[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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