similar to: CallerID information on H.323 channel

I've recently noticed some oddities in my CDR records. In some cases the original CallerID that I've set in the .conf file for the extension showed-up in the CDR as the originating extension (on Zap/ devices on the channel bank), and in other places it was the one that I set using Set(CALLERID(num)=<something>) (SIP/ devices). Digging around a little in the source (and doing some

Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Hi, I have some trouble with setting my CallerID if i make an international Call. No Problems with National Calls, i can set whatever I want. We pay for this service but our telephone provider was not able to state clear, wether the number we set on an international call should be shown on the other side. Actually only our base number shows up. If I understand it correctly, in every call the

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing

We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via asterisk, the call goes through OK, but we don't get the number. How can I make this work? h323.conf ======= [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf

Hi FOlks, I am trying to setup remotely an "AudioCodes Mediant 2000" MG Module 2 to work with Asterisk through SIP or H323. But since I don't the product manual, it's being a little hard. Anybody would the manual in PDF(file or URL) to indicate to me? Thanks a lot, Isamar

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

On ISDN lines it's possible to prohibit the presentation of caller id, what if I have a SIP gateway, something like an Audiocodes Mediant 1000. How do I prohibit the caller id presentation on that one? Regards, Kristian -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian@netatonce.se

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Hi, would be glad to be contributing with this question to all the comunity. I?m having a weird issue, suddenly I get channels on IDLE 0x00 state until I do a dahdi restart. Trying to do a call trace to see whats going on deeper, I get surprised when tried to open the .call file to see the log this is incomplete. This is what I see [root at localhost telefonica]# nano

Hi pros, Newbie to asterisk, need some help. My existing senerio is we have 6 analog quintums and 1 digital H323, and our gatekeeper is gnugk openh323 located in US. Our business is Call Center and our method of dial is using prefix and gateway IP provided my Carrier. I also brought two AudioCodes MP108 8 FXS gateways, as our gateway runs on h323 my friend suggested to go for Asterisk. If

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up

Hi, does someone knows a IAX2 provider that allows to freely set the USER_NUMBER to the german pstn? It's no problem if setting the NETWORK_NUMBER is restricted, i'm using this for call-forwarding with the correct number, not something illegal... (i'm not sure about the terminology here, but i think USER_NUMBER is callerid and NETWORK_NUMBER is ANI in the US, correct?)

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a