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Summary:ASTERISK-03079: sip unable to create/find channel
Reporter:heison (heison)Labels:
Date Opened:2004-12-22 23:02:08.000-0600Date Closed:2011-06-07 14:10:28
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) extensions.conf.txt
( 1) sip.conf.txt
( 2) sipdebug-120804.txt
( 3) sipdebug-1222.txt
Description:on the 7960, i see calling (out INV)... but looks like the invite is not going anywhere. coz CLI complains about sip unable to create/find channel:

Asterisk CVS-HEAD-12/22/04-23:52:15 built by root@pbx.ykz.zealnetworks.com on a i686 running Linux

Dec 22 23:59:54 NOTICE[1160]: chan_sip.c:7612 handle_request: Unable to create/find channel
Dec 23 00:00:00 WARNING[1160]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 003094c2-9c86006b-60f6bd1e-71e823ce@10.155.10.5 for seqno 101 (Non-critical Response)
Dec 23 00:00:05 NOTICE[1160]: chan_sip.c:7612 handle_request: Unable to create/find channel
Dec 23 00:00:07 WARNING[1160]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 003094c2-9c86006b-60f6bd1e-71e823ce@10.155.10.5 for seqno 101 (Non-critical Response)
Dec 23 00:00:15 WARNING[1160]: chan_sip.c:701 retrans_pkt: Maximum retries exceeded on call 003094c2-9c86006b-60f6bd1e-71e823ce@10.155.10.5 for seqno 101 (Non-critical Response)
Comments:By: Olle Johansson (oej) 2004-12-23 01:11:07.000-0600

"unable to create channel" is usually a configuration error. Do you really have a confirmed bug?

Also, according to the bug guidelines, when reporting SIP bugs you have to add output from "sip debug" in your asterisk server. In this case, also sip.conf and extensions.conf. There is no way I can see if there is a bug with the information provided.

If you have questions about this, please find a bug marshal in the #asterisk-bugs channel on IRC.

/Olle
By: Olle Johansson (oej) 2004-12-23 01:11:47.000-0600

...and - again - this is not qualified as a MAJOR bug...
By: Clod Patry (junky) 2004-12-23 05:23:38.000-0600

If you can provide us any debug file that would help a lot.
And also, if you can paste ur part of sip.conf and your Dial line in extensions.conf, that would be really cool.
By: Mark Spencer (markster) 2004-12-23 05:59:01.000-0600

Since you are placing a SIP bug, you should include a SIP debug message.

You already have -4 karma.  If you continue to place bugs without reading the bug guidelines, they will simply be deleted in the future.
By: heison (heison) 2004-12-23 06:51:27.000-0600

comparing the 2 CVS (CVS-HEAD-12/08/04-23:37:47 & CVS-HEAD-12/22/04-23:52:15), here is what I see:

with cisco1 calling extension 1520 being an internal call, the later CVS doesn't seem to believe it is a call with no NAT and tried to transmit with NAT. Comparing sip.conf.sample of the 2 CVS, I can't find behaviour change:

124d123
< ;registertimeout=20           ; retry registration calls every 20 seconds (default)
147a147
> ; auth                        auth
By: heison (heison) 2004-12-23 07:00:29.000-0600

The dial line in extension.conf:

KITCHEN=Zap/4
exten => 1520,1,Macro(stdexten,1599,${KITCHEN})
By: heison (heison) 2004-12-23 07:02:18.000-0600

and the macro:

[macro-stdexten]
exten => s,1,Dial(${ARG2},20,Tr)
exten => s,2,Voicemail(u${ARG1})
By: Olle Johansson (oej) 2004-12-23 07:20:05.000-0600

Neither debug file includes the text "Unable to create/find channel". And in your example you are dialing a ZAP channel.

You need to follow up with debug files that include the problem you report.
By: heison (heison) 2004-12-23 07:38:16.000-0600

[cisco1]
type=friend
disallow=all
allow=ulaw
;allow=alaw
;allow=g729
username=cisco1
secret=cisco1
;host=sip
host=dynamic
;dtmfmode=outband
canreinvite=no
mailbox=1508
;context=sip
qualify=yes
callerid="Heison Chak" <1508>
By: heison (heison) 2004-12-23 07:44:55.000-0600

thanks oej for pointing out... it was in fact a config mistake. nat=yes was in sip.conf and triggered by user->nat = global_nat;
By: Olle Johansson (oej) 2004-12-23 07:47:50.000-0600

Ok, the user was affected by the patch in ASTERISK-3067 that changed the behaviour in his current installation. We've sorted this out over the IRC. It wasn't a bug, but introduction of a change in the logic that may look like a bug since there was no change in the config files.

I recommend no change of karma.
Heison: Please remember to report properly in the future, with all facts like SIp debug, config files. We could have sorted this out more quickly with a proper report. Thank you and merry xmas!