[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 update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )but ooh323 works fine with me thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to decrease answer time !
thanks William # solved my problem /Salaque On 6/1/06, William Piper [EMAIL PROTECTED] wrote: That's an issue with your IP phone. Check your configuration. I believe most phones call that digit timeout or something like that... it should be set to about 3-4 seconds. You can also try 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 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 my other two Asterisk boxes and those r working fine . but now i am trying this into a new Dell PC (GX series) and facing that strange problem. i think its something related to my PC . or Asterisk setting. Could anyone guide me where to look for thanks Salaque -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. 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 call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confused !
how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able 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 Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: 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 all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. 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 call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
Re: [Asterisk-Users] Confused !
thanks i alreday did that , you r very helpfull thanks again /Salaque On 5/15/06, AR Tarzi [EMAIL PROTECTED] wrote: 1. In the extension definition, insert canreinvite=yes for each of your clients. 2. In the trunk definition, insert canreinvite=yes Read about reinvite at http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite Apparently some hardware does not like it, and obviously, both the client and the service provider with have to be able to use the same codec (for them to be able to talk to each other) but better if Asterisk is restricted to that codec on both sides to start 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 Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 14:16 Subject: Re: [Asterisk-Users] Confused ! how to use reinvite in my asterisk setup ? thanks Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: I'm not an authority but why don't you get some g729 codecs (10 or so) and use g729 all around. Not allowing for ADSL overheads you can calculate your own requirements on http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double the results since each call is turned around to your service provider.) I would have thought it would be better if you could use reinvite to let your clients speak directly to your service providers. Someone who knows better ought to be able 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 Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, May 14, 2006 11:27 Subject: 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 all the audio file is in gsm format. but what my understanding was it should drop the connection as i only allow g723 . what is found today from one of my frnd telling me that actual bandwidth calculation For codec g723 incoming and g723 outgoing we need: 48.89kbps For codec g723 incoming and g711 outgoing we need: 114.03kbps So, to run 8 calls we will require 902.15 kbps or 0.9 mbps For 10 calls we need 1140.3 kbps or 1.1mbps Each call has RTP, UDP, IP, Codec and SIP overhead. so what u guys suggest , should i record all my ivr file in g723 format all . increase my bandwidth! /Salaque On 5/14/06, AR Tarzi [EMAIL PROTECTED] wrote: Unless reinviting works, wouldn't that add up to what he's experiencing ? client - asterisk - service provider.. makes that 180k each connection so 4 of them would give 800k or so. What I can't understand is: if only g723 is allowed, and Asterisk only allows it as passthrough, how's the A2billing IVR working ? I have to assume G711 (ulaw or alaw) is used. - Original Message - From: Woodoo People .pGa! [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 13, 2006 23:36 Subject: Re: [Asterisk-Users] Confused ! Install iptraf, that will allow you to check incoming and outgoing traffic (or trafshow what do that on /host basis, but not so detailed info) If you choose ulaw, that should take about 90kbps fullduplex traffic. 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 call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? -- WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com [EMAIL PROTECTED]@RedHat.users
[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 call to my Asterisk A2billing's ivr respoce , my client authenticate there pin and call . all my IVR file is gsm format (i got that from a2billing by default) i configured each client disallow=all context=from-internal canreinvite=no callerid=device 20004 allow=g723 so client is only using g723 i think.. but the problem i am facing now . when there are 4 calls in my server i saw my bandwidth reach around 1 mbps /1 mbps . why my server taking so much bandwidth ? Please suggest me what to do ? /Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 gateways. I am planing to use 3 tenor ( with 8 ports fxo each ) and 24 telullar but will that work? as i did that using land line and its working fine . pls let me know iif anyone has solutions (under my budget) Thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 nat=yes host=dynamic dtmfmode=rfc2833 disallow=all context=Office-lan canreinvite=no allow=g723 thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, 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, that IVR even can't get wrong pin number . just get disconnected as no pin number provided. what could be the problem ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my mail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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, that IVR even can't get wrong pin number . just get disconnected as no pin number provided. what could be the problem ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and gateway
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 PROTECTED] wrote: Hi add that information ( which you got from SIP provider) in to sip.conf 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 needs to call overseas so we are looking for cheap gateway with SIP supported. Now . we found one gateway . they just give me there softwitch's ip and told to me to send my call there. i have no idea how to router my traffic from my Asterisk box to that ip . as from all other service i got a username/passwd . Could anyone give me any idea ? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my mail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and gateway
Dear Ram .
u miss something . as i told u . my provider didn't give me any
username /passwd.
they just give me IP address . as they are gateway provider not gatekeeper .
i need to send my traffic to there IP address. they give me only IP
address nothing else
thanks
Salaque
On 3/22/06, ram [EMAIL PROTECTED] wrote:
Hi
ya thats correct
[voip.provider.net-out]
type=peer
secret=password
username=2345
host=ipaddress
fromuser=2345
nat=yes
In extensions.conf you'd then use a statement like this:
exten =
_9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
just example above 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 PROTECTED] wrote:
Hi
add that information ( which you got from SIP provider)
in to sip.conf
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 needs to call overseas so we are looking for cheap
gateway with SIP supported.
Now . we found one gateway . they just give me there softwitch's ip
and told to me to send my call there.
i have no idea how to router my traffic from my Asterisk box to that
ip . as from all other service i got a username/passwd .
Could anyone give me any idea ?
thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Got my mail ?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Got my mail ?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions base policy
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 On 3/10/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, Maybe this isn't the right way...but this is the first thing that popped into my head; 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 Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 11:44 PM Subject: [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 . so group A only allowed to call one or certain predefine number but group B could call anywhere. could any one give me an example of that configuration ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my mail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 On 3/10/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, Maybe this isn't the right way...but this is the first thing that popped into my head; 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 Salaque [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 11:44 PM Subject: [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 . so group A only allowed to call one or certain predefine number but group B could call anywhere. could any one give me an example of that configuration ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my mail ? -- Got my mail ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions base policy
thanks yep . it should ok . let me try
thank again
Salaque
On 3/10/06, Melcon Moraes [EMAIL PROTECTED] wrote:
I guess this is the right way. That's the context's job. Don't forget to
setup your users'es context in iax.conf and/or sip.conf according to
contexts you just created in extensions.conf
For instance:
# iax.conf/sip.conf
[user_grp_A]
context=grp_A]
...
all_the_other_stuff
[user_grp_B]
context=grp_B
...
all_the_other_stuff
# extensions.conf
[grp_A]
exten = _${pre-defined-number},1,Dial(Technology/Channel/${EXTEN})
exten = _${pre-defined-number},n,Hangup
[grp_B]
exten = _X.,1,Dial(Technology/Channel/${EXTEN})
exten = _X.,n,Hangup
Of course this is too simple, but I think it is enough to give you an idea.
Regards
Melcon Moraes
Gabriel Afana wrote:
Hi,
Maybe this isn't the right way...but this is the first thing that
popped into my head;
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 Salaque [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 09, 2006 11:44 PM
Subject: [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 .
so group A only allowed to call one or certain predefine number but
group B could call anywhere.
could any one give me an example of that configuration ?
thanks
Salaque
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Got my mail ?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[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 . so group A only allowed to call one or certain predefine number but group B could call anywhere. could any one give me an example of that configuration ? thanks Salaque ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
