Hi again<br><br>today when i was doing my research on this issue i found that even dialing a sip user by it's IP also raises this problem. here is what i did,<br><br>First I dialed my registered user in normal way like this,<br>
<br>Dial(SIP/XYZ,30,rtT)<br><br>and during conversation audio was OK in both ways. Then I dialed the registered user via it's ip and port to which it was registered. like this,<br><br>Dial(SIP/XYZ@xxxxxxxxxxxx:5062,30,rtT)<br>
<br>during conversation audio was one way just like before (calling party can hear called party but called party can not hear calling).<br><br>after taking debug trace of both methods what I found was that a SIP HEADER parameter "rinstance" was missing in "to" and "INVITEt" header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk.<br>
<br><u>NORMAL DIAL </u><br>INVITE sip:XYZ@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0<br>To: <sip:XYZ@xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588><br>Contact: <sip:1334225544@xxxxxxxxxxx:5060><br><br>
<u>IP DIAL</u><br>INVITE sip:XYZ@xxxxxxxxxxx:28614 SIP/2.0<br>To: <sip:XYZ@xxxxxxxxxxxx:28614><br>
Contact: <sip:1334225544@xxxxxxxxxxx:5060><br>
<br>hope this research will ease a bit the quest to find a solution. now question is<br><br>1) how can we assign this parameter when dialing via IP:PORT?<br>2) what else options do we have if we want to dial using IP:PORT mechanism.<br>
<br> waiting for your kind resopnse.<br><br>Nasir Javaid.<br><br><br>-------------------------------------------------------------------------------------------------------------------------------------------------------<br>
-------------------------------------------------------------------------------------------------------------------------------------------------------<br><br>sorry for the typo mistake. the actual dial string that I used is like this<br>
<br>Dial(SIP/XYZ@xxxxxxxxxxxx:5062-096afee8,30,rtT)<br>Dial(SIP/XYZ@xxxxxxxxxxxx:64290-0966ab80,30,rtT)<br><br><br>it is not <br><br>Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)<br>
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)<br><br>it was just a typing mistake that may have diverted all of you. hope this clears what i am trying to do.<br><br>regards,<br><br>Nasir Javaid<br><br><br>-----------------------------------------------------------------------------------------------------------------------------------------------<br>
<br> I am sure you can't achieve what you are trying to achieve here. Simply use two different extensions instead of one.
<p>Considering how SIP communication works, I believe SIP doesn't allow
multiple registrations like this. Maybe somebody can correct me here if
I am wrong.<br></p>
<p>Zeeshan A Zakaria</p>
<p>--<br>
<a rel="nofollow" href="http://www.ilovetovoip.com/">www.ilovetovoip.com</a></p>
On 2010-07-19 12:28 PM, "Nasir Javaid" <<a rel="nofollow" href="mailto:nasirjavaidnasir@xxxxxxxxx">nasirjavaidnasir@xxxxxxxxx</a>> wrote:<br><br><div><p>thanks a lot zishan and philipp,</p><p>
probably
that is the problem that is occurring. I am gonna take some wireshark
or etherial trace to further investigate the problem. <br></p>i
don't wanna stuck into port forwarding issue as it will waste lot of
time 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
id and dial it directly like simple Dial(SIP/xyz ) dialing<br>
<br>for example <br><br>Dial(SIP/192.168.0.20:5062-096afee8,30,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 also it is available only when dialing...<br>
what are my options?<br><p>your help will be highly appreciated.</p><p>regards,</p><p><br></p><p>Naisr Javaid<br></p><p>-----------------------------------------------------------------------------------------------------------------------------------------------------------------<br>
</p><p>Based on the info you provided (though wireshark analysis will tell
more about it), I am sure what is happening is that rtp coming back
from the called doesn't know which ip to go to, because asterisk knows
two ip addressses for the same extension due to the way you dialed it,
i.e. in ringgroup fashion</p>
<p>I have had this problem once and I never tried registering same extension from two different places after that.</p>
<p>Try Phillip's suggestion, maybe it'll work for you.</p>
<p>Zeeshan A Zakaria</p>
<p>--<br>
<a rel="nofollow" href="http://www.ilovetovoip.com/" target="_blank">www.ilovetovoip.com</a></p>
On 2010-07-15 11:42 AM, "Philipp von Klitzing" <<a rel="nofollow" href="mailto:klitzing@xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx" target="_blank">klitzing@xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx</a>> wrote:<br><br>
<p><font color="#500050">Hi!<br>
<br>> I am working on calling 2 registrations of same user on 2 different ip or<br>> ports. It works f...</font></p>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 not get any satisfying answer. I wish if you
could help me out. below is my post.<br><br>==============================<div>==============================================================<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 rel="nofollow" href="http://XYZ@xxxxxxxxxxx:5060/" target="_blank">XYZ@xxxxxxxxxxx:5060</a><br>
<br>
SIP/<a rel="nofollow" href="http://XYZ@xxxxxxxxxxxx:5678/" target="_blank">XYZ@xxxxxxxxxxxx:5678</a><br>
<br>
i dial using following dial string<br>
<br>
Dial(
SIP/XYZ@xxxxxxxxxxx:5060&
SIP/<a rel="nofollow" href="http://XYZ@xxxxxxxxxxxx:5678/" target="_blank">XYZ@xxxxxxxxxxxx: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 ...</div>
</div>