03-03-2010 02:32 PM - edited 03-15-2019 09:37 PM
We have some troubles with call through a SIP trunk.
There is a SIP trunk beetween a CUCM and an external (public network) SIP endpoint.
Phone calls can be established with success but the SIP calls are established between the phone device and the external SIP endpoint.
We want to force the call go through the CUCM.
Is that possible?
Many Thanks
03-03-2010 04:14 PM
Hi
Typically you would use an IP-IP gateway for this sort of thing.
You can do something similar by forcing the trunk to use an MTP (a software MTP runs on the CallManagers as part of the Voice Media Streaming service). The RTP will then be proxied via a CCM server...
To do this, check the 'Media Termination Point Required' on the trunk config page.
Bear in mind the software MTP supports G711 only.
Regards
Aaron
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03-07-2010 10:07 AM
Hi Aaron,
thanks for reply.
I have already try to use the "require MTP" as option on the trunk configuration, but when checked the call cannot be established.
Now you told me that the MTP supports only G711 codec so that is the problem: the SIP trunk is based on a G729a and phone devices use G722 (as default).
There is a way to overcome this "restriction"?
Many thanks
Angelo
03-08-2010 01:09 AM
Hi
In that case you probably want to look into using something like CUBE - http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml,
Ordering info:
Regards
Aaron
03-08-2010 02:04 PM
Unfortunately,our budget do not let us to buy a CUBE device/license.
Did you think we can do the same with the 2800 router used to connect the ISDN lines?
Or something else.
Any other idea is appreciated.
Many Thanks
A.
03-22-2010 03:41 AM
You could use a trixbox between the CUCM and SIP
SP...
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