[asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?

[Pavel Jezek]

I think, callmanager needs media termination point (mtp) for sip trunk, 
so rtp stream will always go through callmanager...


JR Richardson wrote:
> Hi All,
>
>  
>
> I haven't started sip traces or debug yet, but was wondering what the deal
> is with the CCM and reinvite, why it doesn't work with Asterisk (using
> 1.2.9.1).  I can make calls back and forth all day with canreinvite=no, when
> I try to reinvite an inbound sip call from the CCM with Asterisk server 1 to
> Asterisk Server 2, I get one-way audio issues.  All the RTP ports are
> configured the same.  I remember Cisco Phones and ATA's have some reinvite
> issues, wondering if the same applies to the CCM and how it handles
> reinvites?
>
>  
>
> In the wiki example of integrating CCM with Asterisk, the SIP context shows
> canreinvite=yes, so should this be working ok, maybe I'm doing something
> wrong?
>
>  
>
> Thanks
>
>  
>
> JR
>
>  
>
> JR Richardson
>
> Engineering for the Masses
>
>  
>
>
>   
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