[asterisk-users] Fwd: Switchvox SOHO 4.5 is Here
If you are having trouble reading this email, read the online versionhttp://now.eloqua.com/es.asp?s=491e=78675elq=55426a8b6c714f5bb6f2bf4b5d37bf55 . http://app.en25.com/e/er.aspx?s=491lid=215elq=55426a8b6c714f5bb6f2bf4b5d37bf55 Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release announcement. You are obliged, under the terms of your NDA with Digium, to keep this information confidential until the Switchvox SOHO 4.5 release is announced on March 30, 2010.* Digium is pleased to announce the upcoming release of Switchvox SOHO 4.5. This new version of Switchvox SOHO incorporates some of the most popular features from Switchvox SMB 4.0 and 4.5, including: - Phone Feature Packs - Video calling HD Voice support - Internationalization (BRI support language support for Italian, Spanish and U.K. English) - New appliance Availability The update to Switchvox SOHO will be available for March 30 2010, with the new appliance shipping April 30. *For a quick overview of the features in Switchvox SOHO, watch this on-demand webcast at your convenience:* [image: Watch the Webcast]http://www.brainshark.com/digium/vu?pi=6801699int2=003800sluErtx=l...@voicefidelity.net If you have any questions be sure to let me know. Jim Butler Director of Global Channel Sales Digium, Inc Copyright © Digium, Inc. - The Asterisk Company 445 Jan Davis Drive NW, Huntsville, AL 35806 Visit our website: Digium.comhttp://now.eloqua.com/e/er.aspx?s=491lid=36elq=55426a8b6c714f5bb6f2bf4b5d37bf55| Unsubscribehttp://now.eloqua.com/e/er.aspx?s=491lid=97elq=55426a8b6c714f5bb6f2bf4b5d37bf55| Update Subscriptionshttp://now.eloqua.com/e/er.aspx?s=491lid=97elq=55426a8b6c714f5bb6f2bf4b5d37bf55 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASTERISK y AGC
IN ENGLISH VERSION: Good night I have mounted the system of predictive marker ASTGUICLIENT in 2 Servants, in one of them who are a Server HP Proliant G3 350 3GB ram with 11 Slackware and Asterisk 1.2.12.1, this single server is in charge of the voice. Soon another server a little but modest (HP ML110 3.2GB), that has Apache and the BD MySQL with ASTGUICLIENT 2.0.1, my agents is connected to this I complete q already the system draws out there, softphone Eyebeam of my agents estan connected to server VOIP, I have in addition in the server voip a card digium to 2 installed ports 2 lines ISDN, In the ASTGUICLIENT I have several campaigns or working, which remove to calls through lines ISDN, single one of these campaigns Extraction the calls through a main IAX or SIP that I have with my branch of another country or my supplier of voice. I have tip maximum of 25 agents connected, the problem this in which arrives a little while in which the marker no longer passes calls to the agents, rather is delayed too much in marking and to pass the call., If one marks from softphone (eyebeam) in pantallla of same I obtain message TRYNG. And after several seconds it manages to remove the call, in the case of my agents q work with the marker, hope by a long time q pass the calls to him. To that it must east problem, the load of my servants is not much not to pass of 5% in his load average? I have updated the version of asterisk that tapeworm and of UNDER thinking q podria to solve but with the new versions the problem even appears? Sera q with 2 servants even sharing the load cannot put but of 20 agents to the system? Previously I had 20 agents everything in a servant, east tapeworm problem but the system was not very slow and towards very difficult the work thus decidi to divide to him to the load to the servant clearing to him the BD and the WEB to happen to him to another servant. Some of you has had east problem, of q forms have solved it Thanks beforehand for its answers Greetings. On 12/8/06, Aldo Alexander Leyva Alvarado [EMAIL PROTECTED] wrote: Buenas noches Tengo montado el sistema de marcador predictivo ASTGUICLIENT en 2 Servidores, en uno de ellos que es un Server HP Proliant G3 350 3GB RAM con Slackware 11 y Asterisk 1.2.12.1 , dicho server solo se encarga de la voz. Luego otro server un poco mas modesto (HP ML110 3.2GB), que tiene Apache y la BD MySQL con el ASTGUICLIENT 2.0.1, mis agentes se conectan a este ultimo ya q el sistema corree alli, los softphone Eyebeam de mis agentes estan conectados al server VOIP, Tengo ademas en el server voip una tarjeta digium de 2 puertos instalados 2 lineas ISDN, En el ASTGUICLIENT tengo varias campañas ya trabajando, las cuales sacan llamadas a traves de las lineas ISDN, solo una de estas campañas Saca las llamadas a traves de una troncal IAX o SIP que tengo con mi filial de otro pais o mi proveedor de voz. Tengo un pico maximo de 25 agentes conectados, el problema esta en que llega un momento en que el marcador ya no pasa llamadas a los agentes, mejor dicho se demora demasiado en marcar y pasar la llamada.,Si uno marca desde un softphone (eyebeam) en la pantallla de mismo obtengo el mensaje TRYNG ..Y despues de varios segundos logra sacar la llamada, en el caso de mi agentes q trabajan con el marcador, esperan por un largo tiempo q le pasen las llamadas. A que se debe este problema, la carga de mis servidores no es mucha no pasar de 5% en su load average? He actualizado la version del asterisk que tenia y del SO pensando q lo podria solucionar pero aun con las nuevas versiones el problema se presenta? Sera q aun con 2 servidores compartiendo la carga no pueda meter mas de 20 agentes al sistema? Anteriormente tuve 20 agentes todo en un servidor, no tenia este problema pero el sistema estaba muy lento y hacia muy dificil el trabajo por lo cual decidi dividirle la carga al servidor quitandole la BD y el WEB para pasarle a otro servidor. Alguno de ustedes ha tenido este problema, de q forma lo han solucionado Gracias de antemano por sus respuestas Saludos Aldo Leyva ___ --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 -- Lito Manansala www.voicefidelity.net Mobile: +63.906.437.0459 PSTN: +63.44.790.6292 sip:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] skype: bulcrack ym: onchang_2000 ___ --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] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
Hello Guys,We have a problem in configuring Sangoma A104. We want the 2 ports to beconfigured as E1 and the 2 ports as T1.We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart.[EMAIL PROTECTED]:/tmp# wanrouter restartShutting down wanpipe1 interface: w1g1Shutting down device: wanpipe4Shutting down device: wanpipe3 Shutting down device: wanpipe2Shutting down device: wanpipe1No devices running, Unloading ModulesStarting WAN Router...Loading WAN drivers: wanpipe done.Starting up device: wanpipe1Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsStarting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !! : ioctl(wanpipe3,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsStarting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !! : ioctl(wanpipe4,ROUTER_SETUP) failed: : 22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errorsConfiguring interfaces: w1g1done.Configuring interfaces: w2g1 w2g1: unknown interface: No such devicedone. Configuring interfaces: w3g1 w3g1: unknown interface: No such devicedone.Configuring interfaces: w4g1 w4g1: unknown interface: No such devicedone.Any help will be appreciated.Thanks, Lito ___ --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] Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
I emailed then last 2 hours ago. Just waiting for their reply.ThanksOn 8/31/06, Moises Silva [EMAIL PROTECTED] wrote:Sangoma has excellent support, why dont you ask them?On 8/31/06, Angelito Manansala [EMAIL PROTECTED] wrote: Hello Guys, We have a problem in configuring Sangoma A104. We want the 2 ports to be configured as E1 and the 2 ports as T1. We already run wancfg and configure the 2 ports as T1 and the last 2 ports as t1. Below is the logs when we issue wanrouter restart. [EMAIL PROTECTED]:/tmp# wanrouter restart Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe4 Shutting down device: wanpipe3 Shutting down device: wanpipe2 Shutting down device: wanpipe1 No devices running, Unloading Modules Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 Starting up device: wanpipe2 wanconfig: WAN device wanpipe2 driver load failed !! : ioctl(wanpipe2,ROUTER_SETUP) failed::22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe3 wanconfig: WAN device wanpipe3 driver load failed !!: ioctl(wanpipe3,ROUTER_SETUP) failed: :22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Starting up device: wanpipe4 wanconfig: WAN device wanpipe4 driver load failed !!: ioctl(wanpipe4,ROUTER_SETUP) failed::22 - Invalid argument Wanpipe driver did not load properly Please check /var/log/wanrouter and /var/log/messages for errors Configuring interfaces: w1g1 done. Configuring interfaces: w2g1 w2g1: unknown interface: No such device done.Configuring interfaces: w3g1 w3g1: unknown interface: No such device done. Configuring interfaces: w4g1 w4g1: unknown interface: No such device done. Any help will be appreciated. Thanks, Lito ___ --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--Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lito Manansala www.voicefidelity.netMobile: +63.906.437.0459PSTN: +63.44.790.6292sip:[EMAIL PROTECTED]msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] Digium TE410P configuration to connect with CIsco 3800
Hello List, Can anyone here has a working configuration of any digium e1 card that is connected to cisco 3800. Any help will be appreciated. THanks, Lito ___ --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] Re: Digium TE410P configuration to connect with CIsco 3800
Thanks for your reply. here is my zapata.conf configuration [trunkgroups] [channels] context=default switchtype=national signalling=pri_cpe usecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group=1 channel = 1-15 channel = 17-31 I noticed that when i reload chan_zap.so command there is a warning like this: == Parsing '/etc/asterisk/zapata.conf': Found Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring switchtype Jun 29 21:50:58 WARNING[739]: chan_zap.c:10886 setup_zap: Ignoring signalling -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling signalling -- Reconfigured channel 3, PRI Signalling signalling -- Reconfigured channel 4, PRI Signalling signalling -- Reconfigured channel 5, PRI Signalling signalling -- Reconfigured channel 6, PRI Signalling signalling -- Reconfigured channel 7, PRI Signalling signalling -- Reconfigured channel 8, PRI Signalling signalling -- Reconfigured channel 9, PRI Signalling signalling -- Reconfigured channel 10, PRI Signalling signalling -- Reconfigured channel 11, PRI Signalling signalling -- Reconfigured channel 12, PRI Signalling signalling -- Reconfigured channel 13, PRI Signalling signalling -- Reconfigured channel 14, PRI Signalling signalling -- Reconfigured channel 15, PRI Signalling signalling -- Reconfigured channel 17, PRI Signalling signalling -- Reconfigured channel 18, PRI Signalling signalling -- Reconfigured channel 19, PRI Signalling signalling -- Reconfigured channel 20, PRI Signalling signalling -- Reconfigured channel 21, PRI Signalling signalling -- Reconfigured channel 22, PRI Signalling signalling -- Reconfigured channel 23, PRI Signalling signalling -- Reconfigured channel 24, PRI Signalling signalling -- Reconfigured channel 25, PRI Signalling signalling -- Reconfigured channel 26, PRI Signalling signalling -- Reconfigured channel 27, PRI Signalling signalling -- Reconfigured channel 28, PRI Signalling signalling -- Reconfigured channel 29, PRI Signalling signalling -- Reconfigured channel 30, PRI Signalling signalling -- Reconfigured channel 31, PRI Signalling signalling Then my zaptel.conf is this loadzone = us defaultzone = us span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 # set this to 16 for E1 On 6/29/06, Massimo Nuvoli [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Angelito Manansala ha scritto: Hello List, Can anyone here has a working configuration of any digium e1 card that is connected to cisco 3800. The problem is the router configuration... you need these setups to try some configuration on the Linux side. Otherwise: try a cross cable (PRI cross cable is really different) and some configuration, with E1 you have only two/four configuration possibile for the D channel, 8 if you consider also CRC. All changes are in zaptel.conf in the span line (see the documentation). This is the last configuration found for a CISCO router with E1 interface, is the SAME configuration for the E1 line directly coming from telco in italy. I made some effort to obtain the router working exactly as the telco. I dont have the router configuration. zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf signalling = pri_cpe channel = 1-15,17-31 resetinterval = never immediate=no overlapdial=yes -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEo9g33r4gvOdjXD0RApSvAKCBW7e0W7uKvbsgR9oH+PcS+J5Y6ACg04PB sGt2zlBRs/vP11FeDoCBDL0= =Lz42 -END PGP SIGNATURE- ___ --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 -- Lito Manansala www.voicefidelity.net Mobile: +63 906 437 0459 DID: (+63) 44 7906292 msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] free sun boxes
how much are you selling that stuff? On 6/18/06, Mike Fedyk [EMAIL PROTECTED] wrote: I'm in southern California, are you close or can you ship? Bob Knight wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, bk... ___ --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 -- Lito Manansala www.voicefidelity.net Mobile: +63 906 437 0459 DID: (+63) 44 7906292 msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] ztdummy
you only need that for conferencing ang trunking.On 6/14/06, Doug Crompton [EMAIL PROTECTED] wrote: Question... is ztdummy required if you use no internal cards, in my casean SPA-3000 fxs/fxo ?I have been running without it and see nothing not working. Using a 2.4 kernel, 1.2.9.1 *Doug*Doug Crompton **Richboro, PA 18954**215-431-6307*** * [EMAIL PROTECTED]** http://www.crompton.com*___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Angelito Manansalawww.voicefidelity.netMobile: +63 906 437 0459 DID: (+63) 44 7906292msn: [EMAIL PROTECTED]skype: bulcrack ___ --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] CLI notice: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15
Hi there,Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4.here are the CLI actual logs: -- Executing SetAccount(Local/[EMAIL PROTECTED] ,2, XX) in new stack -- Executing AGI(Local/[EMAIL PROTECTED],2, call_log.agi|50015308467418) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi -- Zap/22-1 is proceeding passing it to Local/[EMAIL PROTECTED],2Mar 26 08:20:26 NOTICE[21367]: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame 15 -- AGI Script call_log.agi completed, returning 0 -- Executing AGI(Zap/11-1, agi-VDADtransfer.agi|8365) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi- VDADtransfer.agi -- AGI Script call_log.agi completed, returning 0 -- Executing DeadAGI(Local/[EMAIL PROTECTED],2, VD_hangup.agi|h) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi -- AGI Script VD_hangup.agi completed, returning 0Thanks,Lito ___ --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] FXS channel banks
rhino channel bankOn 3/23/06, C F [EMAIL PROTECTED] wrote: Carrier Access Adit 600On 3/23/06, Curt Shaffer [EMAIL PROTECTED] wrote: Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations? Thanks Curt ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelito Manansalawww.voicefidelity.netMobile: +63 906 437 0459DID: (+63) 44 7906770US DID: +1 619 399 0128 msn: [EMAIL PROTECTED]skype: bulcrack ___ --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] TE40X zapata.conf configuration sample
Hello list,can anyone give me sample conf file for TE406Any help will be appreciated.ThanksLito ___ --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] Is Voxee down?
Hi Guys,We cant send calls and register to voxee server. however we already send support ticket and waiting for their reply. anybody experience this also on voxee?-- Best Regards,Angelito Manansala www.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] Is Voxee down?
can you send calls?On 1/27/06, Guillermo Salas M [EMAIL PROTECTED] wrote: Con fecha 26/1/2006, Angelito Manansala [EMAIL PROTECTED]escribió:Hi Guys,We cant send calls and register to voxee server. however we already send support ticket and waiting for theirreply. anybody experience this also on voxee?The same issue. Can not register.--Best Regards,Angelito Manansala www.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrac ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Best Regards,Angelito Manansala www.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED] skype: bulcrack ___ --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 view Q.931 Disconnect code
got it.. well it work thanks buddy!On 1/25/06, Andy Kuo [EMAIL PROTECTED] wrote:
I replied the following and got a returned to sender message fromthe MAILER-DAEMON.Not sure if you got it.Here it is again...On 1/23/06, Andy Kuo [EMAIL PROTECTED]
wrote: Hi, Try exten = h,1,NoOp(${HANGUPCAUSE}) in your extensions.conf Cheers. Andy On 1/23/06, Angelito Manansala
[EMAIL PROTECTED] wrote: Hi there, Can anyone know how to view asterisk disconnect code.? -- Best Regards,
Angelito Manansala www.voicefidelity.net Mobile: +63 917 542 5807 DID: (+63) 44 7906770 US DID: +1 619 399 0128
msn: [EMAIL PROTECTED] skype: bulcrack___ --Bandwidth and Colocation provided by
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-- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128
msn: [EMAIL PROTECTED]skype: bulcrack
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Re: [Asterisk-Users] iax provider
voipject sucks! choppy line ang too much pdd..i use voxee.comOn 1/25/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: You can try with www,iax.cc too but i guesst not luck with a test account.. Dan On 25/01/06, Nilesh Londhe [EMAIL PROTECTED] wrote: I use www.voipjet.com and find it OK.On 1/24/06, Roberto Pereyra [EMAIL PROTECTED] wrote: Hi I looking a good IAX service for a emerging voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte técnico ISPs Jabber ID: [EMAIL PROTECTED] For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Best Regards,Angelito Manansalawww.voicefidelity.net Mobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrack ___ --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] Re: How to disable WARNINGS in CLI
thanks buddyOn 1/23/06, Cameron Grant [EMAIL PROTECTED] wrote: check /etc/asterisk/logger.confregards,cameron -- Forwarded message -- From: Angelito Manansala [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 22 Jan 2006 06:57:05 +0800 Subject: [Asterisk-Users] How to disable WARNINGS in CLI Hi guys, anyone knows how to disable the WARNINGS in cli, i set verbose 0 but the warning still show.. Thanks, Lito___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807DID: (+63) 44 7906770US DID: +1 619 399 0128 msn: [EMAIL PROTECTED]skype: bulcrack ___ --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 view Q.931 Disconnect code
Hi there,Can anyone know how to view asterisk disconnect code.?-- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807 DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrack ___ --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 disable WARNINGS in CLI
Hi guys,anyone knows how to disable the WARNINGS in cli, i set verbose 0 butthe warning still show..Thanks,Lito ___ --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] Call Center and Predictive dialing
Vicidial will works smooth on your 100 agents. but you must focus on what trunk, server codec you will use.On 1/17/06, C F [EMAIL PROTECTED] wrote:I know this question has been asked a lot before, but please I would like to know from personal experience.I'm looking to use Asterisk in a call center environment, where mostof the calls will be outbound calls. They will have at start 100agents.I have looked at vicidial and looks promising, however I would like to hear from users what they use and how they like it compared to otherproducts they have tried.Of interest are:* Ease of use* Stability* Feature set* Open SourceThank You___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807 DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrack ___ --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 connect to voicemaster configuration 1.7
Hi to All,Is anyone here has a settings on VM and asterisk for interconnection via SIP.ThanksLito ___ --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 connect to voicemaster configuration 1.7
what i mean is asterisk will send to Voicemaster then VM to VOIP provider?any configuration example?On 12/31/05, Bill Gibbs [EMAIL PROTECTED] wrote: Voicemaster is a commercial softswitch. www.sysmaster.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Moises Silva Sent: Friday, December 30, 2005 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk connect to voicemaster configuration 1.7 hu? On 12/30/05, Angelito Manansala [EMAIL PROTECTED] wrote: Hi to All, Is anyone here has a settings on VM and asterisk for interconnection via SIP. Thanks Lito ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +639175425807DID: (+63) 44 7906770 msn: [EMAIL PROTECTED]skype: bulcrack ___ --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] Philippines asterisk mailing list / yahoo groups! (PINOY AKO! PINOY TAYO!)
Hello Pinoy asteriskers, This is also an annoucement of a new mailist list for Filipino Asterisk users. please visit - http://groups.yahoo.com/group/asterisk-ph Thanks, Lito ___ --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 callback system
thats right On 11/27/05, C F [EMAIL PROTECTED] wrote: Just fire up vi and start typing. On 11/25/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi list: what are the steps to do to asterisk to be ready fro callback system? __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] Philippines Asterisk users, anyone?
-- Best Regards, Angelito Manansala Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Philippines Asterisk users, anyone?
if you have msn, add me in your list.. maybe i can help in your newbie question ehehe On 11/26/05, John Fraser [EMAIL PROTECTED] wrote: Im one! John, also a new user. cheers John [EMAIL PROTECTED] On Fri, 25 Nov 2005 18:54:02 +0100, Michael Kenjie Nukui wrote Im one! kenjie pre, i am just a new user of asterisk. regards, kenjie nukui [EMAIL PROTECTED] On 11/25/05, Angelito Manansala [EMAIL PROTECTED] wrote: -- Best Regards, Angelito Manansala Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --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] IAX and Firewall
how much traffic do you have to philippines? please contact me offlist On 11/19/05, Joseph [EMAIL PROTECTED] wrote: The problem are almost solved. With regards to Teialx it appears that I was registered to their server co1 that worked in the past (but not now) as I'm suppose to be on co2. So switching back to co2 server solved the problem. VOIPJET - one of their servers was down, so switching the server IP solved the problem. Is it possible to specify two or three alternate server in iax.conf just in case once goes down so the system will try alternate server connection? I've tried connecting to Philippines from both providers and it appear to me something must be wrong with the line to the Philippines, the connection will not go through. -- #Joseph On Fri, 2005-11-18 at 16:01 -0600, Piotr A. Sygula wrote: If teliax ever wants to connect to your asterisk box, as in if they're providing a DID for you, you will need to allow teliax through the firewall. If you're the one originating the connection to them, you don't need to open the ingress port. I don't believe so. By registering with the remote server, you are giving them the NAT port to get back into your server with. All communications will take place on that port. Registration has nothing to do with NAT. The key here is which side initiates the connection. Of course this is all under the assumption that Joseph's firewall is statefull. ** *** Peter A. Sygula President/Chief Security Officer - NetShapers, Inc. http://www.net-shapers.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --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 question
yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hangup detection - TDM400P
hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have disconnect supervision, and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Changing 5060 port
try [general] port=81 :) On 11/16/05, Abdul Lateef [EMAIL PROTECTED] wrote: Hi friends, I want to change the standard 5060 sip port to our any defined port. i made some change in sip.conf but it is not working, I have 2 softphone which are able to register with 81 port but the any kind of hardphone is not able to register using 81 port. here is my sip.conf configuration [general] port=5060 [123456] type=friend username=123456 host=dynamic port=81 ;the hardphone should be register with 81 port context=voip allow=g729 allow=alaw allow=g723.1 Please help me how i can register with 81 port? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to check how many G729 codec license installed
Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to check how many G729 codec license installed
*CLI show g729 No such command 'show g729' (type 'help' for help) this means i have no g729 codec installed, right? On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote: That's easy... Just go into asterisk cli and type show g729 It will tell you how many are active and how many you have in total Regards Zafer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelito Manansala Sent: Sunday, 13 November 2005 10:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to check how many G729 codec license installed Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to check how many G729 codec license installed
sahil, what is the relation of aligator and g729codec? On 11/13/05, Sahil Gupta [EMAIL PROTECTED] wrote: Right :) Regards, Sahil Gupta VoiceValley On Sun, 13 Nov 2005, Angelito Manansala wrote: *CLI show g729 No such command 'show g729' (type 'help' for help) this means i have no g729 codec installed, right? On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote: That's easy... Just go into asterisk cli and type show g729 It will tell you how many are active and how many you have in total Regards Zafer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelito Manansala Sent: Sunday, 13 November 2005 10:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to check how many G729 codec license installed Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to check how many G729 codec licenseinstalled
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 3 3 4 3 2 9 - - 131 ulaw - 5 - 1 3 2 1 8 - - 130 alaw - 5 1 - 3 2 1 8 - - 130 g726 - 6 3 3 - 3 2 9 - - 131 adpcm - 5 2 2 3 - 1 8 - - 130 slin - 4 1 1 2 1 - 7 - - 129 lpc10 - 8 5 5 6 5 4 - - - 133 g729 - - - - - - - - - - - speex - - - - - - - - - - - ilbc - 9 6 6 7 6 512 - - - this means i have no g729 codec installed.. thanks guys! :p On 11/13/05, Gentian Bajraktari [EMAIL PROTECTED] wrote: Do: *CLI show translations If you see - (lines) on the G729 row/columns than you do not have any G729 support. RG. - Original Message - From: Sahil Gupta [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 13, 2005 1:03 PM Subject: Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled Right :) Regards, Sahil Gupta VoiceValley On Sun, 13 Nov 2005, Angelito Manansala wrote: *CLI show g729 No such command 'show g729' (type 'help' for help) this means i have no g729 codec installed, right? On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote: That's easy... Just go into asterisk cli and type show g729 It will tell you how many are active and how many you have in total Regards Zafer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelito Manansala Sent: Sunday, 13 November 2005 10:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to check how many G729 codec license installed Guys, is the any CLI commands or info files where you can check how many g729 codec license installed. Regards, Lito ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Consultant
Im willing to help for free contact me via msn messenger my id is [EMAIL PROTECTED] On 11/11/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: I have been posting this issue for over a month and I am not irritated. I appreciate all the users who helped before and I thank in advance any offer for help including reasonable paid time. You may sent me your price me list at [EMAIL PROTECTED] Regards; chawki --- Rob Lith [EMAIL PROTECTED] wrote: Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :) Rob On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote: The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 http://213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.comhttp://Easynews.com -- 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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Nextone - Asterisk - DID provider
i dont know if this will work but also try:
exten = _999NXXNXX,1,Dial(SIP/username:[EMAIL PROTECTED]/${EXTEN})
//
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
I have created a bridge between my nextone softswitch and my DID
provider using Asterisk 1.0.9.
Asterisk is registered on both side and I can receive incoming from
my DID provider and pass it on to Nextone and terminate the call.
Now I want to send the call to my DID provider from nextone via
Asterisk. I did all the routings on the nextone box and I do see the
call coming to Asterisk box when I do sip debug, but the calls
fails with fast busy tone.
sip.conf
register = user:pass:[EMAIL PROTECTED] DID provider
register = user:[EMAIL PROTECTED] Nextone box
extension.conf
[default]
exten = _999NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]:5060) //
When I receive the call from my DID provider I will send it to my
nextone box.
Now how can I send back to my DID provider?
Thanks,
Neal
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Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Re: [Asterisk-Users] Nextone - Asterisk - DID provider
can you send the call flow
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks for your reply,
I can send the call from my DID provider to my nextone via
Asterisk... Thats OK.
But I am failing to send the call from Nextone to my DID provider via
Asterisk... Fail.
exten = _999NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
The above exten works fine... it sends the call to nextone from DID
provider.
I need exten parameter for sending call back to DID provider from
Nextone...
Thanks,
Neal
On Nov 11, 2005, at 8:21 PM, Angelito Manansala wrote:
i dont know if this will work but also try:
exten = _999NXXNXX,1,Dial(SIP/
username:[EMAIL PROTECTED]/${EXTEN})
//
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
I have created a bridge between my nextone softswitch and my DID
provider using Asterisk 1.0.9.
Asterisk is registered on both side and I can receive incoming from
my DID provider and pass it on to Nextone and terminate the call.
Now I want to send the call to my DID provider from nextone via
Asterisk. I did all the routings on the nextone box and I do see the
call coming to Asterisk box when I do sip debug, but the calls
fails with fast busy tone.
sip.conf
register = user:pass:[EMAIL PROTECTED] DID provider
register = user:[EMAIL PROTECTED] Nextone box
extension.conf
[default]
exten = _999NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]:5060) //
When I receive the call from my DID provider I will send it to my
nextone box.
Now how can I send back to my DID provider?
Thanks,
Neal
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Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
2070 Business Center Drive, Suite 210
Irvine, California 92612
Tel: 1-949-851-8737
Fax: 1-949-851-8011
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Re: [Asterisk-Users] Nextone - Asterisk - DID provider
did you put register statement in your asterisk pointed to your did provider?
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
OK,
1) When external user dials the DID number, it hits the DID providers
switch.
2) DID provider forwards the external user to Asterisk.
3) Asterisk sends the call to Nextone using exten = _999NXXNXX,
1,Dial(SIP/[EMAIL PROTECTED]) statement which is in
extension.conf. Call is send to nextone with 999 prefix for billing
purpose.
Now I want to send a Call from Nextone to Asterisk and then Asterisk
forwards that call to DID provider.
1) Nextone sends a call with Prefix 9998887+Number to Asterisk.
2) Asterisk will strip the prefix 9998887 and sends the call to DID
provider for termination.
When I do sip debug on my CLI, I see Nextone sends the call with
prefix but Asterisk fails to receive.
In which context do I specify for outbound calls? For example exten
= _9998887NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060)
Hope this helps
Thanks,
Neal
On Nov 11, 2005, at 9:55 PM, Angelito Manansala wrote:
can you send the call flow
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
Thanks for your reply,
I can send the call from my DID provider to my nextone via
Asterisk... Thats OK.
But I am failing to send the call from Nextone to my DID provider via
Asterisk... Fail.
exten = _999NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
The above exten works fine... it sends the call to nextone from DID
provider.
I need exten parameter for sending call back to DID provider from
Nextone...
Thanks,
Neal
On Nov 11, 2005, at 8:21 PM, Angelito Manansala wrote:
i dont know if this will work but also try:
exten = _999NXXNXX,1,Dial(SIP/
username:[EMAIL PROTECTED]/${EXTEN})
//
On 11/12/05, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
I have created a bridge between my nextone softswitch and my DID
provider using Asterisk 1.0.9.
Asterisk is registered on both side and I can receive incoming from
my DID provider and pass it on to Nextone and terminate the call.
Now I want to send the call to my DID provider from nextone via
Asterisk. I did all the routings on the nextone box and I do see
the
call coming to Asterisk box when I do sip debug, but the calls
fails with fast busy tone.
sip.conf
register = user:pass:[EMAIL PROTECTED] DID provider
register = user:[EMAIL PROTECTED] Nextone box
extension.conf
[default]
exten = _999NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]:
5060) //
When I receive the call from my DID provider I will send it to my
nextone box.
Now how can I send back to my DID provider?
Thanks,
Neal
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Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
2070 Business Center Drive, Suite 210
Irvine, California 92612
Tel: 1-949-851-8737
Fax: 1-949-851-8011
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
___
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
2070 Business Center Drive, Suite 210
Irvine, California 92612
Tel: 1-949-851-8737
Fax: 1-949-851-8011
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED
Re: [Asterisk-Users] ITS Telecom Hardware
how much is that per pc.? On 11/10/05, Pete Barnwell [EMAIL PROTECTED] wrote: Hi, Has anybody tried using ITS Telecom Analog::GSM gateway devices with * ? http://www.its-tel.com/main/home/doc.asp?mCatID=1977mCatPID=1972tpMID=0 They appear to be very favourably priced... Rgds Pete ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Consultant
i can fix that please contact me off list, i have setup now that same as yours and i encountered that problem. On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote: The only pointer I got is a $50/hr Mark phillip offered. I can make VOIP calls between my Asterisk server and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time The call get accepted, but it seems there is no acknowledgement from my server to receive the call from the provider. Thanks; --- Mark Phillips [EMAIL PROTECTED] wrote: He did. And he got pointers to the relevant howto's. Matt Riddell wrote: chawki hammoud wrote: Hi: I posted my problem several times about being unable to make IAX calls from my Asterisk box to another IAX server without luck. So, what's your problem? Post some details. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] codecs
i think gsm you mention is gsm sound files not gsm codecs. On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote: Hi all, We use asterisk as a local pbx and we connect to a pstn/sip provider for calls to pstn. Since the messages on asterisk are on gsm format, we need gsm, but to call pstn, we need g729 or g723. How can we enable both codecs to be able to call pstn and hearing voicemail messages for example? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] how to setup Agent dialing in multiple asterisk servers
not so complicated.. use IAX trunking to share dialplans On 11/9/05, KRTorio [EMAIL PROTECTED] wrote: Our Setup: In our company we run multiple asterisk servers, and agents login (using AgentCallbackLogin) to any of these. One person, one agent number ID. The Problem: Dialing an agent number from within one pbx is easy, but if you want to dial an agent logged in another pbx, its more complicated. Our current dialplan performs guessing which pbx the agent is logged, by dialing all of them. We have to redesign this everytime we add another pbx, and we're looking for a more efficient method of doing this. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can't make calls from Asterisk IAX to other IAX
can you paste you iax.conf On 11/8/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi: I have been having this problem for sometime that I am not able to solve and I hope someone can help. I can make VOIP calls between my Asterisk box and my VOIP provider using sip channel without a problem. But when I attempt to make a call using IAX, the call get accepted and then get a hangup message: This is the message I get when I attempt to make an IAX call: Executing Dial(OSS/dsp, IAX2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 -- Call accepted by 213.61.187.150 (format gsm) -- Format for call is gsm -- Hungup 'IAX2/callshopcompany/1' == No one is available to answeer at this time __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to dial direclty from PBX extension to IP phone
try this
exten = _111XXX,1,Dial(SIP/${EXTEN:3},30)
On 11/4/05, Tharanga [EMAIL PROTECTED] wrote:
Hi All,
Iam having a asterisk PBX with 2 TDM04B cards. one card use to dial out side
(PSTN) and other card is used to dial office normal PBX extensions. now i
have a IVR. and if my users (normal pbx user) need to dial to IP extension.
he has to to dial a specific extension and at the IVR prompt need to dial
SIP extension. but this is time consuming...
as a example say i have a SIP extension 123. and users at normal PBX can
reach the asterisk IVR by pressing 111. IS IT POSSIBLE TO DIAL 23
direcly to sip extension from normal PBX extension. if so..how to do
that
really appriciate if some one can guide me..
thanks in advance
Tharanga
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Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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