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

ROZA, Ariel Ariel.ROZA at LA.LOGICALIS.COM
Wed Aug 6 14:09:55 EDT 2008


Jonathan,
 
    I thinkyou need the command
 
    redirect ip2ip under voice service voip (Global) or under each Dial¨-Peer (specific)
 
Regards,
 
    Ariel 

________________________________

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 02:56 p.m.
To: Nguyen Le
Cc: OSL CCIE Voice Lab Exam; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] [OSL | CCIE_Voice] IPIPGW Sip to Sip


Call is from x1008 (on CCM) to x3003 (on CCME)

On the IPIPGW:


voice service voip 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
 modem passthrough nse codec g711ulaw
 sip

dial-peer voice 500 voip
 destination-pattern 1008
 session protocol sipv2
 session target ipv4:10.0.0.124
 dtmf-relay sip-notify rtp-nte
 codec g711ulaw
!
dial-peer voice 501 voip
 destination-pattern 3...
 session target ipv4:10.0.0.131
 incoming called-number 3003
 codec g711ulaw
!

On CCME:


dial-peer voice 201 voip
 destination-pattern 1008
 session protocol sipv2
 session target ipv4:10.0.0.63
 incoming called-number 3003
 dtmf-relay rtp-nte
 codec g711ulaw




Jonathan


On Wed, Aug 6, 2008 at 12:54 PM, Nguyen Le <nguyenbot at gmail.com> wrote:


	You also have under

	Voice service voip

	Allow connections sip to sip ?

	 

	Also, just double check and make sure your SIP Trunk is in a region that is set to G711 to all other sites

	 

	Nguyen

	 

	 

	From: Jonathan Charles [mailto:jonvoip at gmail.com] 
	Sent: Wednesday, August 06, 2008 12:52 PM
	To: Nguyen Le
	Cc: OSL CCIE Voice Lab Exam; cisco-voip at puck.nether.net
	Subject: Re: [OSL | CCIE_Voice] IPIPGW Sip to Sip

	 

	Yeah, all dial-peers have the codec hard set to 711ulaw
	
	
	Jonathan

	On Wed, Aug 6, 2008 at 12:43 PM, Nguyen Le <nguyenbot at gmail.com> wrote:

	Jonathan - 

	 

	Make sure your call codec is g711

	 

	Nguyen

	 

	 

	From: ccie_voice-bounces at onlinestudylist.com [mailto:ccie_voice-bounces at onlinestudylist.com] On Behalf Of Jonathan Charles
	Sent: Wednesday, August 06, 2008 12:41 PM
	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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