sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows<br><br><br>
SIP/<a href="http://XYZ@119.68.0.90:5060/" target="_blank">XYZ@119.68.0.90:5060</a><br>
SIP/<a href="http://XYZ@202.16.34.10:5678/" target="_blank">XYZ@202.16.34.10:5678</a><br>
<br>so dial command with unique-id i want to use will be<br><br>Dial(SIP/XYZ@192.168.0.20:5062-096afee8,30,rtT)<br><div id=":10m" class="ii gt">
Dial(SIP/XYZ@192.168.0.12:64290-0966ab80,30,rtT)<br>
</div><br>and not <br><br>Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)<br><div id=":10m" class="ii gt">
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)<br>
</div><br><br>now what you say is it still impossible?<br><br><p>also can you explain what you mean by extension. username that i am using
(XYZ) is registered at both ips and we dial 22129889035 which is
associated with XYZ. like below<br></p><p>user name: XYZ</p>
<p>extension associated with XYZ: 22129889035<br></p><p>registered at: SIP/XYZ@<a href="http://192.168.0.20:5062/" target="_blank">192.168.0.20:5062</a> and SIP/XYZ@<a href="http://192.168.0.12:64290/" target="_blank">192.168.0.12:64290</a><br>
</p><p>do you mean i use 2 different usernames or assign 2 different
extensions to same user? as we already have 2-5 extension numbers
associated with each username other than 22129889035.<br></p><p>i attached sip debug trace but it was too heavy to be posted. if you say i can try posting it. is there any way for setting rtp port in dialplan. using functions like sipAddheader etc.</p>
<p>so that i can set rtp ports with the channels involved in conversation at runtime.</p>sincere regards,<br>Nasir Javaid<br><br>-------------------------------------------------------------------------------------------------------------------<br>
<br>
Message: 2<br>
Date: Mon, 19 Jul 2010 13:41:32 -0400<br>
From: Zeeshan Zakaria <<a href="mailto:zishanov@gmail.com">zishanov@gmail.com</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:<br>
<<a href="mailto:AANLkTikOU9Eec7pFL2CsXi83oYG0hnHZn34jECNqurYl@mail.gmail.com">AANLkTikOU9Eec7pFL2CsXi83oYG0hnHZn34jECNqurYl@mail.gmail.com</a>><br>
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<br>
I am sure you can't achieve what you are trying to achieve here. Simply use<br>
two different extensions instead of one.<br>
<br>
Considering how SIP communication works, I believe SIP doesn't allow<br>
multiple registrations like this. Maybe somebody can correct me here if I am<br>
wrong.<br>
<br>
Zeeshan A Zakaria<br>
<br>
--<br>
<a href="http://www.ilovetovoip.com/" target="_blank">www.ilovetovoip.com</a><br>
<br>
On 2010-07-19 12:28 PM, "Nasir Javaid" <<a href="mailto:nasirjavaidnasir@gmail.com">nasirjavaidnasir@gmail.com</a>> wrote:<br>
<br>
thanks a lot zishan and philipp,<br>
<br>
probably that is the problem that is occurring. I am gonna take some<br>
wireshark or etherial trace to further investigate the problem.<br>
i don't wanna stuck into port forwarding issue as it will waste lot of time<br>
and also normal calling is working on my current port forwarding.<br>
<br>
what i am currently trying to grab the channel name along with it's unique<br>
id and dial it directly like simple Dial(SIP/xyz ) dialing<br>
<br>
for example<br>
<br>
Dial(SIP/192.168.0.20:5062-<div id=":10m" class="ii gt">096afee8,30,rtT)<br>
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)<br>
^<br>
|<br>
|________<br>
|<br>
but problem is that asterisk assigns random unique-id for every call. and<br>
also it is available only when dialing...<br>
what are my options?<br>
<br>
your help will be highly appreciated.<br>
<br>
regards,<br>
<br>
<br>
Naisr Javaid<br>
<br>
-----------------------------------------------------------------------------------------------------------------------------------------------------------------<br>
<br>
Based on the info you provided (though wireshark analysis will tell more<br>
about it), I am sure what is happening is that rtp coming back from the<br>
called doesn't know which ip to go to, because asterisk knows two ip<br>
addressses for the same extension due to the way you dialed it, i.e. in<br>
ringgroup fashion<br>
<br>
I have had this problem once and I never tried registering same extension<br>
from two different places after that.<br>
<br>
Try Phillip's suggestion, maybe it'll work for you.<br>
<br>
Zeeshan A Zakaria<br>
<br>
--<br>
<a href="http://www.ilovetovoip.com/" target="_blank">www.ilovetovoip.com</a><br>
On 2010-07-15 11:42 AM, "Philipp von Klitzing" <<br>
klitzing@xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx> wrote:<br>
<br>
Hi!<br>
<br>
> I am working on calling 2 registrations of same user on 2 different ip or<br>
> ports. It works f...<br>
You need to make sure that these two phones use *different* RTP ports,<br>
and that this is handled correctly in your router/NAT device (by port<br>
forwarding or other methods).<br>
<br>
Philipp<br>
<br>
-----------------------------------------------------------------------------------------------------------------------<br>
Hi Zeeshan,<br>
<br>
I saw many of your posts on forum. i also put my problem on forum but did<br>
not get any satisfying answer. I wish if you could help me out. below is my<br>
post.<br>
<br>
==============================<br>
==============================================================<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 i<br>
can't hear calling party.<br>
<br>
here is the scenario..<br>
<br>
SIP/<a href="http://XYZ@119.68.0.90:5060/" target="_blank">XYZ@119.68.0.90:5060</a><br>
<br>
SIP/<a href="http://XYZ@202.16.34.10:5678/" target="_blank">XYZ@202.16.34.10:5678</a><br>
<br>
i dial using following dial string<br>
<br>
Dial( SIP/XYZ@119.68.0.90:5060& SIP/<a href="http://XYZ@202.16.34.10:5678/" target="_blank">XYZ@202.16.34.10:5678</a>,30,tTog)<br>
<br>
both destinations ring at the same time and one that is answered starts<br>
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>
thanks in advance ...<br>
</div>