[Asterisk-Users] IVR Applications

Hello, Could someone please help refer me to a resource where I can find material on how to write IVR applications. I am using [EMAIL PROTECTED] ver. 2.8. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

[Asterisk-Users] AGI to read MySQL

Hello everyone, I am not an expert in Asterisk programming yet. So, can someone help me put my first steps on how to use AGI to access MySQL tables and do queries. Any reference or help is appreciated. My target is to get Festival to read TTS from data stored in MySQL table based on the ID

[Asterisk-Users] MWI not working

Hi everyone, I noticed that the waiting message indicator does not lit when I have a message in my voice mail. Any suggestion why this is happening? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

RE: [Asterisk-Users] AGI to read MySQL

: [Asterisk-Users] AGI to read MySQL Walid, Check the ASTCC agi script ; it just does exactly that: http://www.voip-info.org/wiki-ASTCC Cheers, Fred - Original Message - From: Walid Azab To: asterisk-users@lists.digium.com Sent: Thursday, June 15, 2006 11:14 Subject

[Asterisk-Users] FW: TTS from MySQL

Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 ([EMAIL PROTECTED]). I understand that I need to use Festival and AGI but do not

[Asterisk-Users] TTS to read from Database

Hi everyone, Is there a way to get Asterisk read from MySQL using Festival Text to speech engine?!! Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] FW: WEB SIP Dialer

Hi, I came across this nice looking web SIP dialer. However I cannot find how I can download it. Anyone know how...?? http://www.geocities.com/babarnazmi/ SIP (Session Initiation Protocol) based PC2Phone Dialer

RE: [Asterisk-Users] Full T38 sip Faxing now Available

Hello, Could you provide me with more information on your solution. Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael D Schelin Sent: Thursday, July 28, 2005 6:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion

RE: [Asterisk-Users] Cisco 7940 - Disappearing Clock

I noticed this yesterday. I am using Cisco 7960 with SIP 7.4. It was my first time to see that in two days. I have no idea why this happened either. If you happen to know why, please drop me an e-mail. Thanks Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On

[Asterisk-Users] SIP WEB Phone (Wanna implement Click to Call)

Hi, I appreciate it if someone knows what is available for SIP web phones out there. I am interested in putting a soft phone on a website that registers with Asterisk using SIP. Then, when someone uses it, it directly calls into an asterisk call queue.. Any ideas?

RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Firmware Upgrade Strange Problem Hi On Tue, Jul 26, 2005 at 09:09:27PM +0200, Walid Azab wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry Sent: Tuesday, July 26, 2005 6:00 PM To: Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] H323 Configuration file

Hi, This is what I have and is working just fine. I disabled Asterisk gatekeeper and registered directly to a Cisco CallManager 3.3.4 via h323 trunk. ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter,

[Asterisk-Users] Dial through IAX to FWD

Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have= 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXX What trunk dial

[Asterisk-Users] Recording suddenly stopped

Hi.. I noticed all recording activities suddenly stopped. It seems as if Asterisk is unable to manipulate files. Here is a sample of a session in which I dialed the Voice Mail system and tried to record my name: Any ideas? Thanks Executing VoiceMail("SIP/100-69a9", "[EMAIL PROTECTED]")

[Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the fileOS79XX.TXT from the TFP server. In the file I added the version

RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Title: Message Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Thanks Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins,

[Asterisk-Users] 7960 from SIP to SKINNY

Hello, Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and need to get back to SCCP to be able to use the phone. ThanksWalid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

Title: Message I wentfrom 5.3 to 6.3 then from 6.3 t 7.5 directly. However, I have the warning message (Protocol Application Invalid) Please any help. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Watkins, BradleySent: Tuesday, July 26, 2005 4:12 PMTo:

RE: [Asterisk-Users] 7960 from SIP to SKINNY

26, 2005 7:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 from SIP to SKINNY On Tue, 2005-07-26 at 18:31 +0200, Walid Azab wrote: Hello, Anyone tried reverting to SKINNY from SIP. I have a problem I cannot fix and need to get back to SCCP

RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem

SIP Firmware Upgrade Strange Problem Walid Azab wrote: Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Walid, you should be able to unzip it anywhere

Re: [Asterisk-Users] Asterisk and Cisco CallManager Integration

install the binaries. This will also install a sample configuration file, if there isn't one. Next, add to your LD_LIBRARY_PATH the path where the oh323wrap library was installed (or edit your /etc/ld.so.conf file, add the library path, and run "ldconfig"). Thanks, ~Vamsi On 6/26/0

[Asterisk-Users] Asterisk on Linksys WRT54G

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please

RE: [Asterisk-Users] Asterisk and Cisco CallManager Integration

both sides register with it, everything works. Walid Azab wrote: Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned

[Asterisk-Users] FW: ZAP to SIP Dial Plan

Hi, I would like to setup Asterisk to route incoming calls to ZAP on my TDM400P to SIP phones. What is the best dial plan to use. We are currently able to route outgoing calls to PSTN from SIP to ZAP. Thanks in advance. Walid ___ Asterisk-Users

[Asterisk-Users] Asterisk and Cisco CallManager Integration

Hello, I have Cisco CallManager 3.3.4 and [EMAIL PROTECTED] latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go

[Asterisk-Users] SIP to ZAP Dialplan

Hi, I would like to setup Asterisk to route incoming calls to ZAP on my TDM400P to SIP phones. What is the best dial plan to use. We are currently able to route outgoing calls to PSTN from SIP to ZAP. Thanks in advance. Walid ___ Asterisk-Users

[Asterisk-Users] Install Asterisk on CCM MCS-7835 Server

Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I

[Asterisk-Users] Install Asterisk on CCM MCS-7835 Server

Hi All, I am replacing Cisco Call Manager with Asterisk. As you know CCM is on a MCS 7835 Server which comes with a custom version of Windows. Does any one know how to install Linux on that H/W. My guess is that someone must have tried the same thing before. I know how to install Linux however I

[Asterisk-Users] FW: SIP Phone Choices

Hi, What are the SIP phone models that proved to be working well with Asterisk? I appreciate your recommendations. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Asterisk and SER

Hi Everyone, Just a curious question. Has anyone heard of any service provider who is using Asterisk and SER to provide their VOIP services? Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Analogue Line to Asterisk (Which Digium Model???)

Guys, I need to use Asterisk to call out PSTN numbers via an analogue line. I understand Digium manufactures these kinds of cards, but can someone tell me which model number it is. I really only need a card with one or 2 analogue ports max. Walid

[Asterisk-Users] Asterisk Versioning

Hi, Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS CVS Head I am a newbie and about to install Asterisk on SUSE Server. Can someone please advise what is the best version type and number should I

RE: [Asterisk-Users] Bellster - cool :-)

Bellster.net say that you can: -put a 'q' in front of the number that you are calling. e.g. 'q12125551212' -- Any one knows how you can do that using the phone dialpad? Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan

RE: [Asterisk-Users] H323 Softphone for iPAQ

and ITSP. I'm just about to try it my self ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab Sent: Sunday, January 16, 2005 8:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] H323 Softphone

[Asterisk-Users] H323 Softphone for iPAQ

Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To

[Asterisk-Users] Asterisk to CCM3.3.4 via H32

Hi.. I need to send calls coming from SIP phones behind asterisk to Cisco Call Manager 3.3.4. We have created an H323 trunk on the call manager.Provided that Asterisk-oh323 is installed, how should h323.conf be configured for that? Also when this is done can I setup CCM to alert phones

[Asterisk-Users] oh323 compile error

I am trying to compile oh323 and having the following error. Can anyone help please?! This is my third post. These are the versions I am using: Compilation Error: -- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib

[Asterisk-Users] Add h323 support to Asterisk

I have asterisk CVS-v1-0-12 Can someone please advise what is the best solution and versions for adding h323 support to asterisk. I am confused between oh323/pwlib/asteris-oh323 versions. Asterisk oh323 0.7 README say I need to getPWlib (v1.6.6) and OpenH323 (v1.13.5) but I cannot find

RE: [Asterisk-Users] Asterisk on a notebook?

You might need to go for [EMAIL PROTECTED] Avery simple and easy to install version of Asterisk. Just burn the ISO image to a CD and boot with it and it will automatically install everything for you. However, it will wipe out all your HD and install CentOS then Asterisk. For SIP, you can start

RE: [Asterisk-Users] Cisco 79XX phones not talking to asterisk

If there is a little X next to the line icon then the phone is not registered. Try showing registered phones from asterisk console using SIP Show peers It will tell you which phones are registered with which IPs. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

[Asterisk-Users] Want to install Oh323 and LOST

I need to install Oh323 in order to get Asterisk connect to Cisco CCM 3.3.4. I cannot find the dependencies needed for that. Can any one send me link: The readme file says they areat the following links but the exact version isn't. Please help. Walid o PWlib (Portable Text and GUI

[Asterisk-Users] Build PWLIB

I am trying to build PWLIB to get OH323 up and running. I am not an expert in linux but can someone help telling me how I can do the following: How can I add a directory to LD_LIBRARY_PATH?! Thanks in advance

[Asterisk-Users] Oh323 compilation errors

Hi, well, I really need your help here. I have tried compiling oh323 many times and I always get the following error when trying to "make opt" open h323. Any ideas?! Compilation Error: -- g++ -o obj_linux_x86_r/simph323 -s -L/root/pwlib/lib -L/root/openh323/lib

[Asterisk-Users] What's the easiest way to get * to call PSTN?

Hi, I just want to know what is the easiest way to have Asterisk route calls to PSTN. Hope any one can help me. PS: Any solution using a Cisco device is preferable. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] H323 on Asterisk@Home

Guys, I am about to install H323 [EMAIL PROTECTED] (Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not set up. I also noticed that many of you here say that it is better to use Oh323. What is the best scenario here for me? Should I go with the

FW: [Asterisk-Users] asterisk - oh323 driver

Hi.. I am running Asterisk CVS-v1-0-12/22/04-05:48:41 built on a i686 running Linux (CENTOS) I do not have h323 neither OH323 configured. Accordingly I do not have h323.conf file under /etc/asterisk/sip.conf. Anyway, I wish to install oh323 in order to trunk Asterisk to Cisco 3745. What

[Asterisk-Users] Asterisk version naming convention!!

Dear list, I am running Asterisk CVS-v1-0-12 what is this called in terms of Asterisk versions convention? Is it Stable , Head, latest release !!! Excuseme if the question is too basic, but your help is appreciated. Thanks Walid ___

RE: [Asterisk-Users] H323 on Asterisk@Home

: [Asterisk-Users] H323 on [EMAIL PROTECTED] Walid Azab wrote: Guys, I am about to install H323 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not set up. I also noticed that many of you here say that it is better

RE: [Asterisk-Users] What's the easiest way to get * to call PSTN?

Wehave Asterisk CVS 1.0.2. I intend to connect Asterisk toCisco 3745 unless there is a better way. Asterisk is not configured with any H/W. Cisco 3745will accordingly send the call to the softswitch. PGW2200 which controls our AS5300. Walid From: [EMAIL PROTECTED] [mailto:[EMAIL

RE: [Asterisk-Users] Asterisk to PSTN

Thanks. Any tips on a dial plan example to route from Asterisk to CCM and vice versa? Also with H323 between * and CCM can I still use SIP phones behind Asterisk. ThanksWalid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jo?o AmaroSent: Monday, January 10, 2005 5:04

[Asterisk-Users] Cisco ATA 186 for PSTN dialing

Hi all.. can I configure Cisco ATA 186 to dial out to PSTN? I need a quick and easy to set up scenario to have SIP phones dial PSTN numbers. Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] How to enable debug

Hi.. Can someone help telling me how to enable a full debug mode and how to turn it off again. I need to see what Asterisk is doing behind the scenes. I am able to see the SIP debug events only now. But I still need to see things like voicemail to e-mail activities. Thanks Walid

[Asterisk-Users] Blank Voice Mail messages

Hello, I am having a couple of problems. Any help is appreciated. 1. The voice mail messages arrive in the mailboxes but when I play them back, the IVR tells the time and date of the message but never plays it. It is as if it skips it. 2. Asterisk never seems to send the voice mail as

RE: [Asterisk-Users] Asterisk Demo

Discussion Subject: Re: [Asterisk-Users] Asterisk Demo On Mon, 2005-01-10 at 07:51, Walid Azab wrote: Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. .0) and a SIP client on HP iPAQ via a wireless hotspot. I need

RE: [Asterisk-Users] Asterisk Demo

Ah, ok.. It is a bit clearer now. I'll give it a try. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Monday, January 10, 2005 12:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

[Asterisk-Users] Asterisk to PSTN

I have installed [EMAIL PROTECTED] on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid

[Asterisk-Users] Asterisk Demo

Hi, I need to setup a demo for asterisk and need some help here please. The demo is connecting to Asterisk a Cisco 7970 SIP (ver. .0) and a SIP client on HP iPAQvia a wireless hotspot. I need to configure both with the same extension with a shared line like in Cisco CallManager. This way

[Asterisk-Users] Asterisk SIP channel (PSTN Calls)

Hello Every one I need to enable Asterisk to route external land line calls to the PSTN. Regarding our environment, we have Cisco CallManager (3.3.4) to which IP phones are connected. E1 terminated on a couple of As 5300's which are controlled by a soft switch (Cisco PGW200 Call Control).

[Asterisk-Users] Aterisk@Home

Hi All, Wehave just installed [EMAIL PROTECTED]. It was straight forward as promised. However, I cannot find any guides or tutorials on how to administer this version of asterisk. We plan to install a bunch of Cisco 7960 and 7905 IP phones. I have a test phone which has already been

RE: [Asterisk-Users] Aterisk@Home

, December 22, 2004 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] Walid Azab wrote: Hi All, We have just installed [EMAIL PROTECTED] It was straight forward as promised. However, I cannot find any guides or tutorials on how

RE: [Asterisk-Users] Guide to Cisco 79xx

Try this e-learning tutorial. It requires macromedia flash. http://www.cisco.com/warp/public/779/largeent/avvid/products/7960/router_page.htm http://www.cisco.com/warp/public/779/largeent/avvid/products/7940/index_1020.htm Regards, Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

[Asterisk-Users] Asterisk Hardware

Can I start using Asterisk with a couple of SIP IP phones and Softphone software on users PCs only? I do not have any cards yet and will still have to wait until I order a card. Regards,Walid ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] ZAP and IAX Trunks

Thanks Dean.. Well, about the hardware then. What do you recommend for beginning with Asterisk. I intend to use Cisco 7940s/7960s with Asterisk. Also which software is recommended to enable Soft phone on users PCs? Regards, Walid From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On

RE: [Asterisk-Users] Asterisk and Cisco IP Phones

also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text

RE: [Asterisk-Users] Asterisk and Cisco IP Phones

+0200 From: Walid Azab [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960

RE: [Asterisk-Users] Asterisk and Cisco IP Phones

] Asterisk and Cisco IP Phones No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones. Message: 16 Date: Sat, 4 Dec 2004 12:45:53 +0200 From: "Walid Azab" [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco IP Phones To: [EMAIL PROTECTED]

[Asterisk-Users] ZAP and IAX Trunks

HelloEveryone, I have recently come across these two terms. I am new at Asterisk and do appreciate your assistance in this. Some tools such as "astGUIclient" and "Asterisk Management Portal" require that the phone system be running Zap or IAX trunks as well as SIP devices. SIP devices are

[Asterisk-Users] Asterisk and Cisco IP Phones

Hello Everyone, I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and 7905. Any info or help is appreciated. I know I'll have to use SIP but if I want to use the phones off site meaning from my home for example, how can this be done? Also, regarding external lines what