[Home]

Summary:ASTERISK-21047: asterisk crashes during an attended transfer SIP -> SIP -> DAHDI
Reporter:piero ferraresso (ferrored)Labels:
Date Opened:2013-02-07 08:33:42.000-0600Date Closed:2013-05-01 17:21:57
Priority:CriticalRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_dahdi Channels/chan_sip/Transfers Resources/res_rtp_asterisk
Versions:11.2.0 11.2.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:debian 5.0.10, opensips v.1.8.2, digium te420B, PBX Aastra MD110Attachments:( 0) backtrace.txt
( 1) chan_dahdi.conf
( 2) messages.txt
( 3) messages-sip-debug-on.txt
( 4) sip.conf
Description:I'm trying to make an attended transfer.
Sip hardphone A call sip hardphone B (both Polycom SPIP 450);
B press the 'transfer' softkey and compose the number of legacy hardphone C (PBX Aastra connected to asterisk with a Qsig trunk using a digium te420B card).
After legacy hardphone C answered,  B press again the 'transfer' softkey to transfer the call to phone A. At this point asterisk crashes.


Ps The some scenario using asterisk 1.4 works
Comments:By: piero ferraresso (ferrored) 2013-02-07 08:36:35.946-0600

backtrace file created according to https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
By: Rusty Newton (rnewton) 2013-02-14 19:34:28.106-0600

Thanks for the report. Please attach a full log with VERBOSE and DEBUG set to level 5 showing the call where the crash occurs. Also, please attach sanitized sip.conf, dahdi.conf and chan_dahdi.conf files. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: piero ferraresso (ferrored) 2013-02-15 03:27:52.159-0600

Files attached
By: Matt Jordan (mjordan) 2013-03-07 10:07:00.538-0600

Please click 'Send Back' when you've provided feedback. Otherwise, the issue stays in the 'Waiting for Feedback' status and Bug Marshals may not see that you've provided the information requested.
By: Rusty Newton (rnewton) 2013-03-08 18:49:38.500-0600

Piero, I see some "bad magic number" errors right before the crash, can you reproduce the issue again with reference count debugging enabled for chan_sip? https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging

Gather the /tmp/refs after reproduction and attach to the issue.


By: Rusty Newton (rnewton) 2013-03-08 18:50:52.023-0600

Another question - can you ever reproduce this crash when transferring to a SIP extension rather than the DAHDI extension? And does this only occur with the DAHDI extension shown, or any DAHDI extension?
By: piero ferraresso (ferrored) 2013-03-11 05:08:11.545-0500

Hi Rusty, I've followed the tutorial but I get "Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: sip_unref_peer" when I try to load chan_sip.so module.

About the other questions, the crash occurs when transferring to any SIP or DAHDI extensions.
By: Matt Jordan (mjordan) 2013-03-12 12:07:52.932-0500

I'm confused. If {{chan_sip}} isn't loaded, how are you performing a transfer to a SIP device?
By: piero ferraresso (ferrored) 2013-03-13 04:35:23.089-0500

Logically I made the tests without compiling asterisk with the option "#define REF_DEBUG 1" that cause the error "Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: sip_unref_peer".
By: Rusty Newton (rnewton) 2013-04-01 17:12:09.861-0500

So it sounds like following the compilation changes for reference count debugging results in you being unable to load chan_sip. I'm not sure why that is happening.

Have you tried re-extracting Asterisk from a fresh tarball before attempting the refrence count debugging customizations and compilation ?

Hopefully someone will be able to investigate without the reference count debug if you can't set it up.
By: piero ferraresso (ferrored) 2013-04-02 02:13:48.106-0500

I've just downloaded the version 11.3.0, but I still get the previous error after setting the reference count debugging customizations
By: Rusty Newton (rnewton) 2013-04-08 21:43:50.224-0500

I see the same issue with the ref count debugging. I filed issue ASTERISK-21391 for it.
By: Rusty Newton (rnewton) 2013-04-09 18:43:32.527-0500

piero, https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging has been updated. See the new step 4 under "Enabling Reference Count Debugging for a Specific Module". You need to also enable REF_DEBUG within the channels/sip/security_events.c file.
By: piero ferraresso (ferrored) 2013-04-19 11:11:12.061-0500

Sorry, but I still get the same error with version 11.2.1 and 11.3.0.
Instead with version 11.4.0-rc1 I am capable of setting the reference count debugging customizations, and, whats more, I notice that the scenario of attended transfer SIP -> SIP -> DAHDI doesn't crash asterisk anymore. However it seems that there are some minor issues in the transfer scenario. I'll make some test and I'll let you know.

By: Rusty Newton (rnewton) 2013-04-25 14:08:03.925-0500

If I understand you correctly:

Can you confirm that with 11.4.0-rc1 you no longer experience the crash reported in this issue?

If this is accurate, then we will close the issue as it was likely fixed in another issue. If you have other problems you feel are a bug then you can file a new issue, but you'll probably want to post on the asterisk-users list first to have others confirm.

By: piero ferraresso (ferrored) 2013-04-30 03:52:37.873-0500

I confirm that with 11.4.0-rc1 the crash no longer occurs.

I'll follow your indication
Thank you