[Asterisk-Users] SIP Express Router & Asterisk

[tan at yointernet.com]

Hi,

We will shortly launch a sip service. Architecture is:

SER: for SIP registration and IP call routing, incoming number
termination, STUN, Nat traversal etc.
Asterisk: outgoing call routing, calling card platform, billing,
extended facilities e.g. voicemail etc.

Works well.

Tan
Telappliant.com
Voiptalk.org


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Anton
Tinchev
Sent: 27 November 2003 13:19
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP Express Router & Asterisk


There was some issues with Audiocodes MP10x - With both Asterisk and
SER. It was fixed in last firmwire release. Hope it is fixed in Mediant
too. It was general SIP issues...

Ryan Tucker wrote:

> Greetings...
> 
> We've been having some interoperability issues between Asterisk and an
> AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 
> somewhere.  So, I've been pondering using iptel.org's SIP server (SIP 
> Express Router) as a "front end" for PSTN calls going out to the
Mediant, 
> while using Asterisk for everything else.
> 
> Has anyone done something similar, or anything at all involving SER 
> and
> Asterisk?
> 
> Thanks!  -rt
> 


_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users