[asterisk-bugs] [Asterisk 0015367]: Call failed to go through, [...] instead of excuting next extension
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jun 24 13:37:03 CDT 2009
The following issue has been CLOSED
======================================================================
https://issues.asterisk.org/view.php?id=15367
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Reported By: kowalma
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 15367
Category: Applications/app_dial
Reproducibility: always
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.1.1
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
======================================================================
Date Submitted: 2009-06-20 15:15 CDT
Last Modified: 2009-06-24 13:37 CDT
======================================================================
Summary: Call failed to go through, [...] instead of excuting
next extension
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 at gsm_out:16]
Dial("Local/0110000100000619501522511 at 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 at ccig-223e;2'
-- Executing [h at gsm_out:1]
Hangup("Local/0110000100000619501522511 at ccig-223e;2", "") in new stack
== Spawn extension (gsm_out, h, 1) exited non-zero on
'Local/0110000100000619501522511 at 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 at 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 at 192.168.9.4>;tag=as6e80ded9..To:
<sip:12501522511 at 192.168.0.20>..Contact: <sip:aster
isk at 192.168.9.4>..Call-ID:
7f6d0da417fea26b4a76de2977932fd6 at 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 at 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 at 192.168.9.4>;tag=as6e80ded9..To:
<sip:12501522511 at 192.168.0.20>..Contact: <sip:aster
isk at 192.168.9.4>..Call-ID:
7f6d0da417fea26b4a76de2977932fd6 at 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 at 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 at 192.168.9.4>;tag=as6e80ded9..To:
<sip:12501522511 at 192.168.0.20>..Contact: <sip:aster
isk at 192.168.9.4>..Call-ID:
7f6d0da417fea26b4a76de2977932fd6 at 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 at 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 at 192.168.9.4>;tag=as6e80ded9..To:
<sip:12501522511 at 192.168.0.20>..Contact: <sip:aster
isk at 192.168.9.4>..Call-ID:
7f6d0da417fea26b4a76de2977932fd6 at 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 at 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 at 192.168.9.4>;tag=as6e80ded9..To:
<sip:12501522511 at 192.168.0.20>..Contact: <sip:aster
isk at 192.168.9.4>..Call-ID:
7f6d0da417fea26b4a76de2977932fd6 at 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
======================================================================
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(0106922) lmadsen (administrator) - 2009-06-24 13:37
https://issues.asterisk.org/view.php?id=15367#c106922
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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.
Issue History
Date Modified Username Field Change
======================================================================
2009-06-24 13:37 lmadsen Note Added: 0106922
2009-06-24 13:37 lmadsen Status new => closed
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