:45] a=ice-options:trickle [May 12 08:35:45] a=fingerprint:sha-256 70:CF:49:3F:D9:3E:47:62:83:13:06:18:0A:0C:A9:7A:91:C9:2B:25:7A:EE:B5:25:41:B9:F7:74:6D:9C:3B:DD [May 12 08:35:45] a=setup:active [May 12 08:35:45] a=mid:0 [May 12 08:35:45] a=sendrecv [May 12 08:35:45] a=msid:anzCXQOyoJxcKLyqBVFKl7KIpxwqQt7pPeeH 8377576b-2175-4ff7-ad99-c90c76c645d5 [May 12 08:35:45] a=rtcp-mux [May 12 08:35:45] a=rtpmap:0 PCMU/8000 [May 12 08:35:45] a=rtpmap:101 telephone-event/8000 [May 12 08:35:45] a=ssrc:4139886211 cname:FxpP+3H6r1KFO/UT [May 12 08:35:45] a=ssrc:4139886211 msid:anzCXQOyoJxcKLyqBVFKl7KIpxwqQt7pPeeH 8377576b-2175-4ff7-ad99-c90c76c645d5 [May 12 08:35:45] a=ssrc:4139886211 mslabel:anzCXQOyoJxcKLyqBVFKl7KIpxwqQt7pPeeH [May 12 08:35:45] a=ssrc:4139886211 label:8377576b-2175-4ff7-ad99-c90c76c645d5 [May 12 08:35:45] <-------------> [May 12 08:35:45] --- (12 headers 28 lines) --- [May 12 08:35:45] Found RTP audio format 0 [May 12 08:35:45] Found RTP audio format 101 [May 12 08:35:45] Found audio description format PCMU for ID 0 [May 12 08:35:45] Found audio description format telephone-event for ID 101 [May 12 08:35:45] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [May 12 08:35:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 12 08:35:45] Peer audio RTP is at port 99.99.999.9:51923 [May 12 08:35:45] sip_route_dump: route/path hop: <sip:v39je08u@192.0.2.197;transport=wss> [May 12 08:35:45] Transmitting (NAT) to 99.99.999.9:52178: [May 12 08:35:45] ACK sip:v39je08u@192.0.2.197;transport=wss SIP/2.0 [May 12 08:35:45] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK3806fdd0;rport [May 12 08:35:45] Max-Forwards: 70 [May 12 08:35:45] From: "ACagcW1589286942101101101101" <sip:5555555555@666.666.666.66:0>;tag=as41307394 [May 12 08:35:45] To: <sip:v39je08u@192.0.2.197;transport=wss>;tag=p9tbinr4vh [May 12 08:35:45] Contact: <sip:5555555555@666.666.666.66:0;transport=ws> [May 12 08:35:45] Call-ID: 7a50ccd135ddc31564980c87570360f4@666.666.666.66:0 [May 12 08:35:45] CSeq: 102 ACK [May 12 08:35:45] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:35:45] Content-Length: 0 [May 12 08:35:45] [May 12 08:35:45] [May 12 08:35:45] --- [May 12 08:35:45] -- SIP/101-00000002 answered [May 12 08:35:45] -- Executing [8600051@default:1] MeetMe("SIP/101-00000002", "8600051,F") in new stack [May 12 08:35:45] -- Created MeetMe conference 1023 for conference '8600051' [May 12 08:35:45] -- <SIP/101-00000002> Playing 'conf-onlyperson.gsm' (language 'en') [May 12 08:35:45] > 0x5599d8a6a800 -- Strict RTP learning after ICE completion [May 12 08:35:45] > 0x5599d8a6a800 -- Strict RTP switching to RTP target address 99.99.999.9:51923 as source [May 12 08:35:46] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:35:50] > 0x5599d8a6a800 -- Strict RTP learning complete - Locking on source address 99.99.999.9:51923 vicibox999*CLI>

99.99.999.9 = my local IP modified for privacy
666.666.666.66 = my server IP modified for privacy
5555555555 = my telnyx phone number modified for privacy

And here is the debug log when I go "active" and "ready" for outbound calls.

Code: Select all: [May 12 08:51:44] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:51:44] NOTICE[6934]: core_local.c:756 local_call: No such extension/context 918067481071@default while calling Local channel [May 12 08:51:44] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:51:44] NOTICE[6938]: core_local.c:756 local_call: No such extension/context 912813375074@default while calling Local channel [May 12 08:51:45] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:51:45] NOTICE[6943]: core_local.c:756 local_call: No such extension/context 914797859000@default while calling Local channel [May 12 08:51:45] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:51:45] NOTICE[6947]: core_local.c:756 local_call: No such extension/context 914066521916@default while calling Local channel [May 12 08:51:45] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:51:45] NOTICE[6951]: core_local.c:756 local_call: No such extension/context 916019362189@default while calling Local channel [May 12 08:51:45] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:51:45] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:51:46] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:51:46] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:51:46] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:51:50] Really destroying SIP dialog '0hrd94bqt1i17bll60u3jf' Method: REGISTER [May 12 08:51:50] ERROR[3322]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor [May 12 08:51:50] ERROR[3322]: tcptls.c:466 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor [May 12 08:51:50] == WebSocket connection from '99.99.999.9:52178' forcefully closed due to fatal write error [May 12 08:51:56] Really destroying SIP dialog 'uoatesrtbs94553rptmmsg' Method: REGISTER [May 12 08:52:03] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:52:03] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:52:03] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:52:03] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:52:08] == Manager 'sendcron' logged on from 127.0.0.1 [May 12 08:52:08] == Manager 'sendcron' logged off from 127.0.0.1 [May 12 08:52:25] Reliably Transmitting (NAT) to 99.99.999.9:52304: [May 12 08:52:25] OPTIONS sip:aukob9p5@192.0.2.248;transport=wss SIP/2.0 [May 12 08:52:25] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK35e3c6fc;rport [May 12 08:52:25] Max-Forwards: 70 [May 12 08:52:25] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as6b988aac [May 12 08:52:25] To: <sip:aukob9p5@192.0.2.248;transport=wss> [May 12 08:52:25] Contact: <sip:asterisk@666.666.666.66:0;transport=ws> [May 12 08:52:25] Call-ID: 7f842daf71d85d587ba2da8d025a21dd@666.666.666.66:0 [May 12 08:52:25] CSeq: 102 OPTIONS [May 12 08:52:25] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:52:25] Date: Tue, 12 May 2020 12:52:25 GMT [May 12 08:52:25] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:52:25] Supported: replaces, timer [May 12 08:52:25] Content-Length: 0 [May 12 08:52:25] [May 12 08:52:25] [May 12 08:52:25] --- [May 12 08:52:25] [May 12 08:52:25] <--- SIP read from WS:99.99.999.9:52304 ---> [May 12 08:52:25] SIP/2.0 200 OK [May 12 08:52:25] Via: SIP/2.0/WS 666.666.666.66:0;branch=z9hG4bK35e3c6fc;rport [May 12 08:52:25] To: <sip:aukob9p5@192.0.2.248;transport=wss>;tag=bbs5baj480 [May 12 08:52:25] From: "asterisk" <sip:asterisk@666.666.666.66:0>;tag=as6b988aac [May 12 08:52:25] Call-ID: 7f842daf71d85d587ba2da8d025a21dd@666.666.666.66:0 [May 12 08:52:25] CSeq: 102 OPTIONS [May 12 08:52:25] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 12 08:52:25] Accept: application/sdp,application/dtmf-relay [May 12 08:52:25] Supported: outbound [May 12 08:52:25] User-Agent: VICIphone 1.0-rc1 [May 12 08:52:25] Content-Length: 0 [May 12 08:52:25] [May 12 08:52:25] <-------------> [May 12 08:52:25] --- (11 headers 0 lines) --- [May 12 08:52:26] Really destroying SIP dialog '7f842daf71d85d587ba2da8d025a21dd@666.666.666.66:0' Method: OPTIONS [May 12 08:52:31] Reliably Transmitting (NAT) to 192.76.120.10:5060: [May 12 08:52:31] OPTIONS sip:sip.telnyx.com SIP/2.0 [May 12 08:52:31] Via: SIP/2.0/UDP 666.666.666.66:5060;branch=z9hG4bK7272cc00;rport [May 12 08:52:31] Max-Forwards: 70 [May 12 08:52:31] From: "asterisk" <sip:asterisk@666.666.666.66>;tag=as3b20e58d [May 12 08:52:31] To: <sip:sip.telnyx.com> [May 12 08:52:31] Contact: <sip:asterisk@666.666.666.66:5060> [May 12 08:52:31] Call-ID: 2a9f11dc627095530710e4103f4b9f38@666.666.666.66:5060 [May 12 08:52:31] CSeq: 102 OPTIONS [May 12 08:52:31] User-Agent: Asterisk PBX 13.29.2-vici [May 12 08:52:31] Date: Tue, 12 May 2020 12:52:31 GMT [May 12 08:52:31] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [May 12 08:52:31] Supported: replaces, timer [May 12 08:52:31] Content-Length: 0 [May 12 08:52:31] [May 12 08:52:31] [May 12 08:52:31] --- [May 12 08:52:31] [May 12 08:52:31] <--- SIP read from UDP:192.76.120.10:5060 ---> [May 12 08:52:31] SIP/2.0 200 Keepalive [May 12 08:52:31] Via: SIP/2.0/UDP 666.666.666.66:5060;branch=z9hG4bK7272cc00;rport=5060;received=666.666.666.66 [May 12 08:52:31] From: "asterisk" <sip:asterisk@666.666.666.66>;tag=as3b20e58d [May 12 08:52:31] To: <sip:sip.telnyx.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.7439 [May 12 08:52:31] Call-ID: 2a9f11dc627095530710e4103f4b9f38@666.666.666.66:5060 [May 12 08:52:31] CSeq: 102 OPTIONS [May 12 08:52:31] Server: kamailio (5.0.8 (x86_64/linux)) [May 12 08:52:31] Content-Length: 0 [May 12 08:52:31] [May 12 08:52:31] <-------------> [May 12 08:52:31] --- (8 headers 0 lines) --- [May 12 08:52:31] Really destroying SIP dialog '2a9f11dc627095530710e4103f4b9f38@666.666.666.66:5060' Method: OPTIONS vicibox01*CLI>

And when I look in my admin "Real-Time Main Report" I do see the calls being dialed but for some reason they are not connecting to my vicidial webphone and are dropping at 100%.

Thank you in advance for your help. I am grateful for this software and appreciate this community!

by **carpenox** » Tue May 12, 2020 8:32 am

I use telnyx, did they activate your tunnel with actual channels yet? you have to ask them to do this....its 12 dollars per channel

by **bossmon** » Tue May 12, 2020 9:00 am

Hey Nox,
not sure. I''m "level 1 verified". How do I request that? Also, can you confirm the carrier settings above?

carpenox wrote:I use telnyx, did they activate your tunnel with actual channels yet? you have to ask them to do this....its 12 dollars per channel

by **bossmon** » Tue May 12, 2020 9:17 am

Ok, I found the button to do it manually and added 1 channel. It says it's for inbound though. I am on an outbound campaign.

After activating the channel I am still having the same issues.

carpenox wrote:I use telnyx, did they activate your tunnel with actual channels yet? you have to ask them to do this....its 12 dollars per channel

by **carpenox** » Tue May 12, 2020 9:19 am

yea u have to talk to their tech support, but honestly i didnt like their service, a lot of issues, i ended up gonig with sip.us

by **bossmon** » Tue May 12, 2020 9:30 am

Were you able to get the calls running with telnyx?

carpenox wrote:yea u have to talk to their tech support, but honestly i didnt like their service, a lot of issues, i ended up gonig with sip.us

by **carpenox** » Tue May 12, 2020 9:32 am

yea, but only after talking to like 4 tech people

by **bossmon** » Wed May 13, 2020 12:17 am

After digging around the forum and reading some related issues I was able to resolve this issue.

I made some adjustments to the carrier info provided by telnyx and was able to get it going.

Here's the new carrier info:

Account Entry

Code: Select all: [telnyx] disallow=all allow=ulaw allow=g729 type=friend insecure=port,invite host=sip.telnyx.com dtmfmode=rfc2833 context=trunkinbound

changed type=peer to type=friend
changed context=default to context=trunkinbound

Dial Plan Entry

Code: Select all: exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91NXXNXXXXXX,2,Dial(${Telnyx}/${EXTEN:1},60,tTor) exten => _91NXXNXXXXXX,3,Hangup

changed the extension from _9NXXXXXXXXXX to _91NXXNXXXXXX

Hope this helps anyone else having issues setting up their vicidial with telnyx.

by **carpenox** » Wed May 13, 2020 6:44 am

good postback

vicidial.org

[Resolved] Need help configuring carrier Telnyx to vicidial

[Resolved] Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: Need help configuring carrier Telnyx to vicidial

Re: [Resolved] Need help configuring carrier Telnyx to vicid

Re: [Resolved] Need help configuring carrier Telnyx to vicid

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