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

[Jonathan Charles]

Nope... same behavior...


Jonathan

On Wed, Aug 6, 2008 at 1:09 PM, ROZA, Ariel <Ariel.ROZA at la.logicalis.com>wrote:

>  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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