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

[Jonathan Charles]

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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