Hi Tony,

Thanks for the reply. Its sipx version 4.2.1-018971. Asterisk is connected
to pstn via e1 isdn pri and mfcr2. I could not remove asterisk since sipxec
doesn't support pstn connection unless I will use different gateway such as
audiocodes.

Call is originating from the pstn passing thru the asterisk and into one of
the extension in sipxecs and the one who will initiate the callpickup is
also another extension in sipxecs.

Any suggestion?

Regards,

Mac

On Mon, Feb 7, 2011 at 6:27 PM, Tony Graziano
<[email protected]>wrote:

> what version of sipx? what kind of pstn connection is it? it seems it would
> be far simpler to remove asterisk (IMO). two different systems to do "one"
> thing is really making it more complex than it needs to be.
>
> On Mon, Feb 7, 2011 at 4:59 AM, Leonimar Cape <[email protected]>wrote:
>
>> Hi Guys,
>>
>> Hope you can shed me some light on whether what I am trying to accomplish
>> is possible and what is the right approach or what I am doing wrong in my
>> current setup.
>>
>> Currently I have setup an Asterisk which acts as a media gateway and
>> handle all pstn calls to voip and a Sipxecs in distributed mode where all
>> the extensions/locals are registered. Asterisk and Sipxecs are connected
>> using the unmanaged gateway configuration. Everything works fine until I was
>> asked to make the call pickup works.
>>
>> I have checked that yealink phones do support the rfc4235 features which
>> needed to make this work. Asterisk 1.6.2 states that it is also supported.
>> But I can't make it work.
>> tshark trace shows that the SUBSCRIBE message is being send to the primary
>> and distributed sipxecs but it seems that I can't find the asterisk
>> connection and returning back a "404 not found" response.
>>
>> Any feedback would be much appreciated.
>>
>> Regards,
>>
>> Mac
>>
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>>
>
>
>
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