Attended Transfer with interference signals

Hello,

my Asterisk 22.0.0 is up and running.
Connected with Sipgate (German SIP provider).
Normal incomming and outgoing calls are without any problems, the quality is G.711, but I acept G.722 too.

In the folling szenario sometimes the audio is with interferences - 50/50 mixed like wrong codec.
It sounds like mixed alaw with ylaw or 50% is set to high.

Caller External is ralling the Asterisk via Sipgate.
User a is Answering and speaking - clear without any interferences
User a (on Snom D385) is pressing the ATXfer button on Snom (ATXFER ist starting)
External is heraing dubidubidu - MOH is playing.
User a is dialing User b (on Snom D385 too or MicroSIP on PC)
User a is speaking with User b “there is external in the line, I will stop an you can speak”
User a is prssing the transfer Button on D385,
User a is ended
External stopped MOH and connect to User b.
User b can speak with External with massive interferences
External can speak with User b with massive interfrerences

User b hang up the line and dials out - external is ringing and the connection is without any problems.

I suspect that the MOH pulls the external participant to some COdec and does not come back after connecting to b.

On Tuesday 12 November 2024 at 17:08:26, humberg via Asterisk Community wrote:

I suspect that the MOH pulls the external participant to some COdec and
does not come back after connecting to b.

I also think this is quite likely.

Do a SIP packet capture (eg: with sngrep) and see what the SDP negotiations
show.

Antony.


1960s: Let’s build a network which can withstand a nuclear war!
1970s: Hm, that looks good, we’ll run it on TCP/IPv4.
1980s: Nice, how about letting everyone join?
1990s: Hey, you can make money out of this!
2000s: Oh, you can lose it, too.
2010s: Alright, let’s just plug absolutely everything into it.
2020s: Meh, my lightswitch is now connected to my lamp via China.

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