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Summary:ASTERISK-28556: pjsip blind transfer fails
Reporter:Remi Quezada (remiq)Labels:pjsip
Date Opened:2019-09-25 09:48:56Date Closed:2019-10-09 12:00:03
Priority:MinorRegression?No
Status:Closed/CompleteComponents:pjproject/pjsip
Versions:13.28.1 Frequency of
Occurrence
Related
Issues:
Environment:Centos 6Attachments:( 0) debug_log_bxfer
Description:I am getting the following error when I blind transfer calls using the conference URI feature in the Cisco 504G and CP-7841-3PCC phones.

[Sep 25 10:31:02] ERROR[31736] res_pjsip_refer.c: Channel 'PJSIP/106-eng5-00000034' from endpoint '106-eng5' attempted blind transfer to 's@sip-outbound-eng5' but target does not exist

Both phone models do not work.  This feature allows you to conference more than 3 calls on your phone by blind transferring calls to a feature code that goes to a conference room.  This was working in asterisk 1.8 using same dialplan when I upgraded to asterisk 13 this stopped working.  Attached are debug logs.
Comments:By: Asterisk Team (asteriskteam) 2019-09-25 09:48:57.091-0500

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.

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By: Joshua C. Colp (jcolp) 2019-09-25 09:56:32.683-0500

The message is quite clear - it tried to blind transfer but no extension exists. What is the dialplan? Have you examined it?

Were you using chan_sip previously with the Cisco patch? That may have added supported for this, which does not exist in PJSIP.
By: Remi Quezada (remiq) 2019-09-25 10:16:05.296-0500

I've examined the dialplan it looks good and same dialplan is working with asterisk 1.8 and chan_sip.  I am not using any custom Cisco patch in chan_sip that is not included the asterisk release.  I don't think it's supposed to be transferring to 's' exten because that is not defined in my dialplan for [sip-outbound-eng5] instead it should send it to *11 or the actual extension.
By: Joshua C. Colp (jcolp) 2019-09-25 10:23:29.562-0500

From the trace you've provided:

{noformat}
Refer-to: <sip:209.191.9.19:5060>
{noformat}

So it is not specifying an extension (in the form of a user in the URI) to blind transfer to thus the default of "s" is used. Asterisk appears to be operating as it should with it.
By: Remi Quezada (remiq) 2019-09-25 11:19:03.065-0500

I think the reason why the phone refers it there is because when it gets the 100 Trying and OK from asterisk for the call that it's trying transfer to the contact is set to:
{noformat}
Contact: <sip:209.191.9.19:5060>
{noformat}

In asterisk 1.8 chan_sip it sets it to:
{noformat}
Contact: <sip:*11@209.191.9.19:5060>
{noformat}
By: Joshua C. Colp (jcolp) 2019-09-25 11:23:10.973-0500

You can try setting the "contact_user" on the endpoint, otherwise there's no real bug here. The Contact doesn't have to contain such information. It's used for in-dialog requests, the endpoint is choosing to use it for this conference call scenario which is not normal.
By: Asterisk Team (asteriskteam) 2019-10-09 12:00:03.390-0500

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