Summary:	ASTERISK-28522: chan_pjsip does not support fallback from t.38 to fax over alaw/ulaw
Reporter:	John Cahill (jcahill)	Labels:	fax pjsip
Date Opened:	2019-09-04 06:58:11	Date Closed:	2019-09-18 12:00:03
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip_t38
Versions:	16.5.0	Frequency of Occurrence
Related Issues:
Environment:	Debian 10.0, 4.9.0-3-amd64 2GB RAM, 4 VCPUs, Xen 4.4 Dell PowerEdge R430	Attachments:	( 0) failed-fallback.pcap
Description:	chan_pjsip does not support fallback from t.38 to fax over alaw/ulaw. Asterisk should re-invite to alaw/ulaw on receiving a t.38 to allow fax to proceed. This works with chan_sip.
Comments:	By: Asterisk Team (asteriskteam) 2019-09-04 06:58:12.613-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: John Cahill (jcahill) 2019-09-04 06:59:25.478-0500 pcap of failure By: Joshua C. Colp (jcolp) 2019-09-04 07:51:59.048-0500 Can you also provide a working case from chan_sip as well as the Asterisk console output? According to the SIP trace you've provided T.38 was never negotiated at all (the attempts were rejected), so there would be no need to re-invite back to audio because the session never changed away from audio in the first place. By: Asterisk Team (asteriskteam) 2019-09-18 12:00:02.992-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines