Alcatel Unleashed

It's asterisk side problem.

Don't use some f****g GUI, it makes you don't know whats going on.
open asterisk console, and give it a try:

dialplan show from-zaptel

Then you will see.

Also:
pri debug span 1
(assuming you have span 1 in asterisk box, and asterisk version to 1.4.x in 1.6.x it's a dahdi not zaptel)

Im fairly certain its not Asterisk, its the Alctel. Using this dial plan:

exten => _3XXX,1,Noop(TEST$EXTEN})
exten => _3XXX,n,Answer()
exten => _3XXX,n,Playback(demo-congrats)
exten => _3XXX,n,Wait(10)
exten => _3XXX,n,Hangup

I get "Congratu <hangup>" every single time. Despite initial connection, I see this line in span debug:

Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
> Ext: 1 Cause: Destination out of order (27), class = Normal Event (1) ]

However, I have since found that if I change "Private Trunk Group" to true in the Alcatel, the call stays connected fine. with just one small issue. Extension numbers then appear with a 9 in front of them, so calling from 3001 will show on Alcatel phones as 93001. So there must be a setting someone that I can change to not have this 9 automatically prefixed by Alcatel PBX?

I tell you it's something on asterisk.

It was here discussed many, many times how to setup asterisk & alcatel, also in the Internet there is a guide how to setup this.

There are many possibilities, but to be honest one of them is you try to connect both switches without basic training on either of them.

Please setup on alcatel trunk in network mode=yes with crc4 checking and on asterisk cpe mode, also with crc4.
Please use simple dialplan on both switches.
Please read about external callback translator to remove '9' (sorry, this is basic of ALU OXE), documentation is right-top corner of this forum.

Is there any way to transfer call with Asterisk? I am using latest Asterisk (1.6.0.5) via Q.SIG connection to Alcatel OXE. Everything is fine but I have no idea how to transfer call with Asterisk. Q.SIG protocol knows this but I do not know how complete Asterisk implementation is and found no example on this topic.

Was discussed on this forum, please do search.

It is possible on asterisk using eicon diva server PRI card and Call Diversion on Alcatel OXE side.

iirc it was developed and described by mflores from this forum.

on normal digium hardware with recent libpri it is also possible, but i needed to use 'svn current' version (development version) of libpri and zaptel/dahdi.

Also sending alphanumeric calling name is possible At least in recent version of OXE (i've tested with R7.1).

Very good alternative to some expensive IVR if you don't want/ don't need integration with CSTA.

Thanks for your answer!

krzysioD wrote:Was discussed on this forum, please do search.

I did without relevant result. I have searched with asterisk call transfer and now with asterisk call diversion.

krzysioD wrote:iirc it was developed and described by mflores from this forum.

Found no results on this forum for mrflores as author or as keyword. Do you have any URL for this discussion?

krzysioD wrote:on normal digium hardware with recent libpri it is also possible, but i needed to use 'svn current' version (development version) of libpri and zaptel/dahdi.

I am using latest release version but I'll take a look on trunk branch.

krzysioD wrote:Also sending alphanumeric calling name is possible At least in recent version of OXE (i've tested with R7.1).

Very good alternative to some expensive IVR if you don't want/ don't need integration with CSTA.

Sounds good indeed. That's why I am looking for some information. I have finished IVR on Asterisk but would be great using Call Transfer or Call Diversion instead of wasting two channels per call.

Huh,
it was mgoegel sorry, wrong forum, wrong nick, too early in the morning.
for start:
viewtopic.php?f=44&t=2092&p=16180#p16180

also libpri in asterisk and search for transfer there.

krzysioD wrote:Huh,
it was mgoegel sorry, wrong forum, wrong nick, too early in the morning.
for start:
viewtopic.php?f=44&t=2092&p=16180#p16180

also libpri in asterisk and search for transfer there.

Much better! Thank you very much!

Because that post has been written a year ago I think that function might included in latest release now. But I still not found any traces how to use this transfer function in my dialplan.

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

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

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...