No joint capabilities for 'audio' media stream between our configuration((ulaw)) and incoming SDP((g722)). But all codecs are allowed

Asterisk Asterisk SIP
1

Hello everyone. I currently maintain one small asterisk sip-trunc. It works great so far but i still have this one little problem. Sometimes when a caller dials our number we can here the call getting in, but as soon as someone accepts the call, the other side only hears the occupied signal. Sometimes this also happens when dialing out. I don’t really understand what the error is telling me and when searching the only solution i found was adding allow=all to all endpoints to allow all codecs. How can codec negotiation fail if i allow all codecs? Can someone help me here to at least demystify the problem?

Thanks in advance and have a nice day.

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You’d have to show the actual PJSIP endpoint configuration as well as SIP trace (pjsip set logger on) - but it is showing exactly as it states, the endpoint was configured for ulaw but the call offered only g722. Enabling ALL codecs is generally a bad idea because there’s too many of them, so it increases the size of the INVITE, and you can actually run out of payloads to be able to support them all. You should only allow the specific codecs you expect/want.

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Thank you. I also forgot to mention i am on Debian 10 and use the apt version of Asterisk (16.16.1).
I get why having big invite requests is bad but the allow=all was more of a test to get it working which didn’t work,
and we don’t really know what codecs will be used to call us.
I still don’t get why asterisk says that the endpoint is not configured for given codec even though all are allowed.
My thought was that allowing all would use the one the other side requested and then transcode internally.

The configuration of the endpoint that didn’t work is:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[netcologne-siptrunk]
type=endpoint
context=netcologne-ctx
allow=all
outbound_auth=netcologne-auth
aors=central,person1-softphone,person2-softphone
transport=transport-udp
timers=no

[central]
type=aor
max_contacts=1
contact=sip:dfs.netcologne.de:5060

We also can’t really reproduce the problem as it just works with all devices we checked with.

I am sorry but i don’t understand how to activate the pjsip logger/where i can find the logs after running
pjsip set logger on

Thank you

4

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

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