[asterisk-users] asterisk-users Digest, Vol 72, Issue 49

Tue Jul 20 10:46:10 CDT 2010

sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows

SIP/XYZ at 119.68.0.90:5060
SIP/XYZ at 202.16.34.10:5678

so dial command with unique-id i want to use will be

Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT)

and not

Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)

now what you say is it still impossible?

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

user name: XYZ

extension associated with XYZ:  22129889035

registered at:        SIP/XYZ at 192.168.0.20:5062        and         SIP/XYZ@
192.168.0.12:64290

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.

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.

so that i can set rtp ports with the channels involved in conversation at
runtime.
sincere regards,
Nasir Javaid

-------------------------------------------------------------------------------------------------------------------

Message: 2
Date: Mon, 19 Jul 2010 13:41:32 -0400
From: Zeeshan Zakaria <zishanov at gmail.com>
Subject: Re: [asterisk-users] One way audio when dialing multiple
       registrations
To: Asterisk Users Mailing List - Non-Commercial Discussion
       <asterisk-users at lists.digium.com>
Message-ID:
       <AANLkTikOU9Eec7pFL2CsXi83oYG0hnHZn34jECNqurYl at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

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 gmail.com> 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 119.68.0.90:5060

SIP/XYZ at 202.16.34.10:5678

i dial using following dial string

Dial( SIP/XYZ at 119.68.0.90:5060& SIP/XYZ at 202.16.34.10:5678,30,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 ...
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