[cisco-voip] Weird MTP issues

[Kenneth Hayes]

If you like we can setup a webex.

Sent from my iPhone

On Aug 12, 2013, at 12:43 PM, Peter Slow <peter.slow at gmail.com> wrote:

> sounds interesting,  I could take a look =)
>
> take the MTP back out and get traces of the call failing the way it
> was originally... Also, "debug ccsip mess" from the CUBE. ,,,And make
> sure the CCM traces are set to detailed and that the options in both
> columns are all enabled except the two labeled "SIP register'
> somethingorother and skinny keepalive, please =)
>
> -Pete
>
> On Sun, Aug 11, 2013 at 8:47 PM, Kenneth Hayes <kennethwhayes at gmail.com> wrote:
>> All,
>>
>>
>>
>> Currently I'm running CUBE in my enviroment and we have a weird issue.
>> Here's my Call Flow first.
>>
>>
>>
>> PSTN->CUBE->CUCMBE->AA
>>
>>
>>
>> So when callers come in they will reach the Auto Attendant and have
>> the option to press 1 for dial-by-name, well when they select option 1
>> they get to the Directory Handler and speak the person they are trying
>> to reach, well when the call begins to transfer it rings the extension
>> for a slight second then it drops. Weird right, well on the SIP trunk
>> between CUCM and CUBE I checked MTP required and saved changes and
>> reset the trunk and attempt the call it works fine, the phone rings
>> normal, and no issues occur. So in my SIP profile that's associated
>> with the SIP trunk I checked the box to insert MTP if needed and
>> unchecked the MTP required on the SIP trunk and the call drops at a
>> slight second.
>>
>>
>>
>> In the CUBE I've added a  voice class sip profile to see if that
>> corrects it, and it has not. Can someone assist me with this issue?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip