[asterisk-dev] Call transfer (302 Moved temporarily) not working with pjsip

[Tom Ray]

On Mar 2, 2023 at 8:05 AM -0500, Karsten Wemheuer <kwem at mail.de>, wrote:
> Hi *,
>
> Maybe I found a small bug or I am doing something wrong.
>
> When I do a "Transfer" on a call that arrives via PJSIP, Asterisk sends
> a "302 Moved Temporarily" to perform the transfer.
>
For the record, this isn't a transfer it is a redirect. They are completely different things. The first thing we would need to know is how you are doing this. Are you immediately using the redirect features in Asterisk to send back a 302 or is more happening that results in a 302 being done?
> Unlike chan_sip, the contact header is set different and maybe
> incorrectly with PJSIP:
>
> chan_sip:
> Contact: Transfer <sip:+49xxx at provider.de>
>
> pjsip:
> Contact: <sip:+49170xxx at 91.2.166.143:5060>
>
We will probably need to see actual SIP debugs and SIP messages so we can see how this is being sent to the carrier.
> The difference are domain (chan_sip) vs. local IP address (pjsip) and
> the additional (wrong?) port number. The IP address is the one of my
> router, but the port number should be 25060, because asterisk is
> configured to use this port.
>
> The transfer works with asterisk 11 and chan_sip. It does not work with
> pjsip and asterisk 18. My provider does not accept the transfer done
> with pjsip. Either the provider expects the domain in the contact
> header or the error is in the wrong port number.
>
Well you should confirm with the carrier how they expect this. What is exactly needed for this to work. Guessing that it might be the contact header or a wrong port will just make extra work to figure out when the provider can give more direct answers.
> Is this a bugf or how to use transfer application in combination with
> pjsip?
>
> Thanks
>
> Karsten
>
>
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