Re: [Asterisk-Users] What to use h323 or oh323 ???
Hi Adeel http://www.inaccessnetworks.com/projects/asterisk-oh323 Please visit there, you will find your way. Bashir - Original Message - From: Adeel -31 To: asterisk-users@lists.digium.com Sent: Saturday, July 02, 2005 9:13 AM Subject: [Asterisk-Users] What to use h323 or oh323 ??? I m new to asterisk n i've got an IP phone that supports h323 protocol but i dont know how to configure asterisk to use it... i m comfortable in using sip iax softphones butthere is no h323.conf in /etc/asterisk/ i read that i've to compile some files but i m confused regarding h323 oh323 .. which one should i use.. plz tell me or atleast give some helpful link __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 , OH323 and Sound Card
Hi all asterisk h323 user Please help me to know is there any sound quality effect without sound card . I am using onboard 32 bit sound card which comes with Asus NCCH-DL mother board. and getting delay from asterisk to any h323 device. but incomming is fine by OH323. 6.5 ver. is there any bit problem or onboard sound card problem or somewhere elase. please let me know. here i got another result , when stop that onboard card and install a old sound card 4 bit , creative and sound problem fixed. now really i am surprise. is there any issue on sound card then please suggest me what model and bit card is perfect for OH323 and my asterisk . Thanks. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
I found a large IAX supported provider beside Voipjet. Now ...? Bashir I still i have good balance with them, I dont know what will be happend. and my canadian DID . - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, June 26, 2005 1:09 AM Subject: Re: [Asterisk-Users] LiveVoip is Bankrupt Yair Hakak wrote: well, i can't say i'm surprised. any company whose approach to customers is you are all scum trying to cheat us, don't ask questions, and we'll help you when we feel like it isn't going to be around for a long time. I agree totally. After seeing some of the issues people were having with their customer support (or better, flying off the handle at their customers) I decided to stay clear of them. Survival of the fittest . . . B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 with g723
Hi I did install and i bought g729 from digium. and my g723 works fine also with quintum but i have now another problem, i found robotic sound. and delay of my outgoing voice. incoming voice is fine. bashir - Original Message - From: Erdem HAKİ [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 20, 2005 1:11 AM Subject: RE: [Asterisk-Users] OH323 with g723 Hi, Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to install g723, but first you have to install g729 http://aussievoip.com.au/wiki-G729-Install I have tested it with Quintum, it works Enjoy :) Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bashir Ullah - www.Lamsre.Com Sent: Monday, June 20, 2005 11:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OH323 with g723 hi is there anybody using g723 with oh323 and sending call by asterisk. if so please let me know how i can use this same, i need to call quintum by g723 . Thanks Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323
Please post ur installation script for chan_h323 - Original Message - From: Atif Rasheed [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, June 20, 2005 7:21 AM Subject: [Asterisk-Users] chan_h323 vs chan_oh323 chan_ooh323 hello there, can somebody please comment which one of these channel drivers will give best output doing g729|g723 pass-thru. only pass-thru is needed no transcoding. please share your experience. if somebody has some figures (simultanous calls using a certain channel driver) it will be apericiated. I have installed chan_h323 (by McNamara) and its working fine with me. I just want to know if I run this driver on a Dual-Xeon machine. can it handle 500 or 500 simultanous calls in pass-thru mode. Regards, -- Atif ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 with g723
hi is there anybody using g723 with oh323 and sending call by asterisk. if so please let me know how i can use this same, i need to call quintum by g723 . Thanks Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register More then 1 by one userid
Hi I tested i can able to register 2 sip phone by same user id and same phone number. I need help to view there IP . i just find one . not two of them, is there any command i can view both registration IP. Thanks. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quantum A800 (SIP) - Asterisk Config
Hi Is there any help for me to register my quantium A800 (SIP) with my Asterisk . Please help me what should me my Sip.conf now present i did [1234567] type=friend context=sip username= secret= nat=yes host=dynamic canreinvite=no defaultip=XXX.XXX.XXX.XXX disallow=all allow=g729 allow=gsm allow=g723.1 allow=ulaw and is there any special change need on quintum? Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly w/*?
Hi I got another problem with Firefly that count time before answer any call. but its sound quality with dialup better then any SIP based free phone. Bashir - Original Message - From: Mojo with Horan Company, LLC [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 19, 2005 9:46 AM Subject: Re: [Asterisk-Users] Firefly w/*? I would recommend connecting Firefly to * with IAX rather than SIP. Firefly's own network is IAX-based so I suspect that their implementation is more full-fledged than their SIP implementation. Firefly crashes from time to time on my system if network connectivity between * and my pc go down, but it's rare enough that I live with it. Biggest two problems I have with Firefly are 1) that it's single line only, and 2) the loud beep when you hang up a call :) Mojo Me wrote: I have seen folks mention FireFly softphone on the list many times. I went to their website but could only find a version which connects directly to their service, it did not seem configurable to use with *. Is FireFly in fact usable with *? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars
I don't see any problem if some one support people to use asterisk friendly then whets other guys problem. I am also consulting and manage my food from asterisk support. Thanks Marks for this, He helps lots of unemployed people to get his own job. Please take it easy way and let them serve and help asterisk to reach door to door. Its 2005 telecommunication year. Hope every body understand what I am want to say, My English is not good, I think this asterisk will take place all other ip based tele product. If i say something excess then forgive me. Bashir - Original Message - From: David Brodbeck [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 12:04 PM Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home andAMPfor over 1000 dollars -Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] I don't think the GPL obliges you to give credit to anybody. In fact, I think that's a key difference between the GPL and the BSD license. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Echo
hi i did not find fxotune under zapte-1.0.6 , please let me know is it different module , need to install seperate, please show me the way , i am having same echo problem and finding its solution for mt tdm fxo. thanks. bashir - Original Message - From: Noah Silverman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 12, 2005 5:43 PM Subject: Re: [Asterisk-Users] Local Echo Great suggestion. I'll try it ASAP. Where do I get fxotune? Thanks! -N Matt Fredrickson wrote: On Tue, Apr 12, 2005 at 10:16:16AM -0700, Noah Silverman wrote: I have a strange echo problem. When speaking on the phone with someone, I hear MY OWN voice with a sever echo. The other party sounds perfect, and they can hear me perfectly. It is as if only the sidetone has an echo. I'm running * on a dedicated box, small LAN, and am using 4 FXO cards to connect the box to PTSN lines. My phones are Polycom IP500 SIP phones. The only echo cancellation stuff that I can find relates to cancelling echo between my system and the PTSN lines. Since the call is perfect, I don't see how this would apply. Any suggestions?? If you're using a TDM card, you might see if the fxotune program will help. It does impedance tuning of the card and finds the line impedance that has the lowest mean power (i.e. least echo). I've been working on it for a while and some people have had some success with it. Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Answer without ringing
Hi all * user I have TDM FXO (4) connected with TELULAR (CELL Phone Device) and they answer without ringing , and also when it goes to phone service provider message like You Have dial wrong number please dial correct number... without any ring and my cdr shows this call answered. Is there any way to avoid this call show as answer. Thanks Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP register more then 1 with same username
Hi all * user I did connected with * from 2 sip-softphone and i registered with asterisk under same username and password and working both fine. but * shows only one. is there any way to find them both by using any tips. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Thanks Yves, Thanks for this good news, that digium going to start h323 channel soon. Oh this is at least one hope i can see. Bashir - Original Message - From: Yves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 22, 2005 11:57 PM Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable If you use open-source software, you have to accept that sometimes project need some times to be stable and have all features. OH323 works - even if there are still a few bugs - and the people around the project are working hard to make to work even better. If you want something that work now, with support , there are plenty of commercial products. I suggest you continue trying oh323, or be ready to pay. I don't know the existence of any other open-source that can do this. Except a post on the dev-mailing telling that Digium was coding his own h323 channel module. WaitSee. Yves Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Hi George I did install and checkup several times, but some times h323 gateway or softswitch cant accept my call and was able to accept call but no sound. so can you help me please to implement a h323 solution. You may contact with me if you want. Thanks Bashir Call. 1-604 323 7991 Mail. [EMAIL PROTECTED] - Original Message - From: George K. Konstantoulakis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 3:11 AM Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable Hello Bashir, what kind of problems are you having with oh323 ? George Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *-1.0.7 DTFM = Not working
Hi I am not good at coding, what i did, i just replace chan_sip.c by version R2 and now my DTMF working , I also faced same problem too. I know this is lay man solution but works. Bashir - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 4:51 PM Subject: [Asterisk-Users] *-1.0.7 DTFM = Not working My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *-1.0.7 DTFM = Not working
that is asterisk-1.0.R2 you can download from digium or asterisk.org - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 5:15 PM Subject: Re: [Asterisk-Users] *-1.0.7 DTFM = Not working If this is the case it would seem to me that chan_sip.c is buggy. Where did you get R2 version? I'll try it. I don't understand how such a major bug got into the CVS-Stable branch. #Joseph On Wed, 2005-03-23 at 17:00 -0800, Bashir Ullah - www.Lamsre.Com wrote: Hi I am not good at coding, what i did, i just replace chan_sip.c by version R2 and now my DTMF working , I also faced same problem too. I know this is lay man solution but works. Bashir - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 4:51 PM Subject: [Asterisk-Users] *-1.0.7 DTFM = Not working My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on my TDM fxo
Hi I am using TDM FXO (4) with one of my server , in middle east and there internet not so good, every time its has some packet loss happend. but speed is good. quite enough for 4 port with ILBC. my problem is i setup the same thing with same config in several country like singapore, bangladesh and finland. they works fine. but there i found lots of echo and i cant stop its echo , I am pazzel last 2 weeks , changing its tx and rx . do u know any best soluation how can i stop them. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.6
Hi after upgrade from R2 to 1.0.6 , my dtmf not working and i cant dial . can any 1.0.6 user help me why i cant do that. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIPJet and g.711
Me too facing same problem . some times it work some times not. i am using ilbc . - Original Message - From: Justin Richards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 12, 2005 8:36 PM Subject: Re: [Asterisk-Users] VoIPJet and g.711 well now i'm confused, because its working again. so i'm not sure if it really had a problem or not. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astcc - how to use **
hi all asterisk user can you help me to find the way for hangup any call by pressing any key like ** or ## in astcc and place another call without providing calling card number. bashir i search google to find out a - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 09, 2005 12:45 AM Subject: meaningful subject [was: Re: [Asterisk-Users] Another Newbie Question] On Wed, Mar 09, 2005 at 06:38:24PM +1100, Callum McGillivray wrote: Hey all, Hi, welcome to this list My apologies if this sounds blindingly obvious, but am I correct in saying that I can use Asterisk to connect two extensions and make calls between them without needing an actual telephone line at all ? I figure it's possible. As I said, probably blindingly obvious. but my techies have gone home for the evening and I was looking for an answer before I left. Suppose someone will have the same question a year from now. He'll try to do the Right Thing and search the archives of this list first. He may get some hits for his search from this thread, but will dismiss them, because the title of the thread was a newbie question and gives no hint to the fact that we're talking about connecting extensions. Cheers -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Exec AGI after hangup.
please use deadagi, and try to do everything inside one agi. - Original Message - From: Dpto. Técnico (Softec) . To: asterisk-users@lists.digium.com Sent: Monday, March 07, 2005 2:08 AM Subject: [Asterisk-Users] Exec AGI after hangup. Hi everybody, I'm trying to implement a enhanced blacklist system using AGI and Perl,configuration in extension.conf is: exten =_numbera,1,AGI,blacklist_2_in.agiexten =_numbera,2,Answerexten =_numbera,3,AGI,xisco_1.agiexten =_numbera,4,AGI,blacklist_2_out.agi The problem that I have now, is that blacklist_2_out.agi doesn't execute. I think this is because in xisco_1.agi the call is hangup at the end. How can I do it in order to execute the AGI? Thanks in advance!!! Have a nice day. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
