similar to: q931.c modifications for CLID Presentation

Hi Guys, Can q931.c be re-compiled using gcc or something else without the need to re-do the whole libpri? Some changes were made to q931.c and I want those to be reflected in .a .o .so .lo files as I think those are the files read by Asterisk vs the .c file. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Hi All, For some reason since moving to Asterisk 1.6. my CDR records are no longer displaying the Clid field. The CDR records contain the Source field be for some reason not the CID details. I am logging CDR to mysql. Is anyone able to help? Regards David. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Please help me, i need transfer a call in asterisk to other telco number and free the channel. Can i do with any q931 function?. Thanks a lot Aris... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090528/2bcd93ae/attachment.htm

Hi, does anyone know if its feasible to run asterisk with a PRI card but not run any q931 signalling.. basically push calls down the PRI and tell asterisk in some other way to pickup a particular Zap channel? Steve

Hi all, I'm looking for a way with any asterisk-version with TE410P (cpe EuroISDN, Q931) for sending an INFORMATION ELEMENT KeypadFacility, eg. *87, during a connected call to the PSTN switch. Are there existing functions in asterisk to generate & send such IE ? If not what existing modules would be best to derive from? TIA, Bruno -------------- next part -------------- A non-text

Specifically: exten => s,1,System(/usr/sbin/myperlscript.pl ${CALLERIDNUM}) will execute myperlscript.pl with the caller id as an argument as the first priority. hth -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Thursday, February 09, 2006 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Possible for Asterisk to

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

Hi all, please forgive me relative lack of knowledge with Asterisk, but I've not played with PBX systems for a while and I'm just re-finding my feet. I've set up my first Asterisk server, I have it configured with a Digium X100P Analogue pots board, I have my Called ID working and everything is hunky dorey with the server itself. But. (and there's always a but) I want to pass

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to "no" and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce <bruceb444 at gmail.com> wrote: > Futher check

Hi all, I have a strange problem, regarding zap channels and cdr. I am using asterisk bristuffed version Asterisk 1.2.2-BRIstuffed-0.3.0-PRE-1i, Copyright (C) 1999 - 2006 Digium, Inc. and others. with two billion ISDN cards. I also installed asterisk addons, last stable version via cvs internal calls, or calls starting from internal sip or iax phone are recorded in the cdr all without any

- I've just installed 0.6.2, & I would like to see the q931 messages going back & forth. I turned on debugging with "h323 debug toggle", which the README says is "very verbose", but I don't see much. Is there a way for me to see more debugging information, like the "debug isdn q931" of IOS? Or am I missing something? Thanks, Glen IAS

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

Hello All, Need a little direction, please. I have searched the lists, WIKI, and googled a problem that I'm sure I'm overlooking. I understand why Asterisk/Zaptel waits two rings to answer (caller ID must be sent) but can I reduce the amount of time it takes before Asterisk/Zaptel answers? In other words, I'm not concerned about Caller ID and want the line answered as quickly as