vicidial.org

Posted: **Fri Apr 24, 2020 10:45 am**

Where is [default] ? I looked in sip.conf, extensions.conf and rtp.conf, and i cant find where its telling me..

[Apr 23 19:24:28] NOTICE[4297][C-0000000f]: chan_sip.c:26515 handle_request_invite: Call from '1002' (10.0.0.58:5060) to extension '95494572' rejected because extension not found in context 'default'.
[Apr 23 19:24:28] Scheduling destruction of SIP dialog 'M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI.' in 6400 ms (Method: INVITE)
[Apr 23 19:24:28] Retransmitting #1 (NAT) to 10.0.0.58:5060:
[Apr 23 19:24:28] SIP/2.0 404 Not Found
[Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z-;received=10.0.0.58;rport=5060
[Apr 23 19:24:28] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41
[Apr 23 19:24:28] To: <sip:95497572@10.0.0.62;transport=UDP>;tag=as1b52db67
[Apr 23 19:24:28] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI.
[Apr 23 19:24:28] CSeq: 2 INVITE
[Apr 23 19:24:28] Server: Asterisk PBX 13.29.2-vici
[Apr 23 19:24:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Apr 23 19:24:28] Supported: replaces, timer
[Apr 23 19:24:28] Content-Length: 0

Full log here:

Code: Select all: <--- SIP read from UDP:10.0.0.55:5061 ---> [Apr 23 19:23:49] [Apr 23 19:23:49] [Apr 23 19:23:49] <-------------> [Apr 23 19:23:57] Reliably Transmitting (NAT) to 10.0.0.55:5061: [Apr 23 19:23:57] OPTIONS sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP SIP/2.0 [Apr 23 19:23:57] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK71f36df9;rport [Apr 23 19:23:57] Max-Forwards: 70 [Apr 23 19:23:57] From: "asterisk" <sip:asterisk@10.0.0.62>;tag=as124a2094 [Apr 23 19:23:57] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 19:23:57] Contact: <sip:asterisk@10.0.0.62:5060> [Apr 23 19:23:57] Call-ID: 498f9e3d47c3fe456a29e990512a1b22@10.0.0.62:5060 [Apr 23 19:23:57] CSeq: 102 OPTIONS [Apr 23 19:23:57] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 19:23:57] Date: Thu, 23 Apr 2020 23:23:57 GMT [Apr 23 19:23:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 19:23:57] Supported: replaces, timer [Apr 23 19:23:57] Content-Length: 0 [Apr 23 19:23:57] [Apr 23 19:23:57] [Apr 23 19:23:57] --- [Apr 23 19:23:57] [Apr 23 19:23:57] <--- SIP read from UDP:10.0.0.55:5061 ---> [Apr 23 19:23:57] SIP/2.0 200 OK [Apr 23 19:23:57] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK71f36df9;rport=5060 [Apr 23 19:23:57] Contact: <sip:10.0.0.55:5061> [Apr 23 19:23:57] To: <sip:1001@73.46.29.100:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=5438dc21 [Apr 23 19:23:57] From: "asterisk"<sip:asterisk@10.0.0.62>;tag=as124a2094 [Apr 23 19:23:57] Call-ID: 498f9e3d47c3fe456a29e990512a1b22@10.0.0.62:5060 [Apr 23 19:23:57] CSeq: 102 OPTIONS [Apr 23 19:23:57] Accept: application/sdp, application/sdp [Apr 23 19:23:57] Accept-Language: en [Apr 23 19:23:57] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Apr 23 19:23:57] User-Agent: Zoiper rev.6751 [Apr 23 19:23:57] Allow-Events: presence [Apr 23 19:23:57] Content-Length: 0 [Apr 23 19:23:57] [Apr 23 19:24:27] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:27] INVITE sip:9549477572@10.0.0.62;transport=UDP SIP/2.0 [Apr 23 19:24:27] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-fa523b8abf4e35a1-1---d8754z- [Apr 23 19:24:27] Max-Forwards: 70 [Apr 23 19:24:27] Contact: <sip:1002@10.0.0.58:5060;transport=UDP> [Apr 23 19:24:27] To: <sip:9549477572@10.0.0.62;transport=UDP> [Apr 23 19:24:27] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:27] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] CSeq: 1 INVITE [Apr 23 19:24:27] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Apr 23 19:24:27] Content-Type: application/sdp [Apr 23 19:24:27] User-Agent: Zoiper rev.6751 [Apr 23 19:24:27] Content-Length: 321 [Apr 23 19:24:27] [Apr 23 19:24:27] v=0 [Apr 23 19:24:27] o=Zoiper_user 0 0 IN IP4 10.0.0.58 [Apr 23 19:24:27] s=Zoiper_session [Apr 23 19:24:27] c=IN IP4 10.0.0.58 [Apr 23 19:24:27] t=0 0 [Apr 23 19:24:27] m=audio 8000 RTP/AVP 3 8 110 98 0 101 [Apr 23 19:24:27] a=rtpmap:3 GSM/8000 [Apr 23 19:24:27] a=rtpmap:8 PCMA/8000 [Apr 23 19:24:27] a=rtpmap:110 speex/8000 [Apr 23 19:24:27] a=rtpmap:98 iLBC/8000 [Apr 23 19:24:27] a=fmtp:98 mode=30 [Apr 23 19:24:27] a=rtpmap:0 PCMU/8000 [Apr 23 19:24:27] a=rtpmap:101 telephone-event/8000 [Apr 23 19:24:27] a=fmtp:101 0-15 [Apr 23 19:24:27] a=sendrecv [Apr 23 19:24:27] <-------------> [Apr 23 19:24:27] --- (12 headers 15 lines) --- [Apr 23 19:24:27] Sending to 10.0.0.58:5060 (NAT) [Apr 23 19:24:27] Sending to 10.0.0.58:5060 (NAT) [Apr 23 19:24:27] Using INVITE request as basis request - M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] Found peer '1002' for '1002' from 10.0.0.58:5060 [Apr 23 19:24:27] [Apr 23 19:24:27] <--- Reliably Transmitting (NAT) to 10.0.0.58:5060 ---> [Apr 23 19:24:27] SIP/2.0 401 Unauthorized [Apr 23 19:24:27] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-fa523b8abf4e35a1-1---d8754z-;received=10.0.0.58;rport=5060 [Apr 23 19:24:27] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:27] To: <sip:9549477572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:27] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] CSeq: 1 INVITE [Apr 23 19:24:27] Server: Asterisk PBX 13.29.2-vici [Apr 23 19:24:27] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 19:24:27] Supported: replaces, timer [Apr 23 19:24:27] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3d559723" [Apr 23 19:24:27] Content-Length: 0 [Apr 23 19:24:27] [Apr 23 19:24:27] [Apr 23 19:24:27] <------------> [Apr 23 19:24:27] Scheduling destruction of SIP dialog 'M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI.' in 6400 ms (Method: INVITE) [Apr 23 19:24:27] [Apr 23 19:24:27] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:27] ACK sip:9549477572@10.0.0.62;transport=UDP SIP/2.0 [Apr 23 19:24:27] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-fa523b8abf4e35a1-1---d8754z- [Apr 23 19:24:27] Max-Forwards: 70 [Apr 23 19:24:27] To: <sip:9549477572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:27] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:27] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] CSeq: 1 ACK [Apr 23 19:24:27] Content-Length: 0 [Apr 23 19:24:27] [Apr 23 19:24:27] <-------------> [Apr 23 19:24:27] --- (8 headers 0 lines) --- [Apr 23 19:24:27] [Apr 23 19:24:27] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:27] INVITE sip:9549477572@10.0.0.62;transport=UDP SIP/2.0 [Apr 23 19:24:27] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z- [Apr 23 19:24:27] Max-Forwards: 70 [Apr 23 19:24:27] Contact: <sip:1002@10.0.0.58:5060;transport=UDP> [Apr 23 19:24:27] To: <sip:9549477572@10.0.0.62;transport=UDP> [Apr 23 19:24:27] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:27] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] CSeq: 2 INVITE [Apr 23 19:24:27] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Apr 23 19:24:27] Content-Type: application/sdp [Apr 23 19:24:27] User-Agent: Zoiper rev.6751 [Apr 23 19:24:27] Authorization: Digest username="1002",realm="asterisk",nonce="3d559723",uri="sip:9549477572@10.0.0.62;transport=UDP",response="d0f9d96721c0977a1811b8cf9066892e",algorithm=MD5 [Apr 23 19:24:27] Content-Length: 321 [Apr 23 19:24:27] [Apr 23 19:24:27] v=0 [Apr 23 19:24:27] o=Zoiper_user 0 0 IN IP4 10.0.0.58 [Apr 23 19:24:27] s=Zoiper_session [Apr 23 19:24:27] c=IN IP4 10.0.0.58 [Apr 23 19:24:27] t=0 0 [Apr 23 19:24:27] m=audio 8000 RTP/AVP 3 8 110 98 0 101 [Apr 23 19:24:27] a=rtpmap:3 GSM/8000 [Apr 23 19:24:27] a=rtpmap:8 PCMA/8000 [Apr 23 19:24:27] a=rtpmap:110 speex/8000 [Apr 23 19:24:27] a=rtpmap:98 iLBC/8000 [Apr 23 19:24:27] a=fmtp:98 mode=30 [Apr 23 19:24:27] a=rtpmap:0 PCMU/8000 [Apr 23 19:24:27] a=rtpmap:101 telephone-event/8000 [Apr 23 19:24:27] a=fmtp:101 0-15 [Apr 23 19:24:27] a=sendrecv [Apr 23 19:24:27] <-------------> [Apr 23 19:24:27] --- (13 headers 15 lines) --- [Apr 23 19:24:27] Sending to 10.0.0.58:5060 (NAT) [Apr 23 19:24:27] Using INVITE request as basis request - M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:27] Found peer '1002' for '1002' from 10.0.0.58:5060 [Apr 23 19:24:28] ERROR[4297][C-0000000f]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("vicibox9", "(null)", ...): Name or service not known [Apr 23 19:24:28] WARNING[4297][C-0000000f]: acl.c:835 resolve_first: Unable to lookup 'vicibox9' [Apr 23 19:24:28] == Using SIP RTP CoS mark 5 [Apr 23 19:24:28] Found RTP audio format 3 [Apr 23 19:24:28] Found RTP audio format 8 [Apr 23 19:24:28] Found RTP audio format 110 [Apr 23 19:24:28] Found RTP audio format 98 [Apr 23 19:24:28] Found RTP audio format 0 [Apr 23 19:24:28] Found RTP audio format 101 [Apr 23 19:24:28] Found audio description format GSM for ID 3 [Apr 23 19:24:28] Found audio description format PCMA for ID 8 [Apr 23 19:24:28] Found audio description format speex for ID 110 [Apr 23 19:24:28] Found audio description format iLBC for ID 98 [Apr 23 19:24:28] Found audio description format PCMU for ID 0 [Apr 23 19:24:28] Found audio description format telephone-event for ID 101 [Apr 23 19:24:28] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|gsm) [Apr 23 19:24:28] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 23 19:24:28] > 0x7fa70c075cb0 -- Strict RTP learning after remote address set to: 10.0.0.58:8000 [Apr 23 19:24:28] Peer audio RTP is at port 10.0.0.58:8000 [Apr 23 19:24:28] Looking for 9549477572 in default (domain 10.0.0.62) [Apr 23 19:24:28] [Apr 23 19:24:28] <--- Reliably Transmitting (NAT) to 10.0.0.58:5060 ---> [Apr 23 19:24:28] SIP/2.0 404 Not Found [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z-;received=10.0.0.58;rport=5060 [Apr 23 19:24:28] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:28] To: <sip:9549477572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:28] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:28] CSeq: 2 INVITE [Apr 23 19:24:28] Server: Asterisk PBX 13.29.2-vici [Apr 23 19:24:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 19:24:28] Supported: replaces, timer [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] [Apr 23 19:24:28] <------------> [Apr 23 19:24:28] NOTICE[4297][C-0000000f]: chan_sip.c:26515 handle_request_invite: Call from '1002' (10.0.0.58:5060) to extension '9549477572' rejected because extension not found in context 'default'. [Apr 23 19:24:28] Scheduling destruction of SIP dialog 'M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI.' in 6400 ms (Method: INVITE) [Apr 23 19:24:28] Retransmitting #1 (NAT) to 10.0.0.58:5060: [Apr 23 19:24:28] SIP/2.0 404 Not Found [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z-;received=10.0.0.58;rport=5060 [Apr 23 19:24:28] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:28] To: <sip:9549477572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:28] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:28] CSeq: 2 INVITE [Apr 23 19:24:28] Server: Asterisk PBX 13.29.2-vici [Apr 23 19:24:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 19:24:28] Supported: replaces, timer [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] [Apr 23 19:24:28] --- [Apr 23 19:24:28] [Apr 23 19:24:28] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:28] ACK sip:9549477572@10.0.0.62;transport=UDP SIP/2.0 [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z- [Apr 23 19:24:28] Max-Forwards: 70 [Apr 23 19:24:28] To: <sip:9549477572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:28] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:28] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:28] CSeq: 2 ACK [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] <-------------> [Apr 23 19:24:28] --- (8 headers 0 lines) --- [Apr 23 19:24:28] [Apr 23 19:24:28] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:28] ACK sip:95494XX572@10.0.0.62;transport=UDP SIP/2.0 [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.58:5060;branch=z9hG4bK-d8754z-9878aec4f6889474-1---d8754z- [Apr 23 19:24:28] Max-Forwards: 70 [Apr 23 19:24:28] To: <sip:95494XX572@10.0.0.62;transport=UDP>;tag=as1b52db67 [Apr 23 19:24:28] From: "1002"<sip:1002@10.0.0.62;transport=UDP>;tag=403a6e41 [Apr 23 19:24:28] Call-ID: M2VmNDNiOTQ0NmRlY2NmNjhlYTNlNTdlOGRjNzMxODI. [Apr 23 19:24:28] CSeq: 2 ACK [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] <-------------> [Apr 23 19:24:28] --- (8 headers 0 lines) --- [Apr 23 19:24:28] Reliably Transmitting (NAT) to 10.0.0.58:5060: [Apr 23 19:24:28] OPTIONS sip:1002@10.0.0.58:5060;rinstance=8734d47f9c92f41b;transport=UDP SIP/2.0 [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK46842d46;rport [Apr 23 19:24:28] Max-Forwards: 70 [Apr 23 19:24:28] From: "asterisk" <sip:asterisk@10.0.0.62>;tag=as4d301149 [Apr 23 19:24:28] To: <sip:1002@10.0.0.58:5060;rinstance=8734d47f9c92f41b;transport=UDP> [Apr 23 19:24:28] Contact: <sip:asterisk@10.0.0.62:5060> [Apr 23 19:24:28] Call-ID: 541e8baf5592ec8b7b94fab30c4c1270@10.0.0.62:5060 [Apr 23 19:24:28] CSeq: 102 OPTIONS [Apr 23 19:24:28] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 19:24:28] Date: Thu, 23 Apr 2020 23:24:28 GMT [Apr 23 19:24:28] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 19:24:28] Supported: replaces, timer [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] [Apr 23 19:24:28] --- [Apr 23 19:24:28] [Apr 23 19:24:28] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 19:24:28] SIP/2.0 200 OK [Apr 23 19:24:28] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK46842d46;rport=5060 [Apr 23 19:24:28] Contact: <sip:10.0.0.58:5060> [Apr 23 19:24:28] To: <sip:1002@10.0.0.58:5060;rinstance=8734d47f9c92f41b;transport=UDP>;tag=882ce80c [Apr 23 19:24:28] From: "asterisk"<sip:asterisk@10.0.0.62>;tag=as4d301149 [Apr 23 19:24:28] Call-ID: 541e8baf5592ec8b7b94fab30c4c1270@10.0.0.62:5060 [Apr 23 19:24:28] CSeq: 102 OPTIONS [Apr 23 19:24:28] Accept: application/sdp, application/sdp [Apr 23 19:24:28] Accept-Language: en [Apr 23 19:24:28] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Apr 23 19:24:28] User-Agent: Zoiper rev.6751 [Apr 23 19:24:28] Allow-Events: presence [Apr 23 19:24:28] Content-Length: 0 [Apr 23 19:24:28] [Apr 23 19:24:28] <-------------> [Apr 23 19:24:28] --- (13 headers 0 lines) --- [Apr 23 19:24:28] Really destroying SIP dialog '541e8baf5592ec8b7b94fab30c4c1270@10.0.0.62:5060' Method: OPTIONS ERROR[4297][C-0000000f]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("vicibox9", "(null)", ...): Name or service not known -bash: syntax error near unexpected token `(' vicibox9:~ # [Apr 23 19:24:28] WARNING[4297][C-0000000f]: acl.c:835 resolve_first: Unable to lookup 'vicibox9' If '[Apr' is not a typo you can use command-not-found to lookup the package that contains it, like this: #################################################### == Spawn extension (default, 8600052, 1) exited non-zero on 'IAX2/2000-1653' [Apr 23 20:06:05] WARNING[18738][C-00000016]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Apr 23 20:06:05] -- Executing [h@default:1] AGI("IAX2/2000-1653", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack [Apr 23 20:06:05] -- <IAX2/2000-1653>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0 [Apr 23 20:06:05] -- Hungup 'IAX2/2000-1653' [Apr 23 20:06:05] Tx-Frame Retry[000] -- OSeqno: 028 ISeqno: 031 Type: IAX Subclass: HANGUP [Apr 23 20:06:05] Timestamp: 179520ms SCall: 01653 DCall: 00175 10.0.0.58:4569 [Apr 23 20:06:05] CAUSE CODE : 0 [Apr 23 20:06:05] [Apr 23 20:06:05] Rx-Frame Retry[ No] -- OSeqno: 031 ISeqno: 029 Type: IAX Subclass: ACK [Apr 23 20:06:05] Timestamp: 179520ms SCall: 00175 DCall: 01653 10.0.0.58:4569 [Apr 23 20:06:05] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 23 20:06:05] -- Called 55558600052@default [Apr 23 20:06:05] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-0000000b;2", "8600052,K") in new stack [Apr 23 20:06:05] WARNING[28078][C-00000017]: app_meetme.c:5261 admin_exec: Conference number '8600052' not found! [Apr 23 20:06:05] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-0000000b;2", "") in new stack [Apr 23 20:06:05] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-0000000b;2' [Apr 23 20:06:05] WARNING[28078][C-00000017]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Apr 23 20:06:05] -- Executing [h@default:1] AGI("Local/55558600052@default-0000000b;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Apr 23 20:06:06] -- <Local/55558600052@default-0000000b;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Apr 23 20:06:06] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:06:07] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:06:08] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 23 20:06:08] -- Called 2000 [Apr 23 20:06:08] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Apr 23 20:06:08] Timestamp: 00012ms SCall: 05137 DCall: 00000 10.0.0.58:4569 [Apr 23 20:06:08] VERSION : 2 [Apr 23 20:06:08] CALLED NUMBER : s [Apr 23 20:06:08] CODEC_PREFS : (ulaw|gsm) [Apr 23 20:06:08] CALLING NUMBER : 7542438008 [Apr 23 20:06:08] CALLING PRESNTN : 0 [Apr 23 20:06:08] CALLING TYPEOFN : 0 [Apr 23 20:06:08] CALLING TRANSIT : 0 [Apr 23 20:06:08] CALLING NAME : S2004232006078600052 [Apr 23 20:06:08] LANGUAGE : en [Apr 23 20:06:08] USERNAME : 2000 [Apr 23 20:06:08] FORMAT : 4 [Apr 23 20:06:08] FORMAT2 : ulaw [Apr 23 20:06:08] CAPABILITY : 6 [Apr 23 20:06:08] CAPABILITY2 : Unknown [Apr 23 20:06:08] ADSICPE : 2 [Apr 23 20:06:08] DATE TIME : 2020-04-23 20:06:08 [Apr 23 20:06:08] [Apr 23 20:06:08] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK [Apr 23 20:06:08] Timestamp: 00012ms SCall: 00182 DCall: 05137 10.0.0.58:4569 [Apr 23 20:06:08] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT [Apr 23 20:06:08] Timestamp: 00006ms SCall: 00182 DCall: 05137 10.0.0.58:4569 [Apr 23 20:06:08] FORMAT : 4 [Apr 23 20:06:08] [Apr 23 20:06:08] -- Call accepted by 10.0.0.58:4569 (format ulaw) [Apr 23 20:06:08] -- Format for call is (ulaw) [Apr 23 20:06:08] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK [Apr 23 20:06:08] Timestamp: 00006ms SCall: 05137 DCall: 00182 10.0.0.58:4569 [Apr 23 20:06:08] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING [Apr 23 20:06:08] Timestamp: 00003ms SCall: 00182 DCall: 05137 10.0.0.58:4569 [Apr 23 20:06:08] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK [Apr 23 20:06:08] Timestamp: 00003ms SCall: 05137 DCall: 00182 10.0.0.58:4569 [Apr 23 20:06:08] -- IAX2/2000-5137 is ringing imestamp: 03400ms SCall: 05137 DCall: 00182 10.0.0.58:4569 [Apr 23 20:06:11] -- IAX2/2000-5137 answered [Apr 23 20:06:11] -- Executing [8600052@default:1] MeetMe("IAX2/2000-5137", "8600052,F") in new stack [Apr 23 20:06:11] -- Created MeetMe conference 1023 for conference '8600052' [Apr 23 20:06:11] -- <IAX2/2000-5137> Playing 'conf-onlyperson.gsm' (language 'en') [Apr 23 20:06:11] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 004 Type: VOICE Subclass: 4 [Apr 23 20:06:11] Timestamp: 03440ms SCall: 05137 DCall: 00182 10.0.0.58:4569 [Apr 23 20:06:11] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 002 Type: IAX Subclass: ACK [Apr 23 20:06:11] Timestamp: 03440ms SCall: 00182 DCall: 05137 10.0.0.58:4569 [Apr 23 20:06:12] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:06:13] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ [Apr 23 20:06:13] Timestamp: 00003ms SCall: 00183 DCall: 00000 10.0.0.58:4569 [Apr 23 20:06:13] USERNAME : 2000 [Apr 23 20:06:13] REFRESH : 60 [Apr 23 20:06:13] [Apr 23 20:06:13] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK [Apr 23 20:06:13] Timestamp: 00003ms SCall: 13445 DCall: 00183 10.0.0.58:4569 [Apr 23 20:06:13] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH [Apr 23 20:06:13] Timestamp: 00009ms SCall: 13445 DCall: 00183 10.0.0.58:4569 [Apr 23 20:06:13] AUTHMETHODS : 2 [Apr 23 20:06:13] CHALLENGE : \x31\x34\x34\x35\x38\x36\x36\x34\x37 [Apr 23 20:06:13] USERNAME : 2000 [Apr 23 20:06:13] [Apr 23 20:06:13] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ [Apr 23 20:06:13] Timestamp: 00023ms SCall: 00183 DCall: 13445 10.0.0.58:4569 [Apr 23 20:06:13] USERNAME : 2000 [Apr 23 20:06:13] MD5 RESULT : 71a0d3395053f26ee981dfefc9d29b59 [Apr 23 20:06:13] [Apr 23 20:06:13] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK [Apr 23 20:06:13] Timestamp: 00023ms SCall: 13445 DCall: 00183 10.0.0.58:4569 [Apr 23 20:06:13] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK [Apr 23 20:06:13] Timestamp: 00040ms SCall: 13445 DCall: 00183 10.0.0.58:4569 [Apr 23 20:06:13] USERNAME : 2000 [Apr 23 20:06:13] DATE TIME : 2020-04-23 20:06:12 [Apr 23 20:06:13] REFRESH : 60 [Apr 23 20:06:13] APPARENT ADDRES : IPV4 10.0.0.58:4569 [Apr 23 20:06:13] MESSAGE COUNT : 0 [Apr 23 20:06:13] CALLING NUMBER : 7542XX8008 [Apr 23 20:06:13] [Apr 23 20:06:13] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK [Apr 23 20:06:13] Timestamp: 00040ms SCall: 00183 DCall: 13445 10.0.0.58:4569 [Apr 23 20:06:14] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE [Apr 23 20:06:14] Timestamp: 00005ms SCall: 04243 DCall: 00000 10.0.0.58:4569 [Apr 23 20:06:14] [Apr 23 20:06:14] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK [Apr 23 20:06:14] Timestamp: 00005ms SCall: 00184 DCall: 04243 10.0.0.58:4569 [Apr 23 20:06:14] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG [Apr 23 20:06:14] Timestamp: 00005ms SCall: 00184 DCall: 04243 10.0.0.58:4569 [Apr 23 20:06:14] RR_JITTER : 0 -- Called 8600052@default [Apr 23 20:06:43] -- Executing [8600052@default:1] MeetMe("Local/8600052@default-0000000c;2", "8600052,F") in new stack [Apr 23 20:06:43] -- Local/8600052@default-0000000c;1 answered [Apr 23 20:06:43] -- Executing [1001@default:1] Dial("Local/8600052@default-0000000c;1", "SIP/1001,60,") in new stack [Apr 23 20:06:43] ERROR[30227][C-0000001a]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("vicibox9", "(null)", ...): Name or service not known [Apr 23 20:06:43] WARNING[30227][C-0000001a]: acl.c:835 resolve_first: Unable to lookup 'vicibox9' [Apr 23 20:06:43] == Using SIP RTP CoS mark 5 [Apr 23 20:06:43] Audio is at 11174 [Apr 23 20:06:43] Adding codec ulaw to SDP [Apr 23 20:06:43] Adding codec gsm to SDP [Apr 23 20:06:43] Adding non-codec 0x1 (telephone-event) to SDP [Apr 23 20:06:43] Reliably Transmitting (NAT) to 10.0.0.55:5061: [Apr 23 20:06:43] INVITE sip:100XX:5061;rinstance=1af97b6b379109de;transport=UDP SIP/2.0 [Apr 23 20:06:43] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK6af7f61b;rport [Apr 23 20:06:43] Max-Forwards: 70 [Apr 23 20:06:43] From: "DV686803W2000200020W" <sip:7542XX8008@10.0.0.62>;tag=as350363de [Apr 23 20:06:43] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:06:43] Contact: <sip:7542XX8008@10.0.0.62:5060> [Apr 23 20:06:43] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:06:43] CSeq: 102 INVITE [Apr 23 20:06:43] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 20:06:43] Date: Fri, 24 Apr 2020 00:06:43 GMT [Apr 23 20:06:43] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 20:06:43] Supported: replaces, timer [Apr 23 20:06:43] Remote-Party-ID: "DV686803W2000200020W" <sip:7542438008@10.0.0.62>;party=calling;privacy=off;screen=no [Apr 23 20:06:43] Content-Type: application/sdp [Apr 23 20:06:43] Content-Length: 272 [Apr 23 20:06:43] [Apr 23 20:06:43] v=0 [Apr 23 20:06:43] o=root 116776649 116776649 IN IP4 10.0.0.62 [Apr 23 20:06:43] s=Asterisk PBX 13.29.2-vici [Apr 23 20:06:43] c=IN IP4 10.0.0.62 [Apr 23 20:06:43] t=0 0 [Apr 23 20:06:43] m=audio 11174 RTP/AVP 0 3 101 [Apr 23 20:06:43] a=rtpmap:0 PCMU/8000 [Apr 23 20:06:43] a=rtpmap:3 GSM/8000 [Apr 23 20:06:43] a=rtpmap:101 telephone-event/8000 [Apr 23 20:06:43] a=fmtp:101 0-16 [Apr 23 20:06:43] a=ptime:20 [Apr 23 20:06:43] a=maxptime:150 [Apr 23 20:06:43] a=sendrecv [Apr 23 20:06:43] [Apr 23 20:06:43] --- [Apr 23 20:06:43] -- Called SIP/1001 [Apr 23 20:06:43] [Apr 23 20:06:43] <--- SIP read from UDP:10.0.0.55:5061 ---> [Apr 23 20:06:43] SIP/2.0 100 Trying [Apr 23 20:06:43] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK6af7f61b;rport=5060 [Apr 23 20:06:43] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:06:43] From: "DV686803W2000200020W" <sip:7542438008@10.0.0.62>;tag=as350363de [Apr 23 20:06:43] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:06:43] CSeq: 102 INVITE [Apr 23 20:06:43] Content-Length: 0 [Apr 23 20:06:43] [Apr 23 20:06:43] <-------------> [Apr 23 20:06:43] --- (7 headers 0 lines) --- [Apr 23 20:06:44] [Apr 23 20:06:44] <--- SIP read from UDP:10.0.0.55:5061 ---> [Apr 23 20:06:44] SIP/2.0 180 Ringing [Apr 23 20:06:44] Via: SIP/2.0/UDP :5061;branch=z9hG4bK6af7f61b;rport=5060 [Apr 23 20:06:44] Contact: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:06:44] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=1a30080d [Apr 23 20:06:44] From: "DV686803W2000200020W"<sip:7542XX8008@10.0.0.62>;tag=as350363de [Apr 23 20:06:44] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:06:44] CSeq: 102 INVITE [Apr 23 20:06:44] User-Agent: Zoiper rev.6751 [Apr 23 20:06:44] Content-Length: 0 [Apr 23 20:06:44] [Apr 23 20:06:44] <-------------> [Apr 23 20:06:44] --- (9 headers 0 lines) --- [Apr 23 20:06:44] sip_route_dump: route/path hop: <sip:1001@73.46.29.100:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:06:44] -- SIP/1001-00000007 is ringing [Apr 23 20:06:44] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:06:47] Reliably Transmitting (NAT) to 10.0.0.58:5060: [Apr 23 20:06:47] OPTIONS sip:1002@10.0.0.58:5060;rinstance=9b67a99268885b8f;transport=UDP SIP/2.0 [Apr 23 20:06:47] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK1a0dd681;rport [Apr 23 20:06:47] Max-Forwards: 70 [Apr 23 20:06:47] From: "asterisk" <sip:asterisk@10.0.0.62>;tag=as24473625 [Apr 23 20:06:47] To: <sip:1002@10.0.0.58:5060;rinstance=9b67a99268885b8f;transport=UDP> [Apr 23 20:06:47] Contact: <sip:asterisk@10.0.0.62:5060> [Apr 23 20:06:47] Call-ID: 318bd8b41e590d8e5be24ebd7360d6f8@10.0.0.62:5060 [Apr 23 20:06:47] CSeq: 102 OPTIONS [Apr 23 20:06:47] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 20:06:47] Date: Fri, 24 Apr 2020 00:06:47 GMT [Apr 23 20:06:47] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Apr 23 20:06:47] Supported: replaces, timer [Apr 23 20:06:47] Content-Length: 0 [Apr 23 20:06:47] [Apr 23 20:06:47] [Apr 23 20:06:47] --- [Apr 23 20:06:47] [Apr 23 20:06:47] <--- SIP read from UDP:10.0.0.58:5060 ---> [Apr 23 20:06:47] SIP/2.0 200 OK [Apr 23 20:06:47] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK1a0dd681;rport=5060 [Apr 23 20:06:47] Contact: <sip:10.0.0.58:5060> [Apr 23 20:06:47] To: <sip:1002@10.0.0.58:5060;rinstance=9b67a99268885b8f;transport=UDP>;tag=d627436e [Apr 23 20:06:47] From: "asterisk"<sip:asterisk@10.0.0.62>;tag=as24473625 [Apr 23 20:06:47] Call-ID: 318bd8b41e590d8e5be24ebd7360d6f8@10.0.0.62:5060 [Apr 23 20:06:47] CSeq: 102 OPTIONS [Apr 23 20:06:47] Accept: application/sdp, application/sdp [Apr 23 20:06:47] Accept-Language: en [Apr 23 20:06:47] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE [Apr 23 20:06:47] User-Agent: Zoiper rev.6751 [Apr 23 20:06:47] Allow-Events: presence [Apr 23 20:06:47] Content-Length: 0 [Apr 23 20:06:47] [Apr 23 20:06:58] v=0 [Apr 23 20:06:58] o=Zoiper_user 0 3 IN IP4 [Apr 23 20:06:58] s=Zoiper_session [Apr 23 20:06:58] c=IN IP4 73.46.29.100 [Apr 23 20:06:58] t=0 0 [Apr 23 20:06:58] m=audio 8002 RTP/AVP 0 3 8 110 98 101 [Apr 23 20:06:58] a=rtpmap:0 PCMU/8000 [Apr 23 20:06:58] a=rtpmap:3 GSM/8000 [Apr 23 20:06:58] a=rtpmap:8 PCMA/8000 [Apr 23 20:06:58] a=rtpmap:110 speex/8000 [Apr 23 20:06:58] a=rtpmap:98 iLBC/8000 [Apr 23 20:06:58] a=fmtp:98 mode=30 [Apr 23 20:06:58] a=rtpmap:101 telephone-event/8000 [Apr 23 20:06:58] a=fmtp:101 0-15 [Apr 23 20:06:58] a=sendrecv [Apr 23 20:06:58] <-------------> [Apr 23 20:06:58] --- (11 headers 15 lines) --- [Apr 23 20:06:58] Found RTP audio format 0 [Apr 23 20:06:58] Found RTP audio format 3 [Apr 23 20:06:58] Found RTP audio format 8 [Apr 23 20:06:58] Found RTP audio format 110 [Apr 23 20:06:58] Found RTP audio format 98 [Apr 23 20:06:58] Found RTP audio format 101 [Apr 23 20:06:58] Found audio description format PCMU for ID 0 [Apr 23 20:06:58] Found audio description format GSM for ID 3 [Apr 23 20:06:58] Found audio description format PCMA for ID 8 [Apr 23 20:06:58] Found audio description format speex for ID 110 [Apr 23 20:06:58] Found audio description format iLBC for ID 98 [Apr 23 20:06:58] Found audio description format telephone-event for ID 101 [Apr 23 20:06:58] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|gsm) [Apr 23 20:06:58] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Apr 23 20:06:58] > 0x7fa710070870 -- Strict RTP learning after remote address set to: :8002 [Apr 23 20:06:58] Peer audio RTP is at port :8002 [Apr 23 20:06:58] sip_route_dump: route/path hop: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:06:58] Transmitting (NAT) to 10.0.0.55:5061: [Apr 23 20:06:58] ACK sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP SIP/2.0 [Apr 23 20:06:58] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK71b0bd0e;rport [Apr 23 20:06:58] Max-Forwards: 70 [Apr 23 20:06:58] From: "DV686803W2000200020W" <sip:7542438008@10.0.0.62>;tag=as350363de [Apr 23 20:06:58] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=1a30080d [Apr 23 20:06:58] Contact: <sip:7542XX8008@10.0.0.62:5060> [Apr 23 20:06:58] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:06:58] CSeq: 102 ACK [Apr 23 20:06:58] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 20:06:58] Content-Length: 0 [Apr 23 20:06:58] [Apr 23 20:06:58] [Apr 23 20:06:58] --- [Apr 23 20:06:58] -- SIP/1001-00000007 answered Local/8600052@default-0000000c;1 [Apr 23 20:06:58] -- Channel SIP/1001-00000007 joined 'simple_bridge' basic-bridge <b65c35c2-b309-4e6e-b577-d7a74fa1977c> [Apr 23 20:06:58] -- Channel Local/8600052@default-0000000c;1 joined 'simple_bridge' basic-bridge <b65c35c2-b309-4e6e-b577-d7a74fa1977c> [Apr 23 20:06:58] > 0x7fa710070870 -- Strict RTP qualifying stream type: audio [Apr 23 20:06:58] > 0x7fa710070870 -- Strict RTP switching source address to 10.0.0.55:8002 [Apr 23 20:07:02] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 23 20:07:03] > 0x7fa710070870 -- Strict RTP learning complete - Locking on source address 10.0.0.55:8002 [Apr 23 20:07:03] == Manager 'sendcron' logged on from 127.0.0.1 -- Manager 'sendcron' from 127.0.0.1, hanging up channel: IAX2/2000-5137 [Apr 23 20:07:14] == Spawn extension (default, 8600052, 1) exited non-zero on 'IAX2/2000-5137' [Apr 23 20:07:14] WARNING[28158][C-00000018]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Apr 23 20:07:14] -- Executing [h@default:1] AGI("IAX2/2000-5137", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack [Apr 23 20:07:14] -- <IAX2/2000-5137>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0 [Apr 23 20:07:14] -- Hungup 'IAX2/2000-5137' [Apr 23 20:07:14] Tx-Frame Retry[000] -- OSeqno: 012 ISeqno: 014 Type: IAX Subclass: HANGUP [Apr 23 20:07:14] Timestamp: 66789ms SCall: 05137 DCall: 00182 10.0.0.58:4569 [Apr 23 20:07:14] CAUSE CODE : 0 [Apr 23 20:07:14] [Apr 23 20:07:14] == Manager 'sendcron' logged on from 127.0.0.1 [Apr 23 20:07:14] Rx-Frame Retry[ No] -- OSeqno: 014 ISeqno: 013 Type: IAX Subclass: ACK [Apr 23 20:07:14] Timestamp: 66789ms SCall: 00182 DCall: 05137 10.0.0.58:4569 [Apr 23 20:07:14] -- Called 55558600052@default [Apr 23 20:07:14] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-0000000d;2", "8600052,K") in new stack [Apr 23 20:07:14] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-0000000d;2", "") in new stack [Apr 23 20:07:14] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-0000000d;2' [Apr 23 20:07:14] WARNING[31289][C-0000001b]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Apr 23 20:07:14] -- Executing [h@default:1] AGI("Local/55558600052@default-0000000d;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Apr 23 20:07:14] -- <Local/55558600052@default-0000000d;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Apr 23 20:07:15] -- <Local/8600052@default-0000000c;2> Playing 'conf-kicked.gsm' (language 'en') [Apr 23 20:07:15] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:07:15] == Manager 'sendcron' logged off from 127.0.0.1 [Apr 23 20:07:17] -- Hungup 'DAHDI/pseudo-2028470289' [Apr 23 20:07:17] -- Executing [8600052@default:2] Hangup("Local/8600052@default-0000000c;2", "") in new stack [Apr 23 20:07:17] == Spawn extension (default, 8600052, 2) exited non-zero on 'Local/8600052@default-0000000c;2' [Apr 23 20:07:17] WARNING[30229][C-00000019]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Apr 23 20:07:17] -- Executing [h@default:1] AGI("Local/8600052@default-0000000c;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Apr 23 20:07:17] -- <Local/8600052@default-0000000c;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Apr 23 20:07:17] -- Channel Local/8600052@default-0000000c;1 left 'simple_bridge' basic-bridge <b65c35c2-b309-4e6e-b577-d7a74fa1977c> [Apr 23 20:07:17] == Spawn extension (default, 1001, 1) exited non-zero on 'Local/8600052@default-0000000c;1' [Apr 23 20:07:17] -- Executing [h@default:1] AGI("Local/8600052@default-0000000c;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----33-----33-----SIP 200 OK)") in new stack [Apr 23 20:07:17] -- Channel SIP/1001-00000007 left 'simple_bridge' basic-bridge <b65c35c2-b309-4e6e-b577-d7a74fa1977c> [Apr 23 20:07:17] Scheduling destruction of SIP dialog '4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060' in 6400 ms (Method: INVITE) [Apr 23 20:07:17] Reliably Transmitting (NAT) to 10.0.0.55:5061: [Apr 23 20:07:17] BYE sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP SIP/2.0 [Apr 23 20:07:17] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK32331e71;rport [Apr 23 20:07:17] Max-Forwards: 70 [Apr 23 20:07:17] From: "DV686803W2000200020W" <sip:7542438008@10.0.0.62>;tag=as350363de [Apr 23 20:07:17] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=1a30080d [Apr 23 20:07:17] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:07:17] CSeq: 103 BYE [Apr 23 20:07:17] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 20:07:17] X-Asterisk-HangupCause: Normal Clearing [Apr 23 20:07:17] X-Asterisk-HangupCauseCode: 16 [Apr 23 20:07:17] Content-Length: 0 [Apr 23 20:07:17] [Apr 23 20:07:17] [Apr 23 20:07:17] --- [Apr 23 20:07:17] -- <Local/8600052@default-0000000c;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----33-----33-----SIP 200 OK) completed, returning 0 [Apr 23 20:07:17] Retransmitting #1 (NAT) to 10.0.0.55:5061: [Apr 23 20:07:17] BYE sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP SIP/2.0 [Apr 23 20:07:17] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK32331e71;rport [Apr 23 20:07:17] Max-Forwards: 70 [Apr 23 20:07:17] From: "DV686803W2000200020W" <sip:7542438008@10.0.0.62>;tag=as350363de [Apr 23 20:07:17] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=1a30080d [Apr 23 20:07:17] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:07:17] CSeq: 103 BYE [Apr 23 20:07:17] User-Agent: Asterisk PBX 13.29.2-vici [Apr 23 20:07:17] X-Asterisk-HangupCause: Normal Clearing [Apr 23 20:07:17] X-Asterisk-HangupCauseCode: 16 [Apr 23 20:07:17] Content-Length: 0 [Apr 23 20:07:17] [Apr 23 20:07:17] [Apr 23 20:07:17] --- [Apr 23 20:07:17] [Apr 23 20:07:17] <--- SIP read from UDP:10.0.0.55:5061 ---> [Apr 23 20:07:17] SIP/2.0 200 OK [Apr 23 20:07:17] Via: SIP/2.0/UDP :5061;branch=z9hG4bK32331e71;rport=5060 [Apr 23 20:07:17] Contact: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP> [Apr 23 20:07:17] To: <sip:1001@:5061;rinstance=1af97b6b379109de;transport=UDP>;tag=1a30080d [Apr 23 20:07:17] From: "DV686803W2000200020W"<sip:75424XX008@10.0.0.62>;tag=as350363de [Apr 23 20:07:17] Call-ID: 4017aa212444e598103cb8a068bdc3d0@10.0.0.62:5060 [Apr 23 20:07:17] CSeq: 103 BYE [Apr 23 20:07:17] User-Agent: Zoiper rev.6751 [Apr 23 20:07:17] Content-Length: 0 [Apr 23 20:07:17] [Apr 23 20:07:17] <------------->

It looks like asterisk is trying to send my internal IP to my sip provider, is there any way to have vicidial or asterisk replace "10.0.0.62" with my public ip before its sent out to my sip carrier?

<--- SIP read from UDP:192.76.120.10:5060 --->
[Apr 24 12:02:54] SIP/2.0 200 Keepalive
[Apr 24 12:02:54] Via: SIP/2.0/UDP 10.0.0.62:5060;branch=z9hG4bK0d311b23;rport=5060;received=publicIP
[Apr 24 12:02:54] From: "asterisk" <sip:asterisk@10.0.0.62>;tag=as5379b0e5
[Apr 24 12:02:54] To: <sip:sip.telnyx.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.dba8
[Apr 24 12:02:54] Call-ID: 63103e6f2016c603186b1868336d1163@10.0.0.62:5060 -
[Apr 24 12:02:54] CSeq: 102 OPTIONS
[Apr 24 12:02:54] Server: kamailio (5.0.8 (x86_64/linux))
[Apr 24 12:02:54] Content-Length: 0

Notice the "Call-ID"

I cant connect from vicidial to make calls to the outside world....my public IP is in the DMZ right now. Thanks for any suggestions

Posted: **Fri Apr 24, 2020 11:14 am**

Page, line and version of your Vicidial Manager's Manual? Seems to be not included in your question.

Are you in the Carrier set up. In your carrier's "Dialplan Entry" field should be the dialplan for this carrier.

The dialplan consists of:

Carrier Prefix (aka: "dial prefix" in the Campaign Modify). This value is used SOLELY to select a carrier, then it is discarded.

Country Code. If dialing internationally, this is present. Obvioiusly it's just missing entirely

Dial Code (aka: "1" in the US or "0" in the UK). This is the "Domestic long distance" digit in most countries. If missing, the number is a local call (ie: are you calling someone on the other side of the street or the other side of the country?). In almost all cases in the US, this is a "1" because all VOIP calls are at least domestic LD. No actual "local" calls.

Phone Number. I think this is obvious.

Here's what the final looks like:

Carrier Prefix - Country Code - Dial Code - Phone Number

Example

9 - 44 - 0 - 2037692294

Next: This number is processed in whatever fashion required to match what the carrier actually needs. For instance, some carriers require a security code or customer code in front of the number being dialed. There are no rules at the carrier level. No standard. So they literally make this up as they go. For instance, it may be necessary to dial "51500" before every number. But remember that the "9" was being used merely to select this carrier, which allows multiple carriers to be installed and usable at the same time on a single dialer. So the "9" would be discarded and the "51500" would be inserted.

So here's an example of how all that would be done:

Code: Select all: exten=>_91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten=>_91NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/51500${EXTEN:1},,To) exten=>_91NXXNXXXXXX,n,Hangup exten=>_9440XXXXXXXXXX ,1,AGI(agi://127.0.0.1:4577/call_log) exten=>_9440XXXXXXXXXX ,n,Dial(${DIAL9TRUNK}/51500${EXTEN:1},,To) exten=>_9440XXXXXXXXXX ,n,Hangup

Breaking it down:

_91NXXNXXXXXX = "_" is the symbol stating that this is a "pattern" not a literal number. Followed by our carrier prefix (9) and the dial code (1) and then "NXXNXXXXXX" which is the US/NANPA dial pattern. The 1st and 4th digits can not be "0" or "1" (N) the rest of the digits can be any digit (X). That covers the US dialing.
_9440XXXXXXXXXX = "_" is the symbol stating that this is a "pattern" not a literal number. Followed by our carrier prefix (9) and the country code (44) and the dial code (0) and then "XXXXXXXXXX" which is the UK 10 digit dial pattern.

Note that _91NXXNXXXXXX and _9440XXXXXXXXXX are UNrelated. They are each their own program, they do not interact. The fact that they are in the same box in the carrier settings does not make them related. They could also be each in their own carrier box or even in an unrelated carrier since they all land in the same place in the extensions-vicidial.conf file.

Note also that @${DIAL9TRUNK}@ requires an entry in "Globals string" of DIAL9TRUNK=SIP/carriercontext where carriercontext is the portion in the account entry in the brackets.

Posted: **Fri Apr 24, 2020 12:34 pm**

I have installed from loading the vicibox server .iso locally and following the instruction manual.

https://supportalg.blogspot.com/2016/01/wiki-vicidial-add-new-carrier-trunk-to.html - managers manual page

i have input my carrier settings as per my sip provider. when i troubleshoot with them, they tell me the packets are not being received on their end, because its being sent out from my end with my Local IP and not my public one.

My dcarrier settings directly from them when it comes to using vicidial:

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

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

I am in south florida.

The main problem I am having is mainly this portion: NOTICE[4297][C-0000000f]: chan_sip.c:26515 handle_request_invite: Call from '1002' (10.0.0.58:5060) to extension '9549477572' rejected because extension not found in context 'default'.

Where/what is [default] ?

And then what my SIP provider told me last night is that this portion shouldnt be sending my internal IP, it should send my public one:
Is there any way to set this in vicidial or asterisk to always use a set ip on outgoing transmissions to carrier?

Call-ID: 541e8baf5592ec8b7b94fab30c4c1270@10.0.0.62:5060

Thank you

Posted: **Fri Apr 24, 2020 1:25 pm**

you dialed

9549XX7572

your dial pattern is

_9NXXXXXXXXXX

those do NOT match. Thus your system responded "Call from '1002' (10.0.0.58:5060) to extension '9549477572' rejected because extension not found in context 'default'"

You could try _NXXXXXXXXXX instead of _9NXXXXXXXXXX which has the proper number of digits.

OR you could dial 99549XX7572 which includes the Carrier Dial Prefix to choose that carrier and would allow you to also have 89549XX7572 and 79549XX7572 carriers active in the system and make choosing a carrier without having to activate/deactivate anything possible. It also makes it possible for different campaigns to use different carriers and for autodial and manual dial to use different carriers.

but in all cases, as the error message states, the dialed number must match a dial pattern in the dialplan. In this case, Vicidial uses [default] and it will look in that context for a dial pattern match.

Posted: **Sat Apr 25, 2020 7:10 pm**

It was so much less complicated than I thought and I am embarrased to admit it but hopefully it helps the next person. The carrier I chose that had a free trial, provided only the SIP trunk, not the channels required in order to make calls. No channels = no calls!! I thought it came with them for testing purposes but it did not, you have to purchase the amount of channels you want to use.

From my research, there is no company out there that will provide you with even one channel for testing purposes without paying first.

Posted: **Tue Apr 28, 2020 2:30 pm**

I can give you one with one dollar on the account for testing. We generally only do this for customers, however, so this would have to be handled off the forum. I'm not allowed to divulge carrier lists to non-clients and certainly not on a public forum!

The $1 must be used within one week of activation, and they generally take a day or two for activation.

That being said, many providers will allow you to open an account with very little money on it. And it's not like you aren't going to use it ...

Also, we only charge $25 to install a carrier in your server. If we were to install that carrier that allows for the $1 free credit test ... you wouldn't even have to be the one to set it up and verify that it works. Some things can wait until later, after you are making money with your dialer. Such as how to set up carriers. This is the one thing that is simply not universal enough to be in the Vicidial Manager's Manual as a step-by-step procedure because every carrier interface is different in some odd fashion.

Posted: **Tue Apr 28, 2020 3:09 pm**

That would be awesome, how would you like me to get in contact?

Posted: **Tue Apr 28, 2020 4:27 pm**

Best method is to email Support@PoundTeam.com and requet a Test SIP Account.

We'll need all your contact information in the email (name/phone/address, etc) to give to the carrier.

vicidial.org

Call from '1002' to extension '9549XX7572' - Rejected

Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected

Re: Call from '1002' to extension '9549XX7572' - Rejected