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

[Karsten Wemheuer]

Hi Joshua,

thank You, for answering.

Am Donnerstag, dem 02.03.2023 um 09:17 -0400 schrieb Joshua C. Colp:
> On Thu, Mar 2, 2023 at 9:04 AM 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.
> 
> What version of Asterisk? What is the precise transport
> configuration?
As written below it was Version 18. The exact version is 18.16.0.

> > 
> > 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>
> > 
> > 
> > 
> > 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.
> > 
> > 
> > 
> > Is this a bugf or how to use transfer application in combination
> > with
> > 
> > pjsip?
> 
> For questions like this in the future please use either the asterisk-
> users mailing list or the community forum[1].
I had used the forum because the user mailing list was not working.
Unfortunately the replies were not helpful and I am not sure if it is a
bug. How can I make the Contact header contain a URI with domain part
(instead of ip address)?

Thanks,

Karsten


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