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Summary:ASTERISK-14348: Call failed to go through, [...] instead of excuting next extension
Reporter:Marcin Kowalczyk (kowalma)Labels:
Date Opened:2009-06-20 15:15:02Date Closed:2011-06-07 14:00:32
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I'm testing new version of my dialplan and I think I've found a bug.

I'm trying to dial GSM gateway and when it's unrechable (ie connection down) dialplan execution ends insead going into next extension:



   -- Executing [12501522511@gsm_out:16] Dial("Local/0110000100000619501522511@ccig-223e;2", "sip/192.168.0.20/12501522511,60,g") in new stack
 == Using SIP RTP CoS mark 5
   -- Called 192.168.0.20/12501522511
[Jun 20 22:02:23] NOTICE[14643]: pbx_spool.c:338 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[Jun 20 22:02:23] NOTICE[14643]: pbx_spool.c:338 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
 == Spawn extension (gsm_out, 12501522511, 16) exited non-zero on 'Local/0110000100000619501522511@ccig-223e;2'
   -- Executing [h@gsm_out:1] Hangup("Local/0110000100000619501522511@ccig-223e;2", "") in new stack
 == Spawn extension (gsm_out, h, 1) exited non-zero on 'Local/0110000100000619501522511@ccig-223e;2'
[Jun 20 22:02:23] ERROR[13916]: pbx.c:8618 device_state_cb: Received invalid event that had no device IE
[Jun 20 22:02:23] ERROR[13916]: pbx.c:8618 device_state_cb: Received invalid event that had no device IE
[Jun 20 22:02:23] ERROR[13916]: app_queue.c:810 device_state_cb: Received invalid event that had no device IE
[Jun 20 22:02:23] ERROR[13916]: app_queue.c:810 device_state_cb: Received invalid event that had no device IE
Asterisk-node1*CLI>


I did ngrep to see packets send to SIP-gateway:

#
U 192.168.9.4:5060 -> 192.168.0.20:5060
 INVITE sip:12501522511@192.168.0.20 SIP/2.0..Via: SIP/2.0/UDP 192.168.9.4:5060;branch=z9hG4bK7040e375;rport..Max-Forwar
 ds: 70..From: "test" <sip:asterisk@192.168.9.4>;tag=as6e80ded9..To: <sip:12501522511@192.168.0.20>..Contact: <sip:aster
 isk@192.168.9.4>..Call-ID: 7f6d0da417fea26b4a76de2977932fd6@192.168.9.4..CSeq: 102 INVITE..User-Agent: Asterisk PBX 1.6
 .1.1..Date: Sat, 20 Jun 2009 20:10:16 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supporte
 d: replaces, timer..Content-Type: application/sdp..Content-Length: 306....v=0..o=root 592439500 592439500 IN IP4 192.16
 8.9.4..s=Asterisk PBX 1.6.1.1..c=IN IP4 192.168.9.4..t=0 0..m=audio 19010 RTP/AVP 0 8 3 101..a=rtpmap:0 PCMU/8000..a=rt
 pmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a
 =ptime:20..a=sendrecv..
#
U 192.168.9.4:5060 -> 192.168.0.20:5060
 INVITE sip:12501522511@192.168.0.20 SIP/2.0..Via: SIP/2.0/UDP 192.168.9.4:5060;branch=z9hG4bK7040e375;rport..Max-Forwar
 ds: 70..From: "test" <sip:asterisk@192.168.9.4>;tag=as6e80ded9..To: <sip:12501522511@192.168.0.20>..Contact: <sip:aster
 isk@192.168.9.4>..Call-ID: 7f6d0da417fea26b4a76de2977932fd6@192.168.9.4..CSeq: 102 INVITE..User-Agent: Asterisk PBX 1.6
 .1.1..Date: Sat, 20 Jun 2009 20:10:16 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supporte
 d: replaces, timer..Content-Type: application/sdp..Content-Length: 306....v=0..o=root 592439500 592439500 IN IP4 192.16
 8.9.4..s=Asterisk PBX 1.6.1.1..c=IN IP4 192.168.9.4..t=0 0..m=audio 19010 RTP/AVP 0 8 3 101..a=rtpmap:0 PCMU/8000..a=rt
 pmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a
 =ptime:20..a=sendrecv..
#
U 192.168.9.4:5060 -> 192.168.0.20:5060
 INVITE sip:12501522511@192.168.0.20 SIP/2.0..Via: SIP/2.0/UDP 192.168.9.4:5060;branch=z9hG4bK7040e375;rport..Max-Forwar
 ds: 70..From: "test" <sip:asterisk@192.168.9.4>;tag=as6e80ded9..To: <sip:12501522511@192.168.0.20>..Contact: <sip:aster
 isk@192.168.9.4>..Call-ID: 7f6d0da417fea26b4a76de2977932fd6@192.168.9.4..CSeq: 102 INVITE..User-Agent: Asterisk PBX 1.6
 .1.1..Date: Sat, 20 Jun 2009 20:10:16 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supporte
 d: replaces, timer..Content-Type: application/sdp..Content-Length: 306....v=0..o=root 592439500 592439500 IN IP4 192.16
 8.9.4..s=Asterisk PBX 1.6.1.1..c=IN IP4 192.168.9.4..t=0 0..m=audio 19010 RTP/AVP 0 8 3 101..a=rtpmap:0 PCMU/8000..a=rt
 pmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a
 =ptime:20..a=sendrecv..
#
U 192.168.9.4:5060 -> 192.168.0.20:5060
 INVITE sip:12501522511@192.168.0.20 SIP/2.0..Via: SIP/2.0/UDP 192.168.9.4:5060;branch=z9hG4bK7040e375;rport..Max-Forwar
 ds: 70..From: "test" <sip:asterisk@192.168.9.4>;tag=as6e80ded9..To: <sip:12501522511@192.168.0.20>..Contact: <sip:aster
 isk@192.168.9.4>..Call-ID: 7f6d0da417fea26b4a76de2977932fd6@192.168.9.4..CSeq: 102 INVITE..User-Agent: Asterisk PBX 1.6
 .1.1..Date: Sat, 20 Jun 2009 20:10:16 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supporte
 d: replaces, timer..Content-Type: application/sdp..Content-Length: 306....v=0..o=root 592439500 592439500 IN IP4 192.16
 8.9.4..s=Asterisk PBX 1.6.1.1..c=IN IP4 192.168.9.4..t=0 0..m=audio 19010 RTP/AVP 0 8 3 101..a=rtpmap:0 PCMU/8000..a=rt
 pmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a
 =ptime:20..a=sendrecv..
#
U 192.168.9.4:5060 -> 192.168.0.20:5060
 INVITE sip:12501522511@192.168.0.20 SIP/2.0..Via: SIP/2.0/UDP 192.168.9.4:5060;branch=z9hG4bK7040e375;rport..Max-Forwar
 ds: 70..From: "test" <sip:asterisk@192.168.9.4>;tag=as6e80ded9..To: <sip:12501522511@192.168.0.20>..Contact: <sip:aster
 isk@192.168.9.4>..Call-ID: 7f6d0da417fea26b4a76de2977932fd6@192.168.9.4..CSeq: 102 INVITE..User-Agent: Asterisk PBX 1.6
 .1.1..Date: Sat, 20 Jun 2009 20:10:16 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supporte
 d: replaces, timer..Content-Type: application/sdp..Content-Length: 306....v=0..o=root 592439500 592439500 IN IP4 192.16
 8.9.4..s=Asterisk PBX 1.6.1.1..c=IN IP4 192.168.9.4..t=0 0..m=audio 19010 RTP/AVP 0 8 3 101..a=rtpmap:0 PCMU/8000..a=rt
 pmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a
 =ptime:20..a=sendrecv..


and when GSM gateway won't send back declinded or sth else asterisk drops connection but it should go to next priority ie to play prompt that connection cannot be established or to reroute call via different gateway
Comments:By: Leif Madsen (lmadsen) 2009-06-24 13:37:03

This looks like a configuration issue to me. The Local channel is the one that is being hung up, thus the 'g' option on the Dial() instead the Local channel has no effect.