Atxferabort not working

Asterisk Asterisk Support
1

Hi to all,
I’d like to have the possibility to get back a call I had initiated an attended transfer for. This should be possible by the atxferabort feature, right?

So, in my features.conf I have

[featuremap]
atxfer => *2 ; Attended transfer – Make sure to set the T and/or t option in the Dial() or Queue() app call!

and

[general]
atxferabort = *1 ; cancel the attended transfer
atxfercomplete = *2 ; complete the attended transfer, dropping out of the call
atxferthreeway = *3 ; complete the attended transfer, but stay in the call. This will turn the call into a multi-party bridge
atxferswap = *4 ; swap to the other party. Once an attended transfer has begun, this options may be used multiple times

Initiating the transfer works fine, but when I press *1 nothing happens but asterisk telling me that I chose an invalid number.

features show

shows

[asterisk*CLI> features show
Builtin Feature Default Current

Pickup *8 *8
Blind Transfer # #
Attended Transfer *2
One Touch Monitor
Disconnect Call * *
Park Call
One Touch MixMonitor

Dynamic Feature Default Current

(none)

Feature Groups:

(none)

When I set debug on, I can see, that the dumf-tones are submitted:

<— SIP read from TCP:telephone_IP:5066 —>
INFO sip:caller_phonenumber@asterisk_IP:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP telephone_IP:5066;branch=z9hG4bK-500b6371
From: <sip:telephone_definition_in_sip.conf @telephone_IP;transport=tcp>;tag=a7b630daebf61d55i0
To: <sip:caller_phonenumber @asterisk_IP>;tag=as10b79346
Call-ID: 6cd58e395d364c11680b452d27e69d39@asterisk_IP:5060
CSeq: 103 INFO
Max-Forwards: 70
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 24
Content-Type: application/dtmf-relay

Signal=*
Duration=100
<------------->
— (10 headers 2 lines) —
Receiving INFO!
* DTMF-relay event received: *

<— Transmitting (NAT) to telephone_IP:5066 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP telephone_IP:5066;branch=z9hG4bK-500b6371;received=telephone_IP;rport=5066
From: <sip:telephone_definition_in_sip.conf @telephone_IP;transport=tcp>;tag=a7b630daebf61d55i0
To: <sip:caller_phonenumber @asterisk_IP>;tag=as10b79346
Call-ID: 6cd58e395d364c11680b452d27e69d39@asterisk_IP:5060
CSeq: 103 INFO
Server: Asterisk PBX 15.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>

<— SIP read from TCP:telephone_IP:5066 —>
INFO sip:caller_phonenumber @asterisk_IP:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP telephone_IP:5066;branch=z9hG4bK-58936c6
From: <sip:telephone_definition_in_sip.conf @telephone_IP;transport=tcp>;tag=a7b630daebf61d55i0
To: <sip:caller_phonenumber @asterisk_IP>;tag=as10b79346
Call-ID: 6cd58e395d364c11680b452d27e69d39@asterisk_IP:5060
CSeq: 104 INFO
Max-Forwards: 70
User-Agent: Cisco/SPA504G-7.4.8a
Content-Length: 24
Content-Type: application/dtmf-relay

Signal=1
Duration=100
<------------->
— (10 headers 2 lines) —
Receiving INFO!
* DTMF-relay event received: 1

but further:

<------------>
Scheduling destruction of SIP dialog ‘44b83dca-9ec80f9a@telephone_IP’ in 32000 ms (Method: NOTIFY)
– Channel SIP/telephone_definition_in_sip.conf-00000088: Dialed ‘*1@default’ does not exist.
– <SIP/telephone_definition_in_sip.conf-00000088> Playing ‘pbx-invalid.slin16’ (language ‘de’)

Can anybody point me to my mistake?

2

BTW, I set the “t”-option in the dial()-application:

– Executing [021@INTERNAL:4] Dial(“SIP/TELEKOM-00000087”, “SIP/_telephone_definition_in_sip.conf,20,mkt”) in new stack