[Asterisk-Users] Extensions base policy

Dear List, I am new in this world (Asterisk) and facing a problem . i want to make some group, base on extensions .so that certain extensions could call to certain predefine number only. let me give u all a short example extensions 1,2,3,4 will be group A , extensions 5,6 will be group B .

Re: [Asterisk-Users] Extensions base policy

contextes. For example, context_A and context_B. For all group A extensions, make context_A their default context and group B extensions to context_B. Then, in each context, define only the extenions that can be reached. - Gabe - Original Message - From: Mohammad Salaque [EMAIL

Re: [Asterisk-Users] Extensions base policy

and one more information . user will dial to pstn number as well as local extensions thanks Salaque On 3/10/06, Mohammad Salaque [EMAIL PROTECTED] wrote: my server will be in one country . and one group will be on another country. . so pppoe will not work in here i think thanks Salaque

[asterisk-users] Voicemail's mail formate

Dear all I am using AMP 1.0.10 . my voicemail system working perfectly, but now i like to sent user PIN ( Password of the extension) number with that mail . how could i read the user passwd value (PIN) so that i could append the mail format thanks Salaque

Re: [asterisk-users] H323

any one try that with g723 codec? thanks Salaque On 8/27/06, Rosli Sukri [EMAIL PROTECTED] wrote: i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeeting On 8/26/06, atik khan [EMAIL PROTECTED] wrote: Hi, i used to work ooh323 with

Re: [Asterisk-Users] Extensions base policy

; Use two contextes. For example, context_A and context_B. For all group A extensions, make context_A their default context and group B extensions to context_B. Then, in each context, define only the extenions that can be reached. - Gabe - Original Message - From: Mohammad

[Asterisk-Users] Asterisk and gateway

Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give

Re: [Asterisk-Users] Asterisk and gateway

and make changes according in extension.conf for routing. ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear all, I am really new in this world, In my office i setup a Asterisk and all extensions are working fine . i also add some trunk to route my local calls. As some user

Re: [Asterisk-Users] Asterisk and gateway

should help you ram On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote: thanks ram all they give me an ip address. and told me to send SIP traffic. so on sip.conf should i add only these ? [worlgateay] host= xxx.xxx.xxx.xxx thanks Salaque On 3/22/06, ram [EMAIL

[Asterisk-Users] Dialling Problem

Dear List, I am facing another strange problem . some of my envisions like to use other prepaid card (whatever they found in market) but when they dial that access number (phone number to put the pin) they get IVR (Please provide your pin number ) but when my user press pin its not going through,

Re: [Asterisk-Users] Dialling Problem

thanks Dovid , i solved that yes it was DTMF issue . thanks Salaque On 3/24/06, Dovid Bender [EMAIL PROTECTED] wrote: probably a DTMF issue. Try changing it. Font have the link here. Go to voip-info.org and search for DTMF type. --- Mohammad Salaque [EMAIL PROTECTED] wrote: Dear List

[Asterisk-Users] WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was 4 bytes from RTP

Hi all , I am gettign this warning in my asterisk log after installing g723 codec :WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was 4 bytes from RTP what that mean ? thanks Salaque ___ --Bandwidth and Colocation provided by

[Asterisk-Users] What codec extensions using now?

Hello list, Another newbie question,. if I put disallow=all and allow=g723 my sip.cof does it mean that extension could only communicate using g723 ? bellow is one of my extension example [10112] username=10112 type=friend secret=x record_out=Adhoc record_in=Adhoc qualify=no port=5060

[Asterisk-Users] Gsm Gateway , again !

Dear all I am looking for suggestion , solution. my scenario is : ppls will call to my gsm number my gsm will response IVR from my Asetrisk box. ( with a2blling ) and could make call now i am looking for cheap hardware solution for that gsmgateway .my Budget is around 5000US$ for 24 gsm

[Asterisk-Users] Confused !

Hello list, I'd like to share something u all , so that i could understand whats going on into my Asterisk box. i have a setup like this client(ip phone) -ip network--- [Asterisk]ip network ---[Service provider] i have configured A2biling in my Asterisk box. so when client

Re: [Asterisk-Users] Confused !

thanks for your replay, after i disallow all codec except g723 i also confused how a2billing is working then what i did , i removed all the codec from /usr/lib/astersik/module without codec_g723.so . then i saw in my log while user calling to my ivr access number a2b is looking for gms codec as

Re: [Asterisk-Users] Confused !

to tell if this would work. Your restriction is what the service provider allows. Most (that I've used) allow g729. I know it uses more bandwidth than g723 but nothing like G711 (ulaw or alaw) and from my experience, the quality is quite reasonable. - Original Message - From: Mohammad

Re: [Asterisk-Users] Confused !

with. Please understand, I am trying to help and I don't know which parts (of what I'm saying) are not entirely accurate but normally if I say something wrong there are enough people who clamour to correct me. - Original Message - From: Mohammad Salaque [EMAIL PROTECTED] To: Asterisk Users

[Asterisk-Users] My Call drop after 60 to 63 Seconds!!

Dear all, I have an Asterisk box running [EMAIL PROTECTED] 2.7 . and A2billing. when my asterisk box dial using dialcommand_param=|45|HL(%timeout%:61000:3) its working fine . but when i use dialcommand_param=|45|L(%timeout%) call got drop after 62 seconds. i used this same setting into

[Asterisk-Users] how to decrease answer time !

Dear list i am using Asterisk 1.2.5 with [EMAIL PROTECTED] . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

Re: [Asterisk-Users] how to decrease answer time !

pressing # after dialing the number. On most phones, that will make it dial the number. Good Luck, bp On 6/1/06, Mohammad Salaque [EMAIL PROTECTED] wrote: Dear list i am using Asterisk 1.2.5 with [EMAIL PROTECTED] . here is my problem. if i dial a number (consider 79) i have to wait around 20

[asterisk-users] How to soft hangup all channels at a time .

Dear all, Anyone can point me how to soft hangup all channels using single command ? I am using Asterisk 1.4.15. thanks Salaque ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or