[cisco-voip] SIP2SIP2PROXY

bilay ozer bilayo at a-c-tgmbh.de
Sun Mar 18 18:17:58 EST 2007


Hi Cumhur ,

 

Thank you for your reply ,

 

IP to IP not supported OS for As5400 ( XM supported  ) also I have tried on
Soft-switchs ( MERA / VMSM / ) applying user information from the anonymous
ingress calls than translations.

Soft-switchs binding their Localhost.localdomain information to at the end
of the packet its shown like this at the from side of contact. I found your
sended documentation on the forum I read than I posted my details.

 

Invite packet

                From : uri sip:callingline at calleripaddress

                            Contach uri  : user at calleripaddress   -- > it
should be Username at calledDomain or outbound proxy actually its not irregular
format all soft phones  working like this if you need to send user name for
requested uri and also if you are directly sending user name to target proxy
it s automatically binding Proxy ip address as a calledDomain

 

 

May be you can suggest any idea after those explanations

 

Regards

 

Bilay

 

 

From: Cumhur Kizilari [mailto:cumhur at gmail.com] 
Sent: Sunday, March 18, 2007 12:45 PM
To: bilay ozer; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SIP2SIP2PROXY

 

Hi,
s 
Are you using IP to IP IOS on your router ? if not you should (in my
opinion) use IP to IP IOS for call forwarding over IP.

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_data_sheet091
86a00801da698.html

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration
_guide_chapter09186a00806b0a14.html

Regards,
Cumhur



On 3/17/07, bilay ozer <bilayo at a-c-tgmbh.de> wrote:

Hi There,


I am trying to apply my scenario as a such ways. I tried too many options
with SS but there is no capability for that

My customer has a SIP-gw they will originate their calls to directly MY
voice GW . My voice-gw will receive this call than will forward to SIP
proxy.

Sip-as5400-sip with uri outboundproxy

Here is the conditions.

Proxy accepting calls with Requested URI mean ( account information user @ 
domainofproxy) I have 10 sip account on my hand is there any way to set that
account under the voip dial-peers with destination SIP proxy also accepting
calls with anonymous @ calleripaddress means with translating with source 

I need a configuration example for that task there are no PSTN or trunk
connection on As5400.

Regards





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