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

[Jonathan Charles]

While it is ringing on the IPIPGW, I do see 2 SIP call legs... for some
reason no audio is getting cut thru.... it rings, I answer it and it drops,
from either direction...

On Wed, Aug 6, 2008 at 12:52 PM, Jonathan Charles <jonvoip at gmail.com> wrote:

> 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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