vicidial.org

by **bronson** » Fri Aug 05, 2022 5:19 pm

Hey Vicicrew, hoping you can help me out.

I am getting the audio message "that number has not yet been assigned, please contact technical support." and then a single busy tone, then immediate hangup after I make an outbound call to any phone number.

I have a fresh Vicibox 10 Express-Install on a dedicated server. I am using Viciphone and it registers fine. I do hear the "you are the only one in this conference" message when I log in.

Specs:
Version: 2.14b0.5
SVN Version: 3612
DB Schema Version: 1662

In case it is relevant, I have enabled the whitelist firewall.

Below are my Viciphone debug outputs:

Personal Data changed for privacy
111.111.11.11 = my carrier ip
222.222.22.22 = my server ip
33.333.333.333 = my agent ip address
CARRIERNAME = The name of my carrier
carrier.domain.com = the domain the carrier provides to mask the carrier ip
carrier.com = the domain from my carrier.
VICIbox10 = my vicibox Hostname

Viciphone Debug from agent interface

Code: Select all: 2022-08-05 16:36:34 => displayName: 1000 uri: 1000@222.222.22.22 authorizationUser: 1000 password: Ph0nePW wsServers: wss://dialer.mydomain.com:8089/ws 2022-08-05 16:36:37 => Got Invite from <0000000000> "ACagcW16597317941000100010001000" 2022-08-05 16:36:37 => Auto-Answered Call 2022-08-05 16:36:37 => Session Accepted Event Fired

Below are my Asterisk debug outputs

- ViciBox v.10.0.1 220503

Code: Select all: VICIbox10:~ # asterisk -r [Aug 5 16:36:13] Asterisk 13.38.2-vici, Copyright (C) 1999 - 2014, Digium, Inc. and others. [Aug 5 16:36:13] Created by Mark Spencer <markster@digium.com> [Aug 5 16:36:13] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Aug 5 16:36:13] This is free software, with components licensed under the GNU General Public [Aug 5 16:36:13] License version 2 and other licenses; you are welcome to redistribute it under [Aug 5 16:36:13] certain conditions. Type 'core show license' for details. [Aug 5 16:36:13] ========================================================================= [Aug 5 16:36:13] Please note that this version of Asterisk no longer receives bug fixes. [Aug 5 16:36:13] Consult the following URL for Asterisk version support status information: [Aug 5 16:36:13] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions [Aug 5 16:36:13] ========================================================================= [Aug 5 16:36:13] Connected to Asterisk 13.38.2-vici currently running on VICIbox10 (pid = 995)

During agent Log in

Code: Select all: [Aug 5 17:22:52] == WebSocket connection from '33.333.333.333:55870' for protocol 'sip' accepted using version '13' [Aug 5 17:22:52] -- Registered SIP '1000' at 33.333.333.333:55870 [Aug 5 17:22:52] NOTICE[11344]: chan_sip.c:24817 handle_response_peerpoke: Peer '1000' is now Reachable. (64ms / 2000ms) [Aug 5 17:22:53] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:22:53] ERROR[11643]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("VICIbox10", "(null)", ...): Name or service not known [Aug 5 17:22:53] WARNING[11643]: acl.c:892 resolve_first: Unable to lookup 'VICIbox10' [Aug 5 17:22:53] == Using SIP RTP CoS mark 5 [Aug 5 17:22:53] -- Called 1000 [Aug 5 17:22:53] -- SIP/1000-00000006 is ringing [Aug 5 17:22:53] > 0x7fa7e4010720 -- Strict RTP learning after remote address set to: 33.333.333.333:61943 [Aug 5 17:22:54] -- SIP/1000-00000006 answered [Aug 5 17:22:54] -- Executing [8600051@default:1] MeetMe("SIP/1000-00000006", "8600051,F") in new stack [Aug 5 17:22:54] -- Created MeetMe conference 1023 for conference '8600051' [Aug 5 17:22:54] -- <SIP/1000-00000006> Playing 'conf-onlyperson.gsm' (language 'en') [Aug 5 17:22:54] > 0x7fa7e4010720 -- Strict RTP learning after ICE completion [Aug 5 17:22:54] > 0x7fa7e4010720 -- Strict RTP learning after remote address set to: 33.333.333.333:61943 [Aug 5 17:22:54] > 0x7fa7e4010720 -- Strict RTP switching to RTP target address 33.333.333.333:61943 as source [Aug 5 17:22:55] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:22:59] > 0x7fa7e4010720 -- Strict RTP learning complete - Locking on source address 33.333.333.333:61943 [Aug 5 17:23:02] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:23:02] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:23:02] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:23:02] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:23:07] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:23:07] == Manager 'sendcron' logged off from 127.0.0.1 VICIbox10*CLI>

SIP Show Peers

Code: Select all: VICIbox10*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description 1000/1000 33.333.333.333 D Yes Yes 55938 OK (69 ms) 2000/2000 (Unspecified) D Yes Yes 0 UNKNOWN 3000/3000 (Unspecified) D Yes Yes 0 UNKNOWN 4000/4000 (Unspecified) D Yes Yes 0 UNKNOWN 5000/5000 (Unspecified) D Yes Yes 0 UNKNOWN 6000/6000 (Unspecified) D Yes Yes 0 UNKNOWN 7000/7000 (Unspecified) D Yes Yes 0 UNKNOWN 7528/7528 (Unspecified) D Yes Yes 0 UNKNOWN 8000/8000 (Unspecified) D Yes Yes 0 UNKNOWN 9000/9000 (Unspecified) D Yes Yes 0 UNKNOWN CARRIERNAME 111.111.11.11 Yes Yes 5060 OK (2 ms) gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN 12 sip peers [Monitored: 2 online, 10 offline Unmonitored: 0 online, 0 offline]

Turning on SIP Debug

Code: Select all: VICIbox10*CLI> sip set debug on SIP Debugging enabled [Aug 5 17:48:06] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:48:06] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:48:07] Reliably Transmitting (NAT) to 33.333.333.333:55938: [Aug 5 17:48:07] OPTIONS sip:t7atthm2@192.0.2.100;transport=wss SIP/2.0 [Aug 5 17:48:07] Via: SIP/2.0/WS 222.222.22.22:0;branch=z9hG4bK5ab9f73d;rport [Aug 5 17:48:07] Max-Forwards: 70 [Aug 5 17:48:07] From: "asterisk" <sip:asterisk@222.222.22.22:0>;tag=as21d416e2 [Aug 5 17:48:07] To: <sip:t7atthm2@192.0.2.100;transport=wss> [Aug 5 17:48:07] Contact: <sip:asterisk@222.222.22.22:0;transport=ws> [Aug 5 17:48:07] Call-ID: 262c29f773ddec9f6e06235838eecfd1@222.222.22.22:0 [Aug 5 17:48:07] CSeq: 102 OPTIONS [Aug 5 17:48:07] User-Agent: Asterisk PBX 13.38.2-vici [Aug 5 17:48:07] Date: Fri, 05 Aug 2022 21:48:07 GMT [Aug 5 17:48:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:48:07] Supported: replaces, timer [Aug 5 17:48:07] Content-Length: 0 [Aug 5 17:48:07] [Aug 5 17:48:07] [Aug 5 17:48:07] --- [Aug 5 17:48:07] [Aug 5 17:48:07] <--- SIP read from WS:33.333.333.333:55938 ---> [Aug 5 17:48:07] SIP/2.0 200 OK [Aug 5 17:48:07] Via: SIP/2.0/WS 222.222.22.22:0;branch=z9hG4bK5ab9f73d;rport [Aug 5 17:48:07] To: <sip:t7atthm2@192.0.2.100;transport=wss>;tag=aa0suu93l8 [Aug 5 17:48:07] From: "asterisk" <sip:asterisk@222.222.22.22:0>;tag=as21d416e2 [Aug 5 17:48:07] Call-ID: 262c29f773ddec9f6e06235838eecfd1@222.222.22.22:0 [Aug 5 17:48:07] CSeq: 102 OPTIONS [Aug 5 17:48:07] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [Aug 5 17:48:07] Accept: application/sdp,application/dtmf-relay [Aug 5 17:48:07] Supported: outbound [Aug 5 17:48:07] User-Agent: VICIphone 1.0-rc1 [Aug 5 17:48:07] Content-Length: 0 [Aug 5 17:48:07] [Aug 5 17:48:07] <-------------> [Aug 5 17:48:07] --- (11 headers 0 lines) --- [Aug 5 17:48:08] Really destroying SIP dialog '262c29f773ddec9f6e06235838eecfd1@222.222.22.22:0' Method: OPTIONS VICIbox10*CLI>

Outbound Manual Call Starts here

Code: Select all: [Aug 5 17:50:01] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:50:01] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:50:01] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:50:02] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:50:06] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:50:06] -- Called 8600051@default [Aug 5 17:50:06] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000008;2", "8600051,F") in new stack [Aug 5 17:50:06] -- Local/8600051@default-00000008;1 answered [Aug 5 17:50:06] -- Executing [916317918378@default:1] AGI("Local/8600051@default-00000008;1", "agi://127.0.0.1:4577/call_log") in new stack [Aug 5 17:50:06] -- <Local/8600051@default-00000008;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Aug 5 17:50:06] -- Executing [916317918378@default:2] Dial("Local/8600051@default-00000008;1", "SIP/CARRIERNAME/6317918378,,tTor") in new stack [Aug 5 17:50:06] ERROR[19619][C-00000012]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("VICIbox10", "(null)", ...): Name or service not known [Aug 5 17:50:06] WARNING[19619][C-00000012]: acl.c:892 resolve_first: Unable to lookup 'VICIbox10' [Aug 5 17:50:06] == Using SIP RTP CoS mark 5 [Aug 5 17:50:06] Audio is at 15546 [Aug 5 17:50:06] Adding codec ulaw to SDP [Aug 5 17:50:06] Adding non-codec 0x1 (telephone-event) to SDP [Aug 5 17:50:06] Reliably Transmitting (NAT) to 111.111.11.11:5060: [Aug 5 17:50:06] INVITE sip:6317918378@carrier.domain.com SIP/2.0 [Aug 5 17:50:06] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK454a16a2;rport [Aug 5 17:50:06] Max-Forwards: 70 [Aug 5 17:50:06] From: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:06] To: <sip:6317918378@carrier.domain.com> [Aug 5 17:50:06] Contact: <sip:0000000000@222.222.22.22:5060> [Aug 5 17:50:06] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:06] CSeq: 102 INVITE [Aug 5 17:50:06] User-Agent: Asterisk PBX 13.38.2-vici [Aug 5 17:50:06] Date: Fri, 05 Aug 2022 21:50:06 GMT [Aug 5 17:50:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:06] Supported: replaces, timer [Aug 5 17:50:06] Remote-Party-ID: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;party=calling;privacy=off;screen=no [Aug 5 17:50:06] Content-Type: application/sdp [Aug 5 17:50:06] Content-Length: 259 [Aug 5 17:50:06] [Aug 5 17:50:06] v=0 [Aug 5 17:50:06] o=root 1654452018 1654452018 IN IP4 222.222.22.22 [Aug 5 17:50:06] s=Asterisk PBX 13.38.2-vici [Aug 5 17:50:06] c=IN IP4 222.222.22.22 [Aug 5 17:50:06] t=0 0 [Aug 5 17:50:06] m=audio 15546 RTP/AVP 0 101 [Aug 5 17:50:06] a=rtpmap:0 PCMU/8000 [Aug 5 17:50:06] a=rtpmap:101 telephone-event/8000 [Aug 5 17:50:06] a=fmtp:101 0-16 [Aug 5 17:50:06] a=ptime:20 [Aug 5 17:50:06] a=maxptime:150 [Aug 5 17:50:06] a=sendrecv [Aug 5 17:50:06] [Aug 5 17:50:06] --- [Aug 5 17:50:06] -- Called SIP/CARRIERNAME/6317918378 [Aug 5 17:50:06] [Aug 5 17:50:06] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 5 17:50:06] SIP/2.0 100 Trying [Aug 5 17:50:06] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK454a16a2;received=222.222.22.22;rport=5060 [Aug 5 17:50:06] From: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:06] To: <sip:6317918378@carrier.domain.com> [Aug 5 17:50:06] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:06] CSeq: 102 INVITE [Aug 5 17:50:06] Server: carrier.com [Aug 5 17:50:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:06] Supported: replaces, timer [Aug 5 17:50:06] Session-Expires: 1800;refresher=uas [Aug 5 17:50:06] Contact: <sip:6317918378@111.111.11.11:5060> [Aug 5 17:50:06] Content-Length: 0 [Aug 5 17:50:06] [Aug 5 17:50:06] <-------------> [Aug 5 17:50:06] --- (12 headers 0 lines) --- [Aug 5 17:50:06] [Aug 5 17:50:06] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 5 17:50:06] SIP/2.0 200 OK [Aug 5 17:50:06] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK454a16a2;received=222.222.22.22;rport=5060 [Aug 5 17:50:06] From: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:06] To: <sip:6317918378@carrier.domain.com>;tag=as4cb9d055 [Aug 5 17:50:06] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:06] CSeq: 102 INVITE [Aug 5 17:50:06] Server: carrier.com [Aug 5 17:50:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:06] Supported: replaces, timer [Aug 5 17:50:06] Session-Expires: 1800;refresher=uas [Aug 5 17:50:06] Contact: <sip:6317918378@111.111.11.11:5060> [Aug 5 17:50:06] Content-Type: application/sdp [Aug 5 17:50:06] Require: timer [Aug 5 17:50:06] Content-Length: 225 [Aug 5 17:50:06] [Aug 5 17:50:06] v=0 [Aug 5 17:50:06] o=root 1439991136 1439991136 IN IP4 111.111.11.11 [Aug 5 17:50:06] s=carrier.com [Aug 5 17:50:06] c=IN IP4 111.111.11.11 [Aug 5 17:50:06] t=0 0 [Aug 5 17:50:06] m=audio 18568 RTP/AVP 0 101 [Aug 5 17:50:06] a=rtpmap:0 PCMU/8000 [Aug 5 17:50:06] a=rtpmap:101 telephone-event/8000 [Aug 5 17:50:06] a=fmtp:101 0-16 [Aug 5 17:50:06] a=ptime:20 [Aug 5 17:50:06] a=sendrecv [Aug 5 17:50:06] <-------------> [Aug 5 17:50:06] --- (14 headers 11 lines) --- [Aug 5 17:50:06] Got SDP version 1439991136 and unique parts [root 1439991136 IN IP4 111.111.11.11] [Aug 5 17:50:06] Found RTP audio format 0 [Aug 5 17:50:06] Found RTP audio format 101 [Aug 5 17:50:06] Found audio description format PCMU for ID 0 [Aug 5 17:50:06] Found audio description format telephone-event for ID 101 [Aug 5 17:50:06] Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Aug 5 17:50:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Aug 5 17:50:06] > 0x7fa80001ff00 -- Strict RTP learning after remote address set to: 111.111.11.11:18568 [Aug 5 17:50:06] Peer audio RTP is at port 111.111.11.11:18568 [Aug 5 17:50:06] sip_route_dump: route/path hop: <sip:6317918378@111.111.11.11:5060> [Aug 5 17:50:06] Transmitting (NAT) to 111.111.11.11:5060: [Aug 5 17:50:06] ACK sip:6317918378@111.111.11.11:5060 SIP/2.0 [Aug 5 17:50:06] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK342660cd;rport [Aug 5 17:50:06] Max-Forwards: 70 [Aug 5 17:50:06] From: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:06] To: <sip:6317918378@carrier.domain.com>;tag=as4cb9d055 [Aug 5 17:50:06] Contact: <sip:0000000000@222.222.22.22:5060> [Aug 5 17:50:06] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:06] CSeq: 102 ACK [Aug 5 17:50:06] User-Agent: Asterisk PBX 13.38.2-vici [Aug 5 17:50:06] Content-Length: 0 [Aug 5 17:50:06] [Aug 5 17:50:06] [Aug 5 17:50:06] --- [Aug 5 17:50:06] -- SIP/CARRIERNAME-00000009 answered Local/8600051@default-00000008;1 [Aug 5 17:50:06] -- Channel SIP/CARRIERNAME-00000009 joined 'simple_bridge' basic-bridge <b68b50d4-4dfd-4433-9279-d988aec11ab9> [Aug 5 17:50:06] -- Channel Local/8600051@default-00000008;1 joined 'simple_bridge' basic-bridge <b68b50d4-4dfd-4433-9279-d988aec11ab9> [Aug 5 17:50:06] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 5 17:50:06] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:50:07] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 5 17:50:07] > 0x7fa80001ff00 -- Strict RTP switching to RTP target address 111.111.11.11:18568 as source [Aug 5 17:50:07] Reliably Transmitting (NAT) to 33.333.333.333:55938: [Aug 5 17:50:07] OPTIONS sip:t7atthm2@192.0.2.100;transport=wss SIP/2.0 [Aug 5 17:50:07] Via: SIP/2.0/WS 222.222.22.22:0;branch=z9hG4bK01ba4d99;rport [Aug 5 17:50:07] Max-Forwards: 70 [Aug 5 17:50:07] From: "asterisk" <sip:asterisk@222.222.22.22:0>;tag=as20335448 [Aug 5 17:50:07] To: <sip:t7atthm2@192.0.2.100;transport=wss> [Aug 5 17:50:07] Contact: <sip:asterisk@222.222.22.22:0;transport=ws> [Aug 5 17:50:07] Call-ID: 5cf788b7480caad239d059dd5de67ad8@222.222.22.22:0 [Aug 5 17:50:07] CSeq: 102 OPTIONS [Aug 5 17:50:07] User-Agent: Asterisk PBX 13.38.2-vici [Aug 5 17:50:07] Date: Fri, 05 Aug 2022 21:50:07 GMT [Aug 5 17:50:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:07] Supported: replaces, timer [Aug 5 17:50:07] Content-Length: 0 [Aug 5 17:50:07] [Aug 5 17:50:07] [Aug 5 17:50:07] --- [Aug 5 17:50:07] [Aug 5 17:50:07] <--- SIP read from WS:33.333.333.333:55938 ---> [Aug 5 17:50:07] SIP/2.0 200 OK [Aug 5 17:50:07] Via: SIP/2.0/WS 222.222.22.22:0;branch=z9hG4bK01ba4d99;rport [Aug 5 17:50:07] To: <sip:t7atthm2@192.0.2.100;transport=wss>;tag=r4g8kcr0gl [Aug 5 17:50:07] From: "asterisk" <sip:asterisk@222.222.22.22:0>;tag=as20335448 [Aug 5 17:50:07] Call-ID: 5cf788b7480caad239d059dd5de67ad8@222.222.22.22:0 [Aug 5 17:50:07] CSeq: 102 OPTIONS [Aug 5 17:50:07] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [Aug 5 17:50:07] Accept: application/sdp,application/dtmf-relay [Aug 5 17:50:07] Supported: outbound [Aug 5 17:50:07] User-Agent: VICIphone 1.0-rc1 [Aug 5 17:50:07] Content-Length: 0 [Aug 5 17:50:07] [Aug 5 17:50:07] <-------------> [Aug 5 17:50:07] --- (11 headers 0 lines) --- [Aug 5 17:50:08] Really destroying SIP dialog '5cf788b7480caad239d059dd5de67ad8@222.222.22.22:0' Method: OPTIONS [Aug 5 17:50:11] > 0x7fa80001ff00 -- Strict RTP learning complete - Locking on source address 111.111.11.11:18568 [Aug 5 17:50:13] [Aug 5 17:50:13] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 5 17:50:13] BYE sip:0000000000@222.222.22.22:5060 SIP/2.0 [Aug 5 17:50:13] Via: SIP/2.0/UDP 111.111.11.11:5060;branch=z9hG4bK60dd4fa5;rport [Aug 5 17:50:13] Max-Forwards: 70 [Aug 5 17:50:13] From: <sip:6317918378@carrier.domain.com>;tag=as4cb9d055 [Aug 5 17:50:13] To: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:13] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:13] CSeq: 102 BYE [Aug 5 17:50:13] User-Agent: carrier.com [Aug 5 17:50:13] X-Asterisk-HangupCause: Unknown [Aug 5 17:50:13] X-Asterisk-HangupCauseCode: 0 [Aug 5 17:50:13] Content-Length: 0 [Aug 5 17:50:13] [Aug 5 17:50:13] <-------------> [Aug 5 17:50:13] --- (11 headers 0 lines) --- [Aug 5 17:50:13] Sending to 111.111.11.11:5060 (NAT) [Aug 5 17:50:13] Scheduling destruction of SIP dialog '31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060' in 6400 ms (Method: BYE) [Aug 5 17:50:13] [Aug 5 17:50:13] <--- Transmitting (NAT) to 111.111.11.11:5060 ---> [Aug 5 17:50:13] SIP/2.0 200 OK [Aug 5 17:50:13] Via: SIP/2.0/UDP 111.111.11.11:5060;branch=z9hG4bK60dd4fa5;received=111.111.11.11;rport=5060 [Aug 5 17:50:13] From: <sip:6317918378@carrier.domain.com>;tag=as4cb9d055 [Aug 5 17:50:13] To: "DV736180W1000100010W" <sip:0000000000@222.222.22.22>;tag=as60b619d3 [Aug 5 17:50:13] Call-ID: 31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060 [Aug 5 17:50:13] CSeq: 102 BYE [Aug 5 17:50:13] Server: Asterisk PBX 13.38.2-vici [Aug 5 17:50:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:13] Supported: replaces, timer [Aug 5 17:50:13] Content-Length: 0 [Aug 5 17:50:13] [Aug 5 17:50:13] [Aug 5 17:50:13] <------------> [Aug 5 17:50:13] -- Channel SIP/CARRIERNAME-00000009 left 'simple_bridge' basic-bridge <b68b50d4-4dfd-4433-9279-d988aec11ab9> [Aug 5 17:50:13] -- Channel Local/8600051@default-00000008;1 left 'simple_bridge' basic-bridge <b68b50d4-4dfd-4433-9279-d988aec11ab9> [Aug 5 17:50:13] == Spawn extension (default, 916317918378, 2) exited non-zero on 'Local/8600051@default-00000008;1' [Aug 5 17:50:13] -- Executing [h@default:1] AGI("Local/8600051@default-00000008;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----7-----SIP 200 OK)") in new stack [Aug 5 17:50:13] -- <Local/8600051@default-00000008;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----7-----SIP 200 OK) completed, returning 0 [Aug 5 17:50:13] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000008;2' [Aug 5 17:50:13] WARNING[19620][C-00000011]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Aug 5 17:50:13] -- Executing [h@default:1] AGI("Local/8600051@default-00000008;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Aug 5 17:50:13] -- <Local/8600051@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Aug 5 17:50:19] Really destroying SIP dialog '31ceb0242b5330b766adf02d266fb79d@222.222.22.22:5060' Method: BYE [Aug 5 17:50:24] Reliably Transmitting (NAT) to 111.111.11.11:5060: [Aug 5 17:50:24] OPTIONS sip:carrier.domain.com SIP/2.0 [Aug 5 17:50:24] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK7d15bd1b;rport [Aug 5 17:50:24] Max-Forwards: 70 [Aug 5 17:50:24] From: "asterisk" <sip:asterisk@222.222.22.22>;tag=as43b83d68 [Aug 5 17:50:24] To: <sip:carrier.domain.com> [Aug 5 17:50:24] Contact: <sip:asterisk@222.222.22.22:5060> [Aug 5 17:50:24] Call-ID: 5a75f08b74d543837bda6bd551e5061a@222.222.22.22:5060 [Aug 5 17:50:24] CSeq: 102 OPTIONS [Aug 5 17:50:24] User-Agent: Asterisk PBX 13.38.2-vici [Aug 5 17:50:24] Date: Fri, 05 Aug 2022 21:50:24 GMT [Aug 5 17:50:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:24] Supported: replaces, timer [Aug 5 17:50:24] Content-Length: 0 [Aug 5 17:50:24] [Aug 5 17:50:24] [Aug 5 17:50:24] --- [Aug 5 17:50:24] [Aug 5 17:50:24] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 5 17:50:24] SIP/2.0 200 OK [Aug 5 17:50:24] Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK7d15bd1b;received=222.222.22.22;rport=5060 [Aug 5 17:50:24] From: "asterisk" <sip:asterisk@222.222.22.22>;tag=as43b83d68 [Aug 5 17:50:24] To: <sip:carrier.domain.com>;tag=as6e370179 [Aug 5 17:50:24] Call-ID: 5a75f08b74d543837bda6bd551e5061a@222.222.22.22:5060 [Aug 5 17:50:24] CSeq: 102 OPTIONS [Aug 5 17:50:24] Server: carrier.com [Aug 5 17:50:24] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 5 17:50:24] Supported: replaces, timer [Aug 5 17:50:24] Contact: <sip:111.111.11.11:5060> [Aug 5 17:50:24] Accept: application/sdp [Aug 5 17:50:24] Content-Length: 0 [Aug 5 17:50:24] [Aug 5 17:50:24] <-------------> [Aug 5 17:50:24] --- (12 headers 0 lines) --- [Aug 5 17:50:24] Really destroying SIP dialog '5a75f08b74d543837bda6bd551e5061a@222.222.22.22:5060' Method: OPTIONS VICIbox10*CLI>

I wasn't able to find any similar problems on the forum so please help me figure this out if you can.

Thank you!

by **striker** » Mon Aug 08, 2022 10:11 am

seems the message is from the Trunk provider, accordign to your log , vicidial dials through the CARRIER and getting answered
"SIP/CARRIERNAME-00000009 answered Local/8600051@default-00000008;1"

either you are dialing wrong number or Trunk provider doesnt have route to that number to accept.
better check with TRUNk vedor..

Just a try
register the same account in a softphone and check wheter the call goes through sucessfull

by **bronson** » Mon Aug 08, 2022 5:00 pm

striker wrote:seems the message is from the Trunk provider, accordign to your log , vicidial dials through the CARRIER and getting answered
"SIP/CARRIERNAME-00000009 answered Local/8600051@default-00000008;1"

either you are dialing wrong number or Trunk provider doesnt have route to that number to accept.
better check with TRUNk vedor..

Just a try
register the same account in a softphone and check wheter the call goes through sucessfull

Thanks Striker,
I forgot to register my new server IP with my carrier. I've done that and now when I make a call I hear a short fast whistle sound and the call is immediately disconnected. any ideas?

Code: Select all: [Aug 8 17:53:21] == Manager 'sendcron' logged on from 127.0.0.1 [Aug 8 17:53:21] -- Called 8600051@default [Aug 8 17:53:21] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000004;2", "8600051,F") in new stack [Aug 8 17:53:21] -- Local/8600051@default-00000004;1 answered [Aug 8 17:53:21] -- Executing [916317918378@default:1] AGI("Local/8600051@default-00000004;1", "agi://127.0.0.1:4577/call_log") in new stack [Aug 8 17:53:21] -- <Local/8600051@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Aug 8 17:53:21] -- Executing [916317918378@default:2] Dial("Local/8600051@default-00000004;1", "SIP/CARRIER/6317918378,,tTor") in new stack [Aug 8 17:53:21] == Using SIP RTP CoS mark 5 [Aug 8 17:53:21] Audio is at 12546 [Aug 8 17:53:21] Adding codec ulaw to SDP [Aug 8 17:53:21] Adding non-codec 0x1 (telephone-event) to SDP [Aug 8 17:53:21] Reliably Transmitting (NAT) to 111.111.11.11:5060: [Aug 8 17:53:21] INVITE sip:6317918378@111.111.11.11 SIP/2.0 [Aug 8 17:53:21] Via: SIP/2.0/UDP 222.222.2.222:5060;branch=z9hG4bK246e6b37;rport [Aug 8 17:53:21] Max-Forwards: 70 [Aug 8 17:53:21] From: "DV995593W1000100010W" <sip:0000000000@222.222.2.222>;tag=as3be3b831 [Aug 8 17:53:21] To: <sip:6317918378@111.111.11.11> [Aug 8 17:53:21] Contact: <sip:0000000000@222.222.2.222:5060> [Aug 8 17:53:21] Call-ID: 441b536d44aacf0767fdda3b039a9192@222.222.2.222:5060 [Aug 8 17:53:21] CSeq: 102 INVITE [Aug 8 17:53:21] User-Agent: Asterisk PBX 13.38.2-vici [Aug 8 17:53:21] Date: Mon, 08 Aug 2022 21:53:21 GMT [Aug 8 17:53:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 8 17:53:21] Supported: replaces, timer [Aug 8 17:53:21] Remote-Party-ID: "DV995593W1000100010W" <sip:0000000000@222.222.2.222>;party=calling;privacy=off;screen=no [Aug 8 17:53:21] Content-Type: application/sdp [Aug 8 17:53:21] Content-Length: 259 [Aug 8 17:53:21] [Aug 8 17:53:21] v=0 [Aug 8 17:53:21] o=root 1632616730 1632616730 IN IP4 222.222.2.222 [Aug 8 17:53:21] s=Asterisk PBX 13.38.2-vici [Aug 8 17:53:21] c=IN IP4 222.222.2.222 [Aug 8 17:53:21] t=0 0 [Aug 8 17:53:21] m=audio 12546 RTP/AVP 0 101 [Aug 8 17:53:21] a=rtpmap:0 PCMU/8000 [Aug 8 17:53:21] a=rtpmap:101 telephone-event/8000 [Aug 8 17:53:21] a=fmtp:101 0-16 [Aug 8 17:53:21] a=ptime:20 [Aug 8 17:53:21] a=maxptime:150 [Aug 8 17:53:21] a=sendrecv [Aug 8 17:53:21] [Aug 8 17:53:21] --- [Aug 8 17:53:21] -- Called SIP/CARRIER/6317918378 [Aug 8 17:53:21] [Aug 8 17:53:21] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 8 17:53:21] SIP/2.0 100 Trying [Aug 8 17:53:21] Via: SIP/2.0/UDP 222.222.2.222:5060;branch=z9hG4bK246e6b37;received=222.222.2.222;rport=5060 [Aug 8 17:53:21] From: "DV995593W1000100010W" <sip:0000000000@222.222.2.222>;tag=as3be3b831 [Aug 8 17:53:21] To: <sip:6317918378@111.111.11.11> [Aug 8 17:53:21] Call-ID: 441b536d44aacf0767fdda3b039a9192@222.222.2.222:5060 [Aug 8 17:53:21] CSeq: 102 INVITE [Aug 8 17:53:21] Server: carrier.com [Aug 8 17:53:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 8 17:53:21] Supported: replaces, timer [Aug 8 17:53:21] Session-Expires: 1800;refresher=uas [Aug 8 17:53:21] Contact: <sip:6317918378@111.111.11.11:5060> [Aug 8 17:53:21] Content-Length: 0 [Aug 8 17:53:21] [Aug 8 17:53:21] <-------------> [Aug 8 17:53:21] --- (12 headers 0 lines) --- [Aug 8 17:53:21] [Aug 8 17:53:21] <--- SIP read from UDP:111.111.11.11:5060 ---> [Aug 8 17:53:21] SIP/2.0 503 Service Unavailable [Aug 8 17:53:21] Via: SIP/2.0/UDP 222.222.2.222:5060;branch=z9hG4bK246e6b37;received=222.222.2.222;rport=5060 [Aug 8 17:53:21] From: "DV995593W1000100010W" <sip:0000000000@222.222.2.222>;tag=as3be3b831 [Aug 8 17:53:21] To: <sip:6317918378@111.111.11.11>;tag=as1fdb461c [Aug 8 17:53:21] Call-ID: 441b536d44aacf0767fdda3b039a9192@222.222.2.222:5060 [Aug 8 17:53:21] CSeq: 102 INVITE [Aug 8 17:53:21] Server: carrier.com [Aug 8 17:53:21] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Aug 8 17:53:21] Supported: replaces, timer [Aug 8 17:53:21] Session-Expires: 1800;refresher=uas [Aug 8 17:53:21] Content-Length: 0 [Aug 8 17:53:21] [Aug 8 17:53:21] <-------------> [Aug 8 17:53:21] --- (11 headers 0 lines) --- [Aug 8 17:53:21] -- Got SIP response 503 "Service Unavailable" back from 111.111.11.11:5060 [Aug 8 17:53:21] Transmitting (NAT) to 111.111.11.11:5060: [Aug 8 17:53:21] ACK sip:6317918378@111.111.11.11 SIP/2.0 [Aug 8 17:53:21] Via: SIP/2.0/UDP 222.222.2.222:5060;branch=z9hG4bK246e6b37;rport [Aug 8 17:53:21] Max-Forwards: 70 [Aug 8 17:53:21] From: "DV995593W1000100010W" <sip:0000000000@222.222.2.222>;tag=as3be3b831 [Aug 8 17:53:21] To: <sip:6317918378@111.111.11.11>;tag=as1fdb461c [Aug 8 17:53:21] Contact: <sip:0000000000@222.222.2.222:5060> [Aug 8 17:53:21] Call-ID: 441b536d44aacf0767fdda3b039a9192@222.222.2.222:5060 [Aug 8 17:53:21] CSeq: 102 ACK [Aug 8 17:53:21] User-Agent: Asterisk PBX 13.38.2-vici [Aug 8 17:53:21] Content-Length: 0 [Aug 8 17:53:21] [Aug 8 17:53:21] [Aug 8 17:53:21] --- [Aug 8 17:53:21] -- SIP/CARRIER-00000005 is circuit-busy [Aug 8 17:53:21] == Everyone is busy/congested at this time (1:0/1/0) [Aug 8 17:53:21] -- Executing [916317918378@default:3] Hangup("Local/8600051@default-00000004;1", "") in new stack [Aug 8 17:53:21] == Spawn extension (default, 916317918378, 3) exited non-zero on 'Local/8600051@default-00000004;1' [Aug 8 17:53:21] -- Executing [h@default:1] AGI("Local/8600051@default-00000004;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------------SIP 503 Service Unavailable)") in new stack [Aug 8 17:53:21] -- <Local/8600051@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION---------------SIP 503 Service Unavailable) completed, returning 0 [Aug 8 17:53:21] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000004;2' [Aug 8 17:53:21] WARNING[14140][C-00000008]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Aug 8 17:53:21] -- Executing [h@default:1] AGI("Local/8600051@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34--------------------)") in new stack [Aug 8 17:53:21] -- <Local/8600051@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34--------------------) completed, returning 0 [Aug 8 17:53:22] == Manager 'sendcron' logged off from 127.0.0.1 [Aug 8 17:53:22] Really destroying SIP dialog '441b536d44aacf0767fdda3b039a9192@222.222.2.222:5060' Method: INVITE VICIbox10*CLI>

*edited to add sip debug output

by **striker** » Tue Aug 09, 2022 12:56 am

[Aug 8 17:53:21] -- SIP/CARRIER-00000005 is circuit-busy
[Aug 8 17:53:21] == Everyone is busy/congested at this time
503 Service Unavailable
your carrier rejects the call, either check the number you have dialled or carrier service is not working fine
check with carrier why reject the call,

by **bronson** » Tue Aug 09, 2022 4:10 pm

striker wrote:[Aug 8 17:53:21] -- SIP/CARRIER-00000005 is circuit-busy
[Aug 8 17:53:21] == Everyone is busy/congested at this time
503 Service Unavailable
your carrier rejects the call, either check the number you have dialled or carrier service is not working fine
check with carrier why reject the call,

Ok, my carrier replied back that the problem was my caller ID.

Hi Bronson

I can see some calls from yesterday failed, tho it seems your calls are sending as caller ID just 0000000000, it is not a valid number, in order to make outbound calls, make sure your call request send a valid 10 digit number from USA or Canada, and let us know the results.

Regards

I updated my caller ID so now calls are connecting but now I have a new problem. The calls are connecting to international numbers instead of local US numbers that I dialed.

it appears that the calls are now connecting to international phone numbers. I received these email notifications from my carrier.

Account 272868_VICIbox10 made a call to Egypt, at number 2076883210 on 2022-08-09 16:47:37, which lasted 00:00:14 at a total cost of 0.06$. The call was made from IP 222.222.2.222.

Account 272868_VICIbox10 made a call to Turkey, at number 9072665145 on 2022-08-09 16:44:30, which lasted 00:00:17 at a total cost of 0.01$. The call was made from IP 222.222.2.222.

Account 272868_VICIbox10 made a call to Philippines, at number 6317918378 on 2022-08-09 16:41:16, which lasted 00:00:32 at a total cost of 0.09$. The call was made from IP 222.222.2.222.

but these are US numbers.

any idea why?

I am making a manual dial from the agent interface using 91 then phone number. ex: I call 912076883210 and it's connecting me to Egypt (which is +20 country code) but it should be US (+1) and the phone number.

by **bronson** » Tue Aug 09, 2022 5:26 pm

Disregard. It was an issue with my dialplan entry. I've resolved the issue. Thank you for your help!

vicidial.org

"that number has not yet been assigned" Error Message

"that number has not yet been assigned" Error Message

Re: "that number has not yet been assigned" Error Message

Re: "that number has not yet been assigned" Error Message

Re: "that number has not yet been assigned" Error Message

Re: "that number has not yet been assigned" Error Message

Re: "that number has not yet been assigned" Error Message

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