[cisco-voip] Fwd: CUCM 6.1 and SIP Trunking

[David Martin]

thanks, I don't need 729, 711 is fine. going to configure an external MTP
anyway just to rule it out.

David


On 26 July 2011 01:05, Nicholas Samios <nsamios at staff.iinet.net.au> wrote:

> The software MTP on UCM only supports 711, if you need to use 729 you’ll
> need to configure an external MTP/transcoder or CUBE.****
>
> ** **
>
> Using a CUBE in the middle as a demarc is best practice, gives you all the
> features of SIP & SDP manipulation (e.g delayed to early, message
> normalization), transcoding, etc and also avoids a direct SIP trunk to your
> UCM.****
>
> ** **
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Cristobal Priego
> *Sent:* Tuesday, July 26, 2011 1:28 AM
> *To:* David Martin
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] CUCM 6.1 and SIP Trunking****
>
> ** **
>
> on version 6.X if you want to send a early offer on your sip trunk you need
> MTP and the codec supported is g.711 only. if you're trying to use g.729 is
> not going to work ****
>
> 2011/7/25 David Martin <adavidm at gmail.com>****
>
> All,****
>
> ** **
>
> I am trying to set up a SIP trunk between CUCM 6.1 and Freeswitch. I only
> need calls outbound from Cisco to Freeswitch at the moment but am having
> some difficulties.****
>
> ** **
>
> The Cisco side never sends an SDP, either in the INVITE or the ACK, and
> Freeswitch needs this to operate. I have tried using "require media
> termination point" and "disable early media" options but there appears to be
> no difference, which is strange. I include a pastebin of the conversation:
> ****
>
> ** **
>
> http://pastebin.com/T3eqms8N****
>
> ** **
>
> and a couple of screenshots of the CUCM config for the trunk.****
>
> ** **
>
> http://img200.imageshack.us/img200/9405/siptrunk.png****
>
> ** **
>
> http://img191.imageshack.us/img191/3863/sipprofile.png****
>
> ** **
>
> Any thoughts on where to start looking? I was wondering if CUCM will
> continue to attempt the call if there is a problem with the MTP( I am using
> software MTP on CUCM at the moment). If this is the case then I can see why
> the SDP does not appear in the INVITE (assuming the MTP is broken), but not
> why it does not appear in the ACK.****
>
> ** **
>
> Thanks in advance.****
>
> ** **
>
> David****
>
>
> _______________________________________________
> cisco-voip mailing list
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> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
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