asterisk users - Dec 2006 - Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?

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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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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>
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