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

Wed Aug 6 15:54:01 EDT 2008

Cause code 16 is normal call clearing.  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, August 06, 2008 3:48 PM
To: Chris Ward
Cc: OSL CCIE Voice Lab Exam; cisco voip; Stephen Collinson
Subject: Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip

CUCM is CCM 4.1.3(sr7)
Yes, the same behavior is exhibited in both directions.
G711ulaw all the way thru... if I change it to SIP to H323, it works
fine in both directions. No transcoders in the network.
Region is 711
The MTP required check box is set on the SIP trunk on CCM.

Jonathan

On Wed, Aug 6, 2008 at 2:45 PM, Chris Ward <chrward at cisco.com> wrote:

If the call works until the phone is answered, then it is probably
media/capability related.

First, is this the call flow?

IP Phone --- CUCM --- SIP Trunk --- IPIPGW --- SIP --- CUCME --- IP
Phone

Now these questions:

1.	What is the version of CUCM? 
2.	Do calls fail in both directions? 
3.	What is the region setting (what codec) between the CUCM IP
Phone and the SIP trunk on CUCM? 
4.	Is the CUCM SIP trunk doing early media (is the "MTP required"
checkbox checked)? 

-Chris

________________________________

From: Jonathan Charles <jonvoip at gmail.com>
Date: Wed, 6 Aug 2008 14:32:31 -0500
To: Stephen Collinson <scollinson at capewave.co.uk>
Cc: OSL CCIE Voice Lab Exam <ccie_voice at onlinestudylist.com>, cisco voip
<cisco-voip at puck.nether.net>
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

________________________________

	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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