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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

...IONED" ; Line 6 Registration Password # NAT/Firewall Traversal nat_address: "" voip_control_port: "5060" start_media_port: "16384" end_media_port: "32766" # Phone Label (Text desired to be displayed in upper right corner) phone_label: "WaZaB-SIP" ; Has no effect on SIP messaging # Time Zone phone will reside in time_zone: EST # Phone prompt/password for telnet/console session phone_prompt: "Cisco7960" ; Telnet/Console Prompt phone_password: "abc"...

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

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:

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

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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.

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 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.

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). What is the best scenario to route external calls to PSTN. Should I use SIP or just connect Asterisk

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 an attachment!! Please see below my config files: Thanks Walid --------------------

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

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 are at the following links but the exact version isn't. Please help. Walid o PWlib (Portable Text and GUI C/C++ Class Library) download from http://sourceforge.net/projects/openh323 (v1.6.6) (required)

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, 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 ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat

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 ./obj_linux_x86_r/main.o -lh323_linux_x86_r -lpt_linux_x86_r -lpthread -lssl -lcrypto -lexpat -lresolv -ldl

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

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

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