[asterisk-users] One way audio when dialing multiple registrations
Nasir Javaid
nasirjavaidnasir at gmail.com
Wed Jul 21 11:00:03 CDT 2010
Hi again
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,
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
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,
Dial(SIP/XYZ at xxxxxxxxxxxx:5062,30,rtT)
during conversation audio was one way just like before (calling party can
hear called party but called party can not hear calling).
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.
*NORMAL DIAL *
INVITE sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0
To: <sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588>
Contact: <sip:1334225544 at xxxxxxxxxxx:5060>
*IP DIAL*
INVITE sip:XYZ at xxxxxxxxxxx:28614 SIP/2.0
To: <sip:XYZ at xxxxxxxxxxxx:28614>
Contact: <sip:1334225544 at xxxxxxxxxxx:5060>
hope this research will ease a bit the quest to find a solution. now
question is
1) how can we assign this parameter when dialing via IP:PORT?
2) what else options do we have if we want to dial using IP:PORT mechanism.
waiting for your kind resopnse.
Nasir Javaid.
-------------------------------------------------------------------------------------------------------------------------------------------------------
-------------------------------------------------------------------------------------------------------------------------------------------------------
sorry for the typo mistake. the actual dial string that I used is like this
Dial(SIP/XYZ at xxxxxxxxxxxx:5062-096afee8,30,rtT)
Dial(SIP/XYZ at xxxxxxxxxxxx:64290-0966ab80,30,rtT)
it is not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)
it was just a typing mistake that may have diverted all of you. hope this
clears what i am trying to do.
regards,
Nasir Javaid
-----------------------------------------------------------------------------------------------------------------------------------------------
I am sure you can't achieve what you are trying to achieve here. Simply use
two different extensions instead of one.
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.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-19 12:28 PM, "Nasir Javaid" <nasirjavaidnasir at xxxxxxxxx> wrote:
thanks a lot zishan and philipp,
probably that is the problem that is occurring. I am gonna take some
wireshark or etherial trace to further investigate the problem.
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.
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
for example
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)
^
|
|________
|
but problem is that asterisk assigns random unique-id for every call. and
also it is available only when dialing...
what are my options?
your help will be highly appreciated.
regards,
Naisr Javaid
-----------------------------------------------------------------------------------------------------------------------------------------------------------------
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
I have had this problem once and I never tried registering same extension
from two different places after that.
Try Phillip's suggestion, maybe it'll work for you.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-15 11:42 AM, "Philipp von Klitzing" <
klitzing at xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx> wrote:
Hi!
> I am working on calling 2 registrations of same user on 2 different ip or
> ports. It works f...
You need to make sure that these two phones use *different* RTP ports,
and that this is handled correctly in your router/NAT device (by port
forwarding or other methods).
Philipp
-----------------------------------------------------------------------------------------------------------------------
Hi Zeeshan,
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.
==============================
==============================================================
Hi,
I am working on calling 2 registrations of same user on 2 different ip
or ports. It works fine and both phones ring simultaneously. the
problem is that there is one way audio, calling party can hear me but i
can't hear calling party.
here is the scenario..
SIP/XYZ at xxxxxxxxxxx:5060
SIP/XYZ at xxxxxxxxxxxx:5678
i dial using following dial string
Dial( SIP/XYZ at xxxxxxxxxxx:5060& SIP/XYZ at xxxxxxxxxxxx:5678,30,tTog)
both destinations ring at the same time and one that is answered starts
conversations. but audio is one sided as i mentioned above.
But simply dialing single registration of XYZ like
Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends.
have any idea what is going wrong??
any help will be highly appreciated
regards,
Nasir Javaid
======================================================================================
thanks in advance ...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/663ed559/attachment-0001.htm
More information about the asterisk-users
mailing list