Hi, when I make an outgoing call sends me a busy here, and no one is making call
Contact: <sip:[email protected]:5060> Content-Length: 0 <------------> -- Executing [984783842@to_pstn:1] Dial("SIP/101-0000004e", "SIP/5001/84783842@,40,rRT") in new stack == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 13780 Video is at 50.X.X.X:18488 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding video codec 200004 (h264) to SDP Adding video codec 200003 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 190.53.38.203:5060: INVITE sip:84783842%[email protected] SIP/2.0 Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport Max-Forwards: 70 From: "Operadora" <sip:[email protected]>;tag=as3708c762 To: <sip:84783842%[email protected]> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: inmaconsa-Voice-Sip-ipbx Date: Mon, 19 Jan 2015 20:17:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Operadora" <sip:[email protected]>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 507 v=0 o=root 541548714 541548714 IN IP4 50.X.X.X s=inamaconsa-Voice-Sip-pbx c=IN IP4 50.X.X.X b=CT:384 t=0 0 m=audio 13780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 18488 RTP/AVP 99 98 a=rtpmap:99 H264/90000 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/90000 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=sendrecv --- -- Called SIP/5001/84783842@ <--- Transmitting (NAT) to 190.X.X.1:41316 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 To: <sip:[email protected]>;tag=as77fb37e2 Call-ID: [email protected] CSeq: 102 INVITE Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:[email protected]:5060> Content-Length: 0 <------------> <--- SIP read from UDP:190.53.38.203:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;received=50.X.X.X;rport=5060 From: "Operadora" <sip:[email protected]>;tag=as3708c762 To: <sip:84783842%[email protected]>;tag=as4bb74f30 Call-ID: [email protected]:5060 CSeq: 102 INVITE Server: VoxStack Wireless Gateway Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Transmitting (NAT) to 190.53.38.203:5060: ACK sip:84783842%[email protected] SIP/2.0 Via: SIP/2.0/UDP 50.X.X.X:5060;branch=z9hG4bK374c2247;rport Max-Forwards: 70 From: "Operadora" <sip:[email protected]>;tag=as3708c762 To: <sip:84783842%[email protected]>;tag=as4bb74f30 Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 ACK User-Agent: inmaconsa-Voice-Sip-ipbx Content-Length: 0 --- [Jan 19 14:17:53] WARNING[11596][C-0000003d]: chan_sip.c:23037 handle_response_invite: Received response: "Forbidden" from '"Operadora" <sip:[email protected]>;tag=as3708c762' Scheduling destruction of SIP dialog '[email protected]:5060' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [984783842@to_pstn:2] Busy("SIP/101-0000004e", "3") in new stack <--- Reliably Transmitting (NAT) to 190.X.X.1:41316 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 To: <sip:[email protected]>;tag=as77fb37e2 Call-ID: [email protected] CSeq: 102 INVITE Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 <------------> == Spawn extension (to_pstn, 984783842, 2) exited non-zero on 'SIP/101-0000004e' <--- SIP read from UDP:190.X.X.1:41316 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-61b74f36 From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 To: <sip:[email protected]>;tag=as30070ac7 Call-ID: [email protected] CSeq: 101 ACK Max-Forwards: 70 Contact: "101" <sip:[email protected]:41316> User-Agent: Cisco/SPA508G-7.5.6 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Retransmitting #1 (NAT) to 190.X.X.1:41316: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 190.X.X.1:41316;branch=z9hG4bK-b7674e2;received=190.X.X.1;rport=41316 From: "101" <sip:[email protected]>;tag=35721c1e3f767ceao4 To: <sip:[email protected]>;tag=as77fb37e2 Call-ID: [email protected] CSeq: 102 INVITE Server: inmaconsa-Voice-Sip-ipbx Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 2015-01-19 10:24 GMT-06:00 ricky gutierrez <[email protected]>: > Hi list, I write on the list looking for help, buy a openvox gw gsm > for four channels and I'm a little disappointed with the support > openvox, for some reason , The call doesn´t get trough > > support tells me it was my asterisk server, but does not really work > me and my internal calls are working perfectly, I tested with another > sangoma FXO gateway and works perfectly. > > the problem is that support openvox is Chinese and the difference in > time zone is high. > > my trunk is connected > > 5001/5001 X.X.X.X D Yes > Yes 5060 > > Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline] > > I follow this guide , but not work > > http://www.lojamundi.com.br/download/gateways-gsm/openvox/Quickstart_Guide_of_OpenVox_GSM_Gateway_VS-GW2120_Series_Connect_with_Asterisk_Server.pdf > > -- > rickygm > > http://gnuforever.homelinux.com -- rickygm http://gnuforever.homelinux.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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