[Asterisk-Dev] Re: [Asterisk-Users] Asterisk not working with session border controller

Olle E. Johansson oej at edvina.net
Tue Jan 6 09:14:21 MST 2004

Venkat Venkataraju wrote:
> Steve Totaro wrote:
>> And you say this is a commercial product that you purchased and they 
>> asked
>> you to fix the problem?
> guys @ middle claim that the Asterisk is breaking the RFC and they are 
> asking me to fix it on Asterisk as our company wants to use Asterisk. 
> The "middle" does seems to work with many other SIP proxies.
>> Where did you purchase it from or are you developing it and need 
>> help?  More
>> docs might help.
> I'm not developing it, but i'm working with a company thats trying to 
> bring consumer VoIP by using Asteisk. I'm working on call processing 
> system that uses astman. But i was pulled into this to get asterisk work 
> with Middle. I dont have any other docs. i've asked for it. i'll post it 
>  if i get one.
> I've got logs from Middle, but i'm not sure how it may help. for the 
> request
> REGISTER sip:sip-xxxxx.homeip.net SIP/2.0
> Via: SIP/2.0/UDP 68.#.#.84:7062;branch=z9hG4bK-middle-4178
> Via: SIP/2.0/UDP 68.#.#.125:5060
> From: <sip:2002 at sip-xxxxx.homeip.net;user=phone>;tag=1722079273
> To: <sip:2002 at sip-xxxxx.homeip.net;user=phone>
> Contact: <sip:2002*sip-xxxxx.homeip.net= at 68.#.#.84:7062>
> Call-ID: 2365955374 at 68.#.#.125
> Content-Length: 0
> User-Agent: Cisco ATA 186  v2.16 ata18x (030401a)
> Asteisk is supposed to respond back to the IP on port 7062 (i was told. 
> i'vnt read the RFC), but it send the request back on the port 7060, the 
> originating port of the request.
Please add a SIP DEBUG trace of a registration, so I see how Asterisk
responds. That's a weird Contact: header...


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