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

[Jonathan Charles]

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 –
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>
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> Make sure your call codec is g711
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>
> Nguyen
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>
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> *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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