yes, actually this scenario is on remote servers. like<br><br> SIP/<a href="http://XYZ@119.18.230.20:5060">XYZ@119.18.230.20:5060</a> <br> SIP/<a href="http://XYZ@202.68.0.90:5678">XYZ@202.68.0.90:5678</a> <br>
<br> audio is ok when dialing without using ip & port as below<br> <br> SIP/XYZ<br> <br> but when i dial using below dialstring<br><br> SIP/<a href="http://XYZ@202.68.0.90:5678">XYZ@202.68.0.90:5678</a> <br>
<br> or<br><br> SIP/<a href="http://XYZ@119.18.230.20:5060">XYZ@119.18.230.20:5060</a> <br><br> then the problem arises<br><br> hope you got the idea..<br> <br> Nasir<br> <br><br> ------------------------------<br>
<br> > Message: 26<br> > Date: Thu, 15 Jul 2010 17:09:06 +0200<br> > From: Jonas Kellens <<a href="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</a>><br> > Subject: Re: [asterisk-users] One way audio when dialing multiple<br>
registrations<br> > To: Asterisk Users Mailing List - Non-Commercial Discussion<br> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br> > Message-ID: <<a href="mailto:4C3F2492.4040201@telenet.be">4C3F2492.4040201@telenet.be</a>><br>
> Content-Type: text/plain; charset="iso-8859-1"<br><br> > One-way audio is mostly firewall problem.<br><br> > Are you behind firewall ?<br><br> > You can check the audio-ports that are being used in the SDP-message by<br>
> doing a /sip debug/.<br><br> > Maybe you do not have enough UDP-ports open for the audio ?<br><br><br> > Jonas.<br><br><br> On 07/15/2010 04:38 PM, Nasir Javaid wrote:<br> >> Hi,<br> >><br>
>> I am working on calling 2 registrations of same user on 2 different ip<br> >> or ports. It works fine and both phones ring simultaneously. the<br> >> problem is that there is one way audio, calling party can hear me but<br>
>> i can't hear calling party.<br> >><br> > here is the scenario..<br> ><br> > SIP/<a href="http://XYZ@192.168.0.20:5060">XYZ@192.168.0.20:5060</a> <<a href="http://XYZ">http://XYZ</a>@<a href="http://192.168.0.20:5060">192.168.0.20:5060</a>><br>
> SIP/<a href="http://XYZ@192.168.0.10:5678">XYZ@192.168.0.10:5678</a> <<a href="http://XYZ">http://XYZ</a>@<a href="http://192.168.0.10:5678">192.168.0.10:5678</a>><br> ><br> > i dial using following dial string<br>
><br> > Dial(SIP/XYZ@192.168.0.20:5060&SIP/<a href="http://XYZ@192.168.0.10:5678">XYZ@192.168.0.10:5678</a><br> > <<a href="http://XYZ">http://XYZ</a>@<a href="http://192.168.0.10:5678">192.168.0.10:5678</a>>,30,tTog)<br>
><br> > both destinations ring at the same time and one that is answered<br> > starts conversations. but audio is one sided as i mentioned above.<br> ><br> > But simply dialing single registration of XYZ like<br>
> Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends.<br> ><br> > have any idea what is going wrong??<br> ><br> > any help will be highly appreciated<br> ><br> > regards,<br>
><br> > Nasir Javaid<br> ><br> ><br> ><br> ><br> -------------- next part --------------<br> An HTML attachment was scrubbed...<br> URL: <a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/b51fed83/attachment-0001.htm">http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/b51fed83/attachment-0001.htm</a><br>
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