Dial with Flag T = No Audio!

1

Hello!

In the dialplan I am connecting the caller via Dial(SIP/XXXX,30,Tt) to an extension. If I do that with the Flags “Tt” the caller voice is not beeing transmitted! If I do the same thing without the flags it works like a charm. What could be the reason for this? I need to allow the call to be transferred.

Thanks a lot in advance!

2

Is Asterisk behind NAT? Do you have direct media enabled? What version of Asterisk?

It may be that if direct media is enabled both sides are told to talk direct which works. With Tt this can’t happen so media has to flow through Asterisk, which is not getting the media.

3

Hey jcolp. Thanks for your reply.

Asterisk is not behind NAT.
Direct media is not enabled.
Asterisk Version is 13.8.1.

Here is my sip-config:
[general]
port=5060
context=unauthenticated
allowguest=no
alwaysauthreject=yes
language=de
autofallthrough=yes
registertimeout=20
transport=udp
allowoverlap=dtmf
dtmfmode=auto
disallow=all
allow=alaw
allow=gsm

[trunk1]
type=peer
host=fpbx.de
fromdomain=fpbx.de
port=5060
username=XXX
fromuser=XXX
secret=XXX
context=inbound
canreinvite=yes
language=de
dtmfmode=auto
disallow=all
allow=alaw
allow=gsm
4

I’d suggest making a complete log accessible with SIP debug (sip set debug on). As well you do appear to have direct media on via “canreinvite=yes” and in the other case the absence of it defaults to on.

5

Not relevant to the problem, but autofallthrough is not a sip.conf option.

6

I was not able to get to the bottom of this. I was even setting up a configuration with only a very basic configuration but it did not work.

The VoIP-Phones of my customer are having a Transfer-Button und that works just fine. I only had to define a few extensions in the dailplan.