similar to: Inbound PSTN CLID irratic with A200

Hi all, Just want to check from the list experienced personal about the Digium HPEC, where I had purchased the HPEC and wish to run with TDM card Sangoma A200. I can't install HPEC to run with Sangama A200 card, even I had changed my hpec file from i686 to i386. The error that I had as bellow; > Found key 'HPEC-XXXXXXXXXX' for 2 channels. > Found valid HPEC licenses for 2

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:

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

Or you could just use "System" and do it the easy way! AGI is better suited for more complicated scripting. In any case, check out the TFOT book - you'll like it. -MC _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Collins Sent: Thursday, February 09, 2006 3:13 PM To: Asterisk Users Mailing List -

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.

For our application we want to read the callers number back to them (calling in on a 800# that would somehow point to multiple lines -- figuring that all out still). I was under the impression that calls to 800#'s are unblockable by the caller (irregardless if they have caller id blocked or not). I'm not sure if this is because of ANI or CLID (or if ANI _is_ CLID) or how any of this

Seri, I think you might just need to find the right variable and pass it to your script. Also, you'll probably need to read up on AGI - I'm assuming that's how you would launch your Perl scripts. Here's the list of variables on the wiki: http://www.voip-info.org/wiki-Asterisk+variables Most likely you're looking for the variable referred to as ${CALLERID(num)}

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

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, 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

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

Hello, I'm using Asterisk 1.6.2.0-beta3 with asterisk-addons-1.6.2.0-rc1 to write cdr into mysql. I followed: http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk, and it is mostly working however CLID is not written to mysql, although it appears in Master.csv. What do I possibly missing? TIA -- Best regards, pepesz76

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 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

Hi I've bought the Digium g729 codec and have installed it correctly (I think) >> voip*CLI> show g729 >> 0/0 encoders/decoders of 3 licensed channels are currently in use if I do an echo test either from a sip Cisco 7960 or another hard phone (unbranded) using g729 it sometimes works and sometimes the announcement about the test (echo-test.gsm) fails part way though but the

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hi All - I've been trying to compile Zaptel w/ HPEC, but I've been unsuccessful. The system is CentOS 4.4, zaptel version 1.2.15. I believe I've got all the requisite files, and they're in the right locations in the zaptel tree. When I compile, I get the following warning from make: Warning: could not find /usr/src/zaptel-1.2.15/hpec/.hpec_x86_32.o.cmd for

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

Hey Everyone, Has any successfully installed this on a Toshiba A200? Any feedback would be great. Kind Regards Matthew Ollerenshaw Network Engineer * +612 1300 887 959 | Ê +612 9890 8349 | * mollerenshaw@visinet.com.au<mailto:mollerenshaw@visinet.com.au> | www.visinet.com.au<http://www.visinet.com.au/> Diploma (Network Engineering) | Microsoft Certified Professional ( MS

Hi Guys, Looking for a powerful box that is compact, can take two hard drives for Raid-1 (no SSD, too expensive), have at least two Gig ports or two 10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card which needs as much room as a PCIe and doesn't need the actual slot. That is for Sangoma A200 + Daughter card. I would be really glad if I can get some sort of a KVM