Hi<br><br>Sorry but please ignore last post.<br>At present I have put force_rport in main route.<br><br>I have registered 4 UAC using same username .<br>Among them 2 are on public ip & the other 2 are behind nat.<br>now when I am calling to PSTN signalling & audio both works properly.<br>
<br>But when getting incoming call from PSTN & pickup call using the UAC which is behind NAT I am getting NO audio.<br><br>It works if I register single UAC using a username.<br>I am having an issue when I do fork.<br>
<br>Please advice what can be the issue.<br><br>Thanks<br>Krunal Patel<br><br><div class="gmail_quote">On Tue, Dec 16, 2008 at 7:36 PM, Krunal Patel <span dir="ltr"><<a href="mailto:krunal.lists@gmail.com">krunal.lists@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br><br>I am doing force_rport.<br>& that's why external port is being added in via <br>
Via: SIP/2.0/UDP [INTERNAL_IP]:5065;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-a85eb9f3;rport=50034.<br><br>But from reply route, reply is not being sent to correct port which is in via.<br>
Please have a look to below given 183 progress trace <br><div class="Ih2E3d"><br>U YYY.YYY.YYY.YYY:5060 -> XXX.XXX.XXX.XXX:5065<br>SIP/2.0 183 Session Progress.<br>Call-ID: df9cc79e-ffa2ffb3@[INTERNAL_IP].<br>CSeq: 102 INVITE.<br>
</div>From: "FROM_NUM" <<a href="mailto:sip%3AFROM_NUM@voip.convergenze.it" target="_blank">sip:FROM_NUM@voip.convergenze.it</a>>;tag=45a59ea2bc86747fo0.<br>
To: <<a href="mailto:sip%3ATO_NUM@voip.convergenze.it" target="_blank">sip:TO_NUM@voip.convergenze.it</a>>;tag=3eddfbdb7400749.<br>Via: SIP/2.0/UDP [INTERNAL_IP]:5065;received=XXX.XXX.XXX.XXX;branch=z9hG4bK-a85eb9f3;rport=50034.<br>
<br>Let me know hwat can be the issue?<div><div></div><div class="Wj3C7c"><br><div class="gmail_quote">On Tue, Dec 16, 2008 at 5:39 PM, Krunal Patel <span dir="ltr"><<a href="mailto:krunal.lists@gmail.com" target="_blank">krunal.lists@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi<br>I put force_rport at the beginning of main route.<br>now signaling issue is fixed but facing audio issue.<br>
& getting error: empty response from mediaproxy for PSTN to behind nated users.<br><br><br><br><div class="gmail_quote"><div><div></div><div>
On Mon, Dec 15, 2008 at 7:58 PM, Jeff Pyle <span dir="ltr"><<a href="mailto:jpyle@fidelityvoice.com" target="_blank">jpyle@fidelityvoice.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><div></div><div>
<div>
<font face="Tahoma, Verdana, Helvetica, Arial"><span style="font-size: 10pt;">Iñaki,<br>
<br>
Some time back I had a lab setup that used force_rport() on everything that came in. This caused problems if the INVITE came from a Broadworks server. Broadworks does not use source port 5060 on its traffic, for load handling they say. Anyway, my replies went to the source port Broadworks used instead of to 5060. And, of course, failed.<br>
<br>
I never did spend time on how to handle this properly. Perhaps someone here has some thoughts.<br><font color="#888888">
<br>
<br>
- Jeff</font><div><br>
<br>
<br>
<br>
<br>
On 12/15/08 8:05 AM, "Iñaki Baz Castillo" <<a href="http://ibc@aliax.net" target="_blank">ibc@aliax.net</a>> wrote:<br>
<br>
<font color="#0000ff">> El Lunes, 15 de Diciembre de 2008, Krunal Patel escribió:
> <br>
> force_rport();
Use force_rport(); at the beginning of your script, just <br>
> there.
And take if off from on_reply_route.
--
Iñaki Baz Castillo
</font></div></span></font>
</div>
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