[cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip

[ROZA, Ariel]

What´s the disconnect cause code shown in a debug voip ccapi inout?


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From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Miércoles, 06 de Agosto de 2008 04:33 p.m.
To: Stephen Collinson
Cc: OSL CCIE Voice Lab Exam; cisco voip
Subject: Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip


Right, the question is, how do you configure it correctly?

What would cuz the audio to not cut thru and the call to drop... I was suspecting codec, but it is G711 all the way thru (hard coded on each dial peer)



Jonathan


On Wed, Aug 6, 2008 at 2:21 PM, Stephen Collinson <scollinson at capewave.co.uk> wrote:


	SIP to SIP should work fine, when configured correctly.

	 

	I was just trying to give you a scenario where we may need to use it. Apologies if this was not helpful

	 

	 

	 

	
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	From: Jonathan Charles [mailto:jonvoip at gmail.com] 
	Sent: 06 August 2008 19:55
	To: Stephen Collinson
	Cc: cisco voip; OSL CCIE Voice Lab Exam
	Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip

	 

	Perhaps I wasn't clear...
	
	
	There is no CUE.
	
	This is a SCCP phone on a CCME, and a SCCP phone on CCM with a SIP trunk to an IPIPGW, and a SIP dial-peer to CCME...
	
	
	Jonathan

	On Wed, Aug 6, 2008 at 1:52 PM, Stephen Collinson <scollinson at capewave.co.uk> wrote:

	Perhaps worth looking at your config.

	 

	You will need sip to sip, say to access CUE VM from a CCM SIP trunk.

	 

	Check all G711 etc.

	 

	Debug CCSIP

	 

	 

	 

	
________________________________


	From: ccie_voice-bounces at onlinestudylist.com [mailto:ccie_voice-bounces at onlinestudylist.com] On Behalf Of Jonathan Charles
	Sent: 06 August 2008 18:41

	
	To: OSL CCIE Voice Lab Exam; cisco-voip at puck.nether.net
	Subject: [OSL | CCIE_Voice] IPIPGW Sip to Sip

	 

	So, I was playing with an IPIPGW

	
	
	CCM on one side (SIP trunk) and CCME on the other (SIP dial-peer)... call worked, but as soon as you answered it dropped.
	
	I changed the SIP dial-peer from the IPIPGW to H.323 (no session protocol) and RTP cuts thru fine...
	
	Am I misreading something, is SIP to SIP not supported, or is my config retarded?
	
	
	
	Jonathan

	 


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