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

[Karsten Wemheuer]

Hi Tom,
thanks for Your answer.
Am Donnerstag, dem 02.03.2023 um 08:18 -0500 schrieb 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?

Sorry, I mean: I use the dialplan application "transfer" to do a 302
Redirect.
With chan_sip it was	Transfer +49xxxThis does not work with pjsip,
so I use	Transfer sip:+49xxx at ip-addressor	Transfer sip:
+49xxx at domain.tld
> > 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.

Doing this in dialplan	Transfer <sip:+491708300432 at tel.t-online.de>
I got this with ngrep:T 192.168.10.70:59371 -> 217.0.149.48:5060 [AP]
#9SIP/2.0 302 Moved Temporarily.Via: SIP/2.0/TCP
217.0.149.48:5060;rport=5060;received=217.0.149.48;branch=z9hG4bKmavodi
-0-264-c43-4-1000000-42600000-5f3cccc047d72-a81-ffffffffffffffff-d5-
cf870000-5f3cccbfe97f2-459902267-5827.Record-Route: <sip:
mavodi-0-266-5eb-4-ffffffff-dc390000-5f3cccc047ab3-a81-ffffffffffffffff-mavsipodi-0-26c-d5-4-cf870000-5f3cccbfe97f2-a81 at 217.0.149.48
:5060;transport=tcp;lr;mavsipodi-0-26c-d5-4-cf870000-5f3cccbfe97f2-
a81>.Call-ID: BW171319464020323-1506406339 at 62.156.74.66.From: <sip:
+492414012948 at unitymedia.de;user=phone>;tag=1594730888-1677773599464-
.To: <sip:+4922842208399255 at tel.t-online.de;user=phone>;tag=ff7ed316-
6fcf-40e6-ae35-c4077a100999;cscf.CSeq: 294435189 INVITE.Server:
Asterisk.Contact: <sip:+491708300432 at 91.2.166.143>.Reason:
Q.850;cause=0.Supported: histinfo.Content-Length:  0.
I am using Asterisk 18.16
Is it possible to put a uri with a domain instead of ip address  in
contact header?  The provider gave me an example where the only obvious
difference is the portion after the @ sign in the Contact header.
Thanks,Karsten
> 
> 
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