The problem is not with cisco as the SIP header on debug doesn't contain the 
called number. It only says To:sip:ip add of cisco gw. It should say number:ip 
addr of cisco gw.

Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria,  +234-702-8000-709 
email:[email protected]
www.starcomms.com


-----Original Message-----
From: [email protected] 
[mailto:[email protected]] On Behalf Of David Backeberg
Sent: Monday, March 15, 2010 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways

On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena <[email protected]> wrote:
> I have been trying to do this since 2 days but couldn't make it....need your 
> help..

Well, you could certainly ask Cisco for help.
You did pay Cisco money, right?

> PSTN-----Cisco AS5350-------Asterisk Box--------VoIP Providers

> I am able to place call from cisco gateway to the asterisk box and also to 
> some softphones extensions but >when making a call from softphone from 
> asterisk box to PSTN, it fails. While I debug on Cisco gateway I found >that 
> the To field is SIP header is coming as sip:41.205.190.15 which is not 
> correct, instead it should be dialed >number:41.205.190.15

Then the problem seems to be between your asterisk box and your Cisco.
Perhaps if you told us what you were trying to SIP dial, we would be
able to tell us what you did wrong.

> Has any one of you tried using Asterisk in this scenario

yes.

> and also to do LCR and Quality based routing of International calls?

I don't know what that means.

> Please let me know if there is any documentation /example of this kind 
> available

There is.
cisco.com
you pay them, then you can use their documentation.

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