Attended transfer fails if destination no answer


I have FreePBX with extensions in PJSIP.

External call A goes to an extension B. B make an attended transfer to C. When C is busy or do not answer B do not recover the A call. A and B are both hanged up.

I tried to debug but do not found what fails. There is any option I can check to change this behaviour?



Are you sure this is an attended transfer and waiting for the answer, or are you using an attended transfer and just hanging up when the phone starts to ring?

If the former, that’s a bug.

If it’s the latter, then that’s the way the system works. If you transfer the call to a phone that is busy or there is no answer, and there is no Voicemail attached to the phone, the default action for the call is to hang up.

There are many possible solutions, including FMFM, voicemail blasting, actual voicemail, etc.

Yes, I’m shure. In can see it in the logs:

The call comes from outside to extension via a queue:

[2018-03-07 10:17:22] VERBOSE[20736][C-000006b9] pbx.c: Executing [[email protected]:52] Dial("Local/[email protected];2", "PJSIP/2100/sip:[email protected]:5060,,HhtrM(aut o-blkvm)Ib(func-apply-sipheaders^s^1)") in new stack

the extension answer the call:

[2018-03-07 10:17:24] VERBOSE[20736][C-000006b9] app_dial.c: PJSIP/2100-00000eb8 answered Local/[email protected];2

After the dtmf, the attended transfer:

[2018-03-07 10:17:34] DEBUG[20737][C-000006b9] bridge_basic.c: Attended transfer to '[email protected]'

The call to the 2108:

`[2018-03-07 10:17:34] VERBOSE[20760][C-000006b9] pbx.c: Executing [[email protected]:52] Dial(“Local/[email protected];2”, “PJSIP/2108/sip:[email protected]:5060;user=phone,HhtrM(auto-blkvm)IIb(func-apply-sipheaders^s^1)”) in new stack

[2018-03-07 10:17:34] DEBUG[20581] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0x7f2db01c9ee0’

[2018-03-07 10:17:34] DEBUG[20581] res_pjsip_t38.c: Not creating outgoing SDP stream: T.38 not enabled

[2018-03-07 10:17:34] DEBUG[20581] res_pjsip_session.c: Sending session refresh SDP via re-INVITE to 2100

[2018-03-07 10:17:34] DEBUG[20581] res_pjsip_session.c: Method is INVITE

[2018-03-07 10:17:34] DEBUG[20581] res_pjsip/pjsip_message_filter.c: Re-wrote Contact URI host/port to (this may be re-written again later)

[2018-03-07 10:17:34] DEBUG[20760][C-000006b9] channel.c: Channel 0x7f2db0f62040 ‘PJSIP/2108-00000eb9’ allocated`

I was sniffing the traffic but I cannot fing this call to the 2108.

After some headers errors (I think freepbx try to get headers fom sip and pjsip allways and fails the sip because is a pjsip channel) and some changes in the bridges:

[2018-03-07 10:17:34] VERBOSE[20760][C-000006b9] app_dial.c: Called PJSIP/2108/sip:[email protected]:5060;user=phone

After some more dialpan:

`[2018-03-07 10:17:34] DEBUG[13020] res_pjsip_session.c: Received response

[2018-03-07 10:17:34] DEBUG[13020] res_pjsip_session.c: Response is 486 Busy

[2018-03-07 10:17:34] DEBUG[20760][C-000006b9] channel.c: Channel 0x7f2db0f62040 ‘PJSIP/2108-00000eb9’ hanging up. Refs: 2.... [2018-03-07 10:17:41] DEBUG[20737][C-000006b9] bridge_basic.c: Transferer on attended transfer 0x7f2de025b1c0 hung up

And after that

`[2018-03-07 10:17:46] DEBUG[20736][C-000006b9] pbx.c: Launching ‘Hangup’

[2018-03-07 10:17:46] VERBOSE[20736][C-000006b9] pbx.c: Executing [[email protected]:6] Hangup(“Local/[email protected];2”, “”) in new stack

[2018-03-07 10:17:46] DEBUG[20736][C-000006b9] channel.c: Soft-Hanging (0x20) up channel ‘Local/[email protected];2’


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