Chan_sip direct rtp codec negotiation problem

Hi, we have a problem with Asterisk 13.31, but it’s also in older versions.

We have the following setup. A telephone dials into Asterisk A where it’ll get a “Hello World” message before being forwarded to Asterisk B and then Asterisk C. (this is actually a setup to diagnose another problem)

Asterisk C has sipmode=inband set so it won’t accept telephone-events. Asterisk A and B have sipmode=auto set.

So Asterisk A sends an invite with telephone-events to Asterisk B. The OK has telephone-events sets. Then Asterisk B plays the prompt and relays the call to Asterisk C. There the INVITE includes telephone-events, but the OK doesn’t. Asterisk B realizes that it no longer needs to be in the stream, therefore it sends INVITES to both Asterisk A and Asterisk C. The INVITE to Asterisk C contains no telephone-events (as it’s not supported), but the INVITE to Asterisk A does. Therefore Asterisk A believes that Asterisk C wants telephone-events and sends them.

Can you change Asterisk B to sipmode=inband ?

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Well it’s a bit different. Asterisk B now doesn’t offer telephone-events any more to Asterisk C. Still the re-invite to Asterisk A contains telephone-events even though Asterisk should know that C doesn’t speak telephone-events.

Is there a peer defined for C on A with correct settings ? And vice-versa (A on C) ?

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