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Summary:ASTERISK-16889: SIP 180 Sent from provider, Asterisk sends 180 and 183 w/SDP. No ringback on calls
Reporter:Ken Myers (nny)Labels:
Date Opened:2010-10-28 17:02:19Date Closed:2011-06-07 14:04:44
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:When calling through a provider, the provider sends 180 to asterisk, and asterisk sends 180 followed by 183 w/sdp to the phone. The caller hears no ringback. See attached screenshots

****** ADDITIONAL INFORMATION ******

Linux version 2.6.27.41-170.2.117.fc10.x86_64 (mockbuild@x86-4.fedora.phx.redhat.com) (gcc version 4.3.2 20081105 (Red Hat 4.3.2-7) (GCC) ) #1 SMP


SIP DEBUG http://pastebin.com/S1f6Yv0S

WIRESHARK data: http://i.imgur.com/dodwg.png http://i.imgur.com/g2KWC.png
Comments:By: Paul Belanger (pabelanger) 2010-10-29 07:21:15

Attach a debug log (see below)
---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Ken Myers (nny) 2010-10-29 12:17:49

Will do ASAP. The system is busy, any output right now would be convoluted by other calls.
By: Ken Myers (nny) 2010-11-01 14:02:29

The ITSP has changed their device to always send a 183. This has resolved the issue, however I still feel there is an underlying issue with how asterisk was responding. Unfortunately I will need some time to have them flip the switch back and duplicate the issue for a full debug. If you want to keep the issue open I will do my best to get it ASAP, but since this is a production system the timing and method will require some steps to ensure it doesn't affect the user's experiences.

Thanks


By: David Woolley (davidw) 2010-11-01 14:13:00

This may relate to something we have observed with Cisco CUCM 8.

When connecting to the PSTN, it is sending 180 with SDP, but sending silence (not silence suppression) in the RTP.

After signalling AST_CONTROL_RINGING, Asterisk seems to respond to the SDP by sending AST_CONTROL_PROGRESS.  In our case, party A is already answered, so there is no 183 to generate, but I suspecte 183 would be generated.

This is currently on our to do list to investigate, although, provisionally I am treating it as not being an Asterisk bug.  I'm not sure how the PSTN connectivity to the Cisco is achieved, and haven't investigated whether there are options to change the behaviour.

RFC 3960 suggests that giving the RTP stream priority over the 180 Ringing is reasonable behaviour, although doesn't mandate it, calling it a local policy decision.

The pastebin version of the debugging info for this issue doesn't seem to have any entries for the B side, so I can't tell whether this is also 180 + SDP.

In our case, we are using a (patched) 1.6.1.0.
By: David Woolley (davidw) 2010-11-04 06:30:30

Re comment %128510 we reproduced the no ringback symptom using a skinny phone directly on CUCM with no Asterisk invovlement.  Pending traces of the downstream side for the original reporter, I can't say whether that is relevant here, or not.
By: Leif Madsen (lmadsen) 2011-01-18 09:49:10.000-0600

I'm closing this issue for now. If someone can reproduce this easily and provide the relevant traces (uploaded as text files, attached to this issue as a file) then please re-open this issue.

Thanks!