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<body style="text-align:left; direction:ltr;"><div>Hi Tom,</div><div><br></div><div>thanks for Your answer.</div><div><br></div><div>Am Donnerstag, dem 02.03.2023 um 08:18 -0500 schrieb Tom Ray:</div><blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex">
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<div dir="auto">On Mar 2, 2023 at 8:05 AM -0500, Karsten Wemheuer <kwem@mail.de>, wrote:</div>
<blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex">Hi *,<br>
<br>
Maybe I found a small bug or I am doing something wrong.<br>
<br>
When I do a "Transfer" on a call that arrives via PJSIP, Asterisk sends<br>
a "302 Moved Temporarily" to perform the transfer.<br>
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<div dir="auto">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?<br></div></div></blockquote><div><br></div><div>Sorry, I mean: I use the dialplan application "transfer" to do a 302 Redirect.</div><div><br></div><div>With chan_sip it was</div><div> Transfer +49xxx</div><div>This does not work with pjsip, so I use</div><div> Transfer sip:<a href="mailto:+49xxx@ip-address">+49xxx@ip-address</a></div><div>or</div><div> <span style="font-size: 13.333333px;">Transfer sip:</span><a href="file:///%3Ca%20href=%22file:///+49xxx@do%22%3E+49xxx@domain%3C/a%3E.t" style="font-size: 13.333333px;"><a href="file:///+49xxx@do">+49xxx@domain</a>.tld</a></div><div><br></div><blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex"><div name="messageReplySection">
<blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex">Unlike chan_sip, the contact header is set different and maybe<br>
incorrectly with PJSIP:<br>
<br>
chan_sip:<br>
Contact: Transfer <sip:+49xxx@provider.de><br>
<br>
pjsip:<br>
Contact: <sip:+49170xxx@91.2.166.143:5060><br>
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<div dir="auto">We will probably need to see actual SIP debugs and SIP messages so we can see how this is being sent to the carrier.<br></div></div></blockquote><div><br></div><div>Doing this in dialplan</div><div> Transfer <sip:<a href="mailto:+491708300432@tel.t-online.de">+491708300432@tel.t-online.de</a>></div><div><br></div><div>I got this with ngrep:</div><div><font color="#0000ff">T 192.168.10.70:59371 -> 217.0.149.48:5060 [AP] #9</font></div><div><font color="#0000ff">SIP/2.0 302 Moved Temporarily.</font></div><div><font color="#0000ff">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.</font></div><div><font color="#0000ff">Record-Route: <sip:<a href="mailto:mavodi-0-266-5eb-4-ffffffff-dc390000-5f3cccc047ab3-a81-ffffffffffffffff-mavsipodi-0-26c-d5-4-cf870000-5f3cccbfe97f2-a81@217.0.149.48">mavodi-0-266-5eb-4-ffffffff-dc390000-5f3cccc047ab3-a81-ffffffffffffffff-mavsipodi-0-26c-d5-4-cf870000-5f3cccbfe97f2-a81@217.0.149.48</a>:5060;transport=tcp;lr;mavsipodi-0-26c-d5-4-cf870000-5f3cccbfe97f2-a81>.</font></div><div><font color="#0000ff">Call-ID: <a href="mailto:BW171319464020323-1506406339@62.156.74.66">BW171319464020323-1506406339@62.156.74.66</a>.</font></div><div><font color="#0000ff">From: <sip:<a href="mailto:+492414012948@unitymedia.de">+492414012948@unitymedia.de</a>;user=phone>;tag=1594730888-1677773599464-.</font></div><div><font color="#0000ff">To: <sip:<a href="mailto:+4922842208399255@tel.t-online.de">+4922842208399255@tel.t-online.de</a>;user=phone>;tag=ff7ed316-6fcf-40e6-ae35-c4077a100999;cscf.</font></div><div><font color="#0000ff">CSeq: 294435189 INVITE.</font></div><div><font color="#0000ff">Server: Asterisk.</font></div><div><font color="#0000ff">Contact: <sip:<a href="mailto:+491708300432@91.2.166.143">+491708300432@91.2.166.143</a>>.</font></div><div><font color="#0000ff">Reason: Q.850;cause=0.</font></div><div><font color="#0000ff">Supported: histinfo.</font></div><div><font color="#0000ff">Content-Length: 0.</font></div><div><br></div><div>I am using Asterisk 18.16</div><div><br></div><div>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.</div><div><br></div><div>Thanks,</div><div>Karsten</div><blockquote type="cite" style="margin:0 0 0 .8ex; border-left:2px #729fcf solid;padding-left:1ex"><div name="messageReplySection">
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