search for: azab

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 check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

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 to configure both with the same extension with a shared line like in Cisco CallManager. This way if the extension is called both iPAQ and the IP phone ring and the user gets to pick up using either.

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home 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 through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

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*1800XXXXXXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XXXXXX instead of

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

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? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am creating a CentOS 5.1 Paravirtualized guest On Xen 3.2.1 / Fedora 8 Dom0. I built DomU kernel from the Xen Linux 2.6.18.8 source. I also created an initrd image (I face some problems there as I had to manually copy some modules line xennet, ahci and libata). Anyway, I dounloaded the CentOS5.1 image from http://jailtime.org/ and copied it to a new partition. When I create the

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 are the options for Asterisk? Thanks Walid -------------- next part -------------- An HTML attachment

Hi All, We have just installed Asterisk@Home. 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 upgraded to SIP 7. Now the box is ready but we don't know what the next step is!! Any help is appreciated.

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050112/1604835f/attachment.htm

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 !!! Excuse me if the question is too basic, but your help is appreciated. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 ---------------------------------------------------------------------------- ---------------------------------------------------------------------------- ---------------------- For

Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050116/f75bd2fe/attachment.htm

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", "b100@default") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing

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 (Asterisk@home). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2743 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060612/bff4ffa6/attachment.jpeg

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2743 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060615/711bfb60/attachment.jpeg

Hi all, I wonder if there is a certain development environment that has been commonly used to develop the Xen code. I am currently trying to play around with the code and it take me a while to figure out the relations between different variables and functions. I am also wondering if there exist any specification of explanation for the source architecture, like some published design document

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 use. My environment is not so big. I only wish to eventually get my asterisk to talk to Cisco CCM 3.3.4.

I have installed Asterisk@Home 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: