Alcatel Unleashed

cavagnaro wrote:And it would be much better if you avoid to duplucate your posts.

Looks like Chrome does not like phpBB...

Mea culpa! Mea maxima culpa! Last time I got an error and I was mindless now thinking it was not showed up. But it is...

The calltransfer should be done in the simplest case without any dialplan action. As soon a call bridge is initiated, the channel should do it on its own.
But queue applications i.e. in the middle of the task may prevent a native bridge, then you will need an diaplan command, which must be implemented in the asterisk channel.

Mario

Have you testet chan_capi or is it already in stable version of libpri and chan_zap.c / chan_dahdi.c in asterisk?

I haven't found any docs showing how to setup a QSIG PRI between Alcatel and Asterisk.

If someone is "happily" running such a setup I would greatly appeciate it if you could share your configuration settings (trunk group settings on Alcatel because I believe that the zapata.conf is pretty straightforward).

Thanks!
PS: I already have a working euroisdn pri link between the two PBXs. What I'm trying to do is migrate to QSIG.
Please take a look at this post: viewtopic.php?f=44&t=6298&hilit=qsig+asterisk

Hi,
I have conected Asterisk to an Alcatel OmniPCX4400 with some sucess. Altgough, i still have a problem.
Here is the scheme:

............E1.....................E1
PSTN<---------->Asterisk<--------->Alcatel

When a PBX extension hangup's a Call (to PSTN, or SIP Phone), the call only disconects after 30secounds, more or less.
If the call is disconnected from the softphone, the hangup happens sucessfully.
Do you have any idea how to solve this problem?

Regards,
André

This is known asterisk bug. Search the internet for it.