Call drops on attended transfer

I’m having an issue with calls dropping during an attended transfer with an external call. Internal calls transfer normally.

The system is a fresh install (yesterday) of FreePBX running on VMWare. Endpoint devices are SPA504G, using the OSS Endpoint Manager to configure. All modules are full updated, and the OS is fully yum-updated.

During the process, the external caller receives the receptionist through an IVR normally, and the call is fine. The receptionist then transfers the call using the “xfer” button on the SPA504G (xfer-ext-dial-xfer). The call then rings on the target, the target hears the ring and picks up, then the call is dropped. After about 10 seconds, the remote call drops as well. The remote caller hears a ring during the process.

We have verified that bxfer does work correctly, as a workaround.

I don’t see anything obviously wrong in the asterisk log. When the transfer happens I see:

[2018-12-07 14:01:54] VERBOSE[9047][C-00000070] app_dial.c: PJSIP/TGTEXT-000000c5 answered PJSIP/inbound-000000c2
[2018-12-07 14:01:54] VERBOSE[9051][C-00000070] bridge_channel.c: Channel PJSIP/TGTEXT-000000c5 joined 'simple_bridge' basic-bridge <4172341b-6699-456f-94b4-097153401453>
[2018-12-07 14:01:54] VERBOSE[9047][C-00000070] bridge_channel.c: Channel PJSIP/inbound-000000c2 joined 'simple_bridge' basic-bridge <4172341b-6699-456f-94b4-097153401453>
[2018-12-07 14:02:09] VERBOSE[9051][C-00000070] bridge_channel.c: Channel PJSIP/TGTEXT-000000c5 left 'simple_bridge' basic-bridge <4172341b-6699-456f-94b4-097153401453>
[2018-12-07 14:02:09] VERBOSE[9047][C-00000070] bridge_channel.c: Channel PJSIP/inbound-000000c2 left 'simple_bridge' basic-bridge <4172341b-6699-456f-94b4-097153401453>

just before the call hangs up.

Any suggestions?

I’d start with a codec mismatch. You might also want to turn up the SIP DEBUG.

A few (say 20) more lines before and after the transfer might also give us a clue. Other than that, I don’t think any of us has enough information.

I saw something like this last week. You are using PJSIP trunks, is this something that worked fine until recently? If so change the trunk parameter ‘rewrite contact’ and see if that helps. If not, set it back and investigate your SIP signalling with wireshark or sngrep.

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.