Re: [asterisk-users] Ebay's SIP for Skype
El mié, 25-03-2009 a las 10:28 -0700, Michael Robertson escribió: OpenSky can be setup for free to allow any Asterisk system to call Skype users. Setup instructions for Asterisk are at: http://www.gizmo5.com/opensky Free calls are available up to 5 minutes. If you need longer calls there's a commercial service you can purchase. Can be used to receive calls from skype? Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: gsa...@manta.telconet.net www : http://www.telconet.net SIP : 6...@sip.manta.telconet.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay's SIP for Skype
El mié, 25-03-2009 a las 19:09 +0100, Administrator TOOTAI escribió: Can be used to receive calls from skype? Yes Great,and how? Have you any link to read? Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: gsa...@manta.telconet.net www : http://www.telconet.net SIP : 6...@sip.manta.telconet.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk h323 module
El vie, 05-12-2008 a las 19:04 +0300, Mikhail Zhirnov escribió: make[2]: cc: Command not found Looks like you need cc installed. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wellgate Asterisk
El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): I've one wellgate 3804 (old version) with 4 fxo ports integrated with asterisk 1.4. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk For Windows ?
El sáb, 11-10-2008 a las 11:07 -0700, Roderick A. Anderson escribió: A quick search using Google gave me http://live.gnome.org/Orca Sound isn't working right now on my workstation so I can't test it but it is installed by default on my CentOS 5 workstation. I've installed it on my laptop running debian sid and works pretty good. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tribox
El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió: Hi triXbox.org can answer these questions. Google may also give a balanced view. But yes, i can assure you, people are using Trixbox from Fonality. Steve On 6 Oct 2008, at 10:24, broadband Voice wrote: Anyone using Tribox from Fonality. I understand its open source and free. Can I use it for a call center functionality? Thanks. Give a try to elastix [1], it haves a very complete callcenter module. [1] www.elastix.org Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tribox
El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió: Hi triXbox.org can answer these questions. Google may also give a balanced view. But yes, i can assure you, people are using Trixbox from Fonality. Steve On 6 Oct 2008, at 10:24, broadband Voice wrote: Anyone using Tribox from Fonality. I understand its open source and free. Can I use it for a call center functionality? Thanks. Give a try to elastix [1], it haves a very complete callcenter module. [1] www.elastix.org Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tribox
El lun, 06-10-2008 a las 13:23 -0700, Ron Stephan escribió: And the documentation (not that trixbox is well documented ) was weak IMHO. Try reading: http://www.elastixconnection.com/downloads/elastix_without_tears.pdf Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk linkedin group
El jue, 28-08-2008 a las 10:32 -0400, BerkHolz, Steven escribió: asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank you, I've joined it. There is a group for spanish users for anyone interested: http://www.linkedin.com/groups?gid=9 Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 protocol
El jue, 28-08-2008 a las 01:32 -0700, mahboob zaman escribió: hi. i have two IP phones that are in H323 protocol. How can i test that these two phones are working? For IP phone (SIP) i used asterisk server. can i use asterisk server to test the ip phone with H323 protocol. I've wrote a small guide to enable chan_h323.so on asterisk 1.4 (is in spanish, sorry): http://www.ecualug.org/?q=2008/04/18/comos/asterisk_14_agregando_soporte_h323_chan_h323so_en_asterisk_14 Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Card in OFFHOOK state
El mar, 26-08-2008 a las 19:46 -0700, Jay Ray escribió: Any pointers on this one? --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote: From: Jay Ray [EMAIL PROTECTED] Subject: [asterisk-users] X100P Card in OFFHOOK state To: asterisk-users@lists.digium.com Date: Tuesday, August 26, 2008, 12:24 PM After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook state as shown here... Signalling Type: FXS Kewlstart Radio: 0re2uk*CLI Owner: None*CLI Real: Nonek*CLI Callwait: NoneI Threeway: NoneI Confno: -12uk*CLI Propagated Conference: -1 Real in conference: 0 DSP: noore2uk*CLI Relax DTMF: noCLI Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: noLI Pulse phone: noLI Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook -- Sometimes it still takes a new call while in this state and sometimes rejects it... How to correct it such that after I hangup a call it goes back to onhook state... reloading wcfxo module using modprobe clears the issue Sounds like your card is not detecting the busy tone, try adding the following line at your zapata.conf file: busydetect=yes busycount=6 Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió: Has anyone have experiencies on this kind of scenario... what version?.. patches?... or any information regarding this goal will be VERY helpful... Hi Arturo, Please ckeck the following URL (on spanish): http://www.sinologic.net/2008-07/como-configurar-un-fax-virtual-t38-en-asterisk/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disto choice for Asterisk with AVM Fritz!PCI cards
El mar, 01-07-2008 a las 06:10 +0200, Dave Cotton escribió: I go along with the above, Ive done this with Mandrake 10.1 and OpenSuse 10.1 and 10.3, What I found was that with the Mandrake I used chan_capi and patched the Suse supplied driver code to work with 2 Frtz cards a lot of work. With the Suse installs I switched to chan_misdn no patching and the config was handled bu misdn_init config automagically. I'me really happy with debian. Always you can use apt-get to install asterisk and modules without pain ;) Latest zaptel modules from debian repository have OSLEC as default echo canceler and works like a charm. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
El mar, 01-07-2008 a las 10:35 -0300, Gustavo A Gonzalez escribió: Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can´t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. I've the same setup with FreePBX and NVFaxDetect. Works fine. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Nextone using H323
El mar, 24-06-2008 a las 12:20 -0300, Everton Goularth escribió: I`m using chan_ooh323 in my asterisk server. This is my ooh323.conf: Have you tried with chan_h323.so? I've one gateways that uses h.323 and works only with chan_h323.so . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
Hi, I'm installing zaptel-source_1.4.10.1~dfsg-1_all.deb (from Debian SID) into my ubuntu 8.04 box with: dpkg -i zaptel-source_1.4.10.1~dfsg-1_all.deb ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel Loading the wcfxo module and/or zaptel: [EMAIL PROTECTED]:~# modprobe wcfxo WARNING: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting wcfxo (/lib/modules/2.6.24-16-server/misc/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) [EMAIL PROTECTED]:~# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) When the build is finished and the box restarted, I'm getting this dmesg output: [ 46.160498] zaptel: Unknown symbol oslec_echo_can_identify [ 46.180909] ztdummy: Unknown symbol zt_receive [ 46.181054] ztdummy: Unknown symbol zt_transmit [ 46.181126] ztdummy: Unknown symbol zt_unregister [ 46.181221] ztdummy: Unknown symbol zt_register [ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify [ 830.122738] wcfxo: Unknown symbol zt_receive [ 830.122890] wcfxo: Unknown symbol zt_ec_chunk [ 830.123037] wcfxo: Unknown symbol zt_transmit [ 830.123112] wcfxo: Unknown symbol zt_unregister [ 830.123212] wcfxo: Unknown symbol zt_hooksig [ 830.123301] wcfxo: Unknown symbol zt_register [ 830.123377] wcfxo: Unknown symbol zt_alarm_notify [ 858.887084] zaptel: Unknown symbol oslec_echo_can_identify [EMAIL PROTECTED]:~# This is the modinfo output: [EMAIL PROTECTED]:~# modinfo zaptel filename: /lib/modules/2.6.24-16-server/misc/zaptel.ko version:1.4.10.1 license:GPL description:Zapata Telephony Interface author: Mark Spencer [EMAIL PROTECTED] srcversion: 927BA7DCB504C0BA7C0CDED depends:oslec,crc-ccitt vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: deftaps:int [EMAIL PROTECTED]:~# modinfo wcfxo filename: /lib/modules/2.6.24-16-server/misc/wcfxo.ko license:GPL author: Mark Spencer [EMAIL PROTECTED] description:Wildcard X100P Zaptel Driver srcversion: 194D48A51D46F480234E26A alias: pci:v1057d5608sv*sd*bc*sc*i* alias: pci:vE159d0001sv8087sd*bc*sc*i* alias: pci:vE159d0001sv8086sd*bc*sc*i* alias: pci:vE159d0001sv8085sd*bc*sc*i* alias: pci:vE159d0001sv8084sd*bc*sc*i* depends:zaptel vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: quiet:int parm: boost:int parm: monitor:int parm: opermode:int [EMAIL PROTECTED]:~# modinfo oslec filename: /lib/modules/2.6.24-16-server/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 9C9E87427F162644A61A1CB depends: vermagic: 2.6.24-16-server SMP mod_unload 686 I've the same trouble installing from sources and patching the zaptel sources with oslec. I've installed before on debian sarge/etch and ubuntu 7.10 without problems. What can be wrong? Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
El mié, 18-06-2008 a las 01:37 +0300, Tzafrir Cohen escribió: That's a strange place. Is there /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko ? find /lib/modules/2.6.24-16-server/ -name oslec.ko I suspect there's an older and incompatible copy of oslec.ko around. You are right: find /lib/modules/2.6.24-16-server/ -name oslec.ko /lib/modules/2.6.24-16-server/oslec.ko /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko I will be deleting all oslec.ko references, modules/zaptel directory and start again. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
El mar, 17-06-2008 a las 18:01 -0500, Guillermo Salas M. escribió: find /lib/modules/2.6.24-16-server/ -name oslec.ko /lib/modules/2.6.24-16-server/oslec.ko /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko I will be deleting all oslec.ko references, modules/zaptel directory and start again. It works :) [EMAIL PROTECTED]:~# lsmod | grep oslec oslec 10396 1 zaptel [EMAIL PROTECTED]:~# lsmod | grep zaptel zaptel195588 6 wcfxo,wcopenpci oslec 10396 1 zaptel crc_ccitt 3072 2 zaptel,hisax Thank you very much for your help. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk video alternatives
El vie, 06-06-2008 a las 00:24 +0200, Matias Surdi escribió: At the company I work for, we use Asterisk to communicate with our offices all around the world. Recently, I've been asked to implement a video conference system, asterisk compatible/interoperable as possible. It's preferred but not required to be an open source solution. Try vmukti http://sourceforge.net/projects/vmukti/ VMukti is leading Asterisk/ Yate enabled web video conferencing application for Web / PSTN. It is world’s first open source mashable PBX and meeting platform for home and office having features like multipoint audio/ video, desktop sharing, whiteboard. What options do I have? wich would you suggest me to try? Any good experience with any of these systems? I've no tested it before, please let us know your experience using it. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Click-to-talk (Java application)
On Mon, 2008-04-21 at 10:31 -0300, equis software wrote: I need to implement click-to-talk web application.(not click-to-call or callback) I try to use njiax, and iaxclient but I can´t made it work. Has anybody other solution?? You can try with jiax: http://www.hem.za.org/jiaxclient/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On Sat, 2008-02-23 at 07:52 +0200, Yehavi Bourvine +972-8-9489444 wrote: The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Yes, is true. You must have to disable the automatic wireless LAN scan option. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On Sat, 2008-02-23 at 00:03 +0530, amit salunkhe wrote: i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web also from Nokia site but they only mention this features as VOIP call from wifi they mentioed only this much info. they not mentioed info about inbulit SIP client to make voip calls without download any third party software. as per my search i found this 3 phones E51,E61i,E65 I've one nokia E65 that works very well with my asterisk box. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Fax
On Tue, 2008-01-01 at 13:48 -0600, Jonn R Taylor wrote: REALY?? Humm I have been doing this for over a year and we receive over 400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo and no voice quality issues. Now we do have a 12mb down 768k up connection. Can you share more details about your implementation? what are you using for faxing? Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
El Mie, 5 de Diciembre de 2007, 11:45, Michael Melia Jr. escribió: Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Try OutCall: http://outcall.sourceforge.net/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free T1 Card?
On Mon, 2007-11-05 at 10:14 -0800, Michael Collins wrote: I recall several months ago that there was a company that was giving away a free 1-port T1 card, with some specific conditions. Do any of you recall who that was? My Google searches are coming up empty and now I’m wondering if I was hallucinating… They sent to me one PIKA inlineMM with 4 FXO ports. Works great. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
On Thu, 2007-09-20 at 14:23 -0400, Mike Clark wrote: Matthew Rubenstein wrote: Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. Mexuar's Coraletta is nice, but isn't GPL. http://www.mexuar.com/products_sdk.shtml I'm using JIAXClient [1], it is GPL, uses IAX2 and works pretty excelent with gsm codec. [1] http://www.hem.za.org/jiaxclient/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank and caller ID from PSTN
On Tue, 2007-09-18 at 19:33 -0500, Guillermo Salas M. wrote:
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
[..]
One thing I suspect is not waiting enough.
Try adding the following to your dialplan:
[pstn-test]
exten = s,1,Wait(1)
exten = s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten = s,n,Playback(tt-monkeys)
exten = s,n,Goto(from-zaptel,s,1)
And then set in zapata.conf: context=pstn-test
Other than that, there are two obvious sanity checks:
1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected
I've connected one phone to the line that was connected on the port 4 of
the astribank. Called from my mobile and the caller id is displayed on
the phone.
2. boot the same system from our live CD and see if caller ID is
detected there.
Booted with the version Xorcom Rapid LiveCD (1.0.2.4131) and configured
the following:
- Created one trunk called g1;
- creted one SIP extension called 666 ;
- edited /etc/asterisk/zapata-channels.conf with:
;;; line=4 XPP_FXO/00/00/3 (no pcm)
signalling=fxs_ks
callerid=asreceived
group=1
context=from-zaptel
channel = 4
context=default
- created one incoming route with freebpx,
- all the calls that are coming on the port 4 of the astribank will be
redirected to the sip extension 666;
Now, dialing from my mobile phone again to the line connected to the
port 4 of the astribank is showing me the called ID on the 666 extension
as Unknown:
[..]
Please check the Zaptel hardware listing from the live cd:
http://pastebin.ca/702694
Regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 985 5138
e-mail : [EMAIL PROTECTED]
www : http://www.manta.telconet.net
http://www.telcocarrier.net
SIP : [EMAIL PROTECTED]
FWD : 558563
Linux User: 255902
Beat me, whip me, make me use Windows!
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Astribank and caller ID from PSTN
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
[..]
One thing I suspect is not waiting enough.
Try adding the following to your dialplan:
[pstn-test]
exten = s,1,Wait(1)
exten = s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten = s,n,Playback(tt-monkeys)
exten = s,n,Goto(from-zaptel,s,1)
And then set in zapata.conf: context=pstn-test
Other than that, there are two obvious sanity checks:
1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected
I've connected one phone to the line that was connected on the port 4 of
the astribank. Called from my mobile and the caller id is displayed on
the phone.
2. boot the same system from our live CD and see if caller ID is
detected there.
Booted with the version Xorcom Rapid LiveCD (1.0.2.4131) and configured
the following:
- Created one trunk called g1;
- creted one SIP extension called 666 ;
- edited /etc/asterisk/zapata-channels.conf with:
;;; line=4 XPP_FXO/00/00/3 (no pcm)
signalling=fxs_ks
callerid=asreceived
group=1
context=from-zaptel
channel = 4
context=default
- created one incoming route with freebpx,
- all the calls that are coming on the port 4 of the astribank will be
redirected to the sip extension 666;
Now, dialing from my mobile phone again to the line connected to the
port 4 of the astribank is showing me the called ID on the 666 extension
as Unknown:
-- Starting simple switch on 'Zap/4-1'
-- Executing NoOp(Zap/4-1, Entering from-zaptel with DID == ) in
new stack
-- Executing Ringing(Zap/4-1, ) in new stack
-- Executing Set(Zap/4-1, DID=s) in new stack
-- Executing NoOp(Zap/4-1, DID is now s) in new stack
-- Executing GotoIf(Zap/4-1, 1?zapok:notzap) in new stack
-- Goto (from-zaptel,s,8)
-- Executing NoOp(Zap/4-1, Is a Zaptel Channel) in new stack
-- Executing Set(Zap/4-1, CHAN=4-1) in new stack
-- Executing Set(Zap/4-1, CHAN=4) in new stack
-- Executing Macro(Zap/4-1, from-zaptel-4|s|1) in new stack
-- Executing NoOp(Zap/4-1, Entering macro-from-zaptel-4 with DID
= s) in new stack
-- Executing Gosub(Zap/4-1, app-blacklist-check|s|1) in new
stack
-- Executing LookupBlacklist(Zap/4-1, ) in new stack
-- Executing GotoIf(Zap/4-1, 0?blacklisted) in new stack
-- Executing Return(Zap/4-1, ) in new stack
-- Executing Set(Zap/4-1, __FROM_DID=s) in new stack
-- Executing Goto(Zap/4-1, ext-local|666|1) in new stack
-- Goto (ext-local,666,1)
== Channel 'Zap/4-1' jumping out of macro 'from-zaptel-4'
-- Executing Macro(Zap/4-1, exten-vm|666|666) in new stack
-- Executing Macro(Zap/4-1, user-callerid) in new stack
-- Executing NoOp(Zap/4-1, user-callerid: ) in new stack
-- Executing GotoIf(Zap/4-1, 0?report) in new stack
-- Executing GotoIf(Zap/4-1, 0?start) in new stack
-- Executing Set(Zap/4-1, REALCALLERIDNUM=) in new stack
-- Executing NoOp(Zap/4-1, REALCALLERIDNUM is ) in new stack
-- Executing Set(Zap/4-1, AMPUSER=) in new stack
-- Executing Set(Zap/4-1, AMPUSERCIDNAME=) in new stack
-- Executing GotoIf(Zap/4-1, 1?report) in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp(Zap/4-1, TTL: ARG1: 666) in new stack
-- Executing GotoIf(Zap/4-1, 0?continue) in new stack
-- Executing Set(Zap/4-1, __TTL=64) in new stack
-- Executing GotoIf(Zap/4-1, 1?continue) in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp(Zap/4-1, Using CallerID ) in new stack
-- Executing Set(Zap/4-1, FROMCONTEXT=exten-vm) in new stack
-- Executing Set(Zap/4-1, VMBOX=666) in new stack
-- Executing Set(Zap/4-1, EXTTOCALL=666) in new stack
-- Executing Set(Zap/4-1, CFUEXT=) in new stack
-- Executing Set(Zap/4-1, CFBEXT=) in new stack
-- Executing Set(Zap/4-1, RT=15) in new stack
-- Executing Macro(Zap/4-1, record-enable|666|IN) in new stack
-- Executing GotoIf(Zap/4-1, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(Zap/4-1, recordingcheck|20070919-002336|
asterisk-5150-1190161411.3) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck|20070919-002336|asterisk-5150-1190161411.3: Inbound
recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(Zap/4-1, No recording needed) in new stack
-- Executing Macro(Zap/4-1, dial|15|tr|666) in new stack
-- Executing DeadAGI(Zap/4-1, dialparties.agi) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf
Re: [asterisk-users] External FXO port.
On Sat, 2007-09-15 at 19:25 +0530, Sanspareils Greenlans wrote: Sir, I have an audiocode MP-118 8 port external FXO gateway and i have connect pstn line to FXO gateway now i want to dial outside call using FXO gateway and receive all outside call. but i donot know what i have add in sip.conf and extension.conf to make it possible. I have also attach digium TDM02b card on asterisk server and all incoming and outgoing call going perfectly. but not sure how to define call receive or dial through external FXO gateway. Please give me information how we can do that. You must have to create the config for the device on sip.conf, check the following link, may can help you: http://www.trixbox.org/forums/trixbox-forums/share-your-trixbox-success-stories/mp-118-and-trixbox-integration-success Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astribank and caller ID from PSTN
Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell ;cidstart=ring hidecallerid=no callwaiting=yes ;usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes ;callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 ;immediate=no callerid=asreceived ;amaflags=default busydetect=yes busycount=8 ;busypattern=500,500 answeronpolarityswitch=no hanguponpolarityswitch=no faxdetect=both ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: FXO ;;; line=1 XPP_FXO/0/0/0 FXSKS signalling=fxs_ks callerid=asreceived group=1 context=from-zaptel channel = 1 When replacing callerid=phone-number I get on my ipphone phone-number as callerid: ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: FXO ;;; line=1 XPP_FXO/0/0/0 FXSKS signalling=fxs_ks callerid=2627839 group=1 context=from-zaptel channel = 1 Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astribank and caller ID from PSTN
Hi Tzafrir:
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
[..]
One thing I suspect is not waiting enough.
Try adding the following to your dialplan:
[pstn-test]
exten = s,1,Wait(1)
exten = s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten = s,n,Playback(tt-monkeys)
exten = s,n,Goto(from-zaptel,s,1)
I've added and used the pstn-test with the channel 4, before I've
tested it connecting a phone and calling to the channel4 number from my
cell, the phone shows me the number of my mobile.
And then set in zapata.conf: context=pstn-test
Done, this is the output of the log when I've one incoming call to the
channel4:
[Sep 15 14:39:13] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:1] Wait(Zap/4-1, 1) in new stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:2] NoOp(Zap/4-1, Got number on Zap/4-1) in new
stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:3] Goto(Zap/4-1, from-zaptel|s|1) in new stack
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Goto (from-zaptel,s,1)
[Sep 15 14:39:14] VERBOSE[25265] logger.c: -- Executing
[EMAIL PROTECTED]:1] NoOp(Zap/4-1, Entering from-zaptel with DID == )
in new stack
Other than that, there are two obvious sanity checks:
1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected
Done, the phone is detecting the ID.
2. boot the same system from our live CD and see if caller ID is
detected there.
I'm going to download the live CD from www.xorcom.com .
Thank you for your suggestions,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 985 5138
e-mail : [EMAIL PROTECTED]
www : http://www.manta.telconet.net
http://www.telcocarrier.net
SIP : [EMAIL PROTECTED]
FWD : 558563
Linux User: 255902
Beat me, whip me, make me use Windows!
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Please avoid the Top Posting, see
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Re: [asterisk-users] special kind of billing
On Sun, 2007-09-09 at 02:44 -0700, bilal ghayyad wrote: Dear Guillermo; Is there an english link that help me in configuration other than: http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos Check the www.asterisk2billing.org documentation page. Also, what about ASTCC? I've not used it yet. Another issue: a2billing support prepaid billing (so it can be used for calling cards)? Yes. Check the features list at http://trac.asterisk2billing.org/cgi-bin/trac.cgi . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special kind of billing
On Wed, 2007-09-05 at 22:44 +0600, Kate Kretz wrote: Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on other side). is there any billing for asterisk which can do that ? Yes, We are using a2billing [1]. You can define serveral trunks and add rates for the destinations, the a2billing can use low cost routing and gives to you a detailed call detail record with the ammount of sell, buy, profit, margin and markup. You can learn to use with this small guide (spanish): http://www.ecualug.org/?q=2006/12/12/comos/configurar_a2billing_en_menos_de_10_minutos [1] www.asterisk2billing.org Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Experimenting- Sip dialing with Zap
On Thu, 2007-08-16 at 16:23 +, John Meksavan wrote:
exten = _XXX,1,Dial({Zap/g0/{EXTEN:1})
Must be:
exten = _XXX,1,Dial(Zap/g0/{EXTEN:1})
Regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 985 5138
e-mail : [EMAIL PROTECTED]
www : http://www.manta.telconet.net
http://www.telcocarrier.net
SIP : [EMAIL PROTECTED]
FWD : 558563
Linux User: 255902
Beat me, whip me, make me use Windows!
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting
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Re: [asterisk-users] ATA phones ring when they register
On Mon, 2007-08-06 at 17:46 +0200, Mr Shunz wrote: we had the same problem and we came to this solution: go under profile settings and set Caller ID Scheme as ETSI-FSK Prior to Ringing with DTAS... best regards I'm experiencing the same issue with linksys pap2. Any knows how to stop the ringing when the ATA registers with my asterisk box. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retail DID provider ?
On Thu, 2007-08-02 at 01:12 +0530, Mail list wrote: I am looking for a retail DID provider which should provide unlimited incoming calls something around 4-5 bucks . Nufone seemed like a good choice at $5 but they are down again :( . Any suggestions please . I'm using www.les.net . Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best softphone work with Asterisk
On Wed, 2007-07-25 at 09:22 -0700, Jaswinder Singh wrote: Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime . However there is more variety in sip softphones . I think zoiper is much better than other iax2 softphones . I like firefly, it can support g729 for free and SIP/IAX2 protocols. Look at the list archives, there is one URL where you can download both, the firefly sofphone and the g729 codec. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone with g729 codec
On Wed, 2007-07-11 at 21:57 -0600, Al lists wrote: Nice! This version supports IAX2 and SIP. Windows users will be happy using it ;) Regards, You can use the older version of firefly that supports IAX2/SIP protocols and g729 codec. Get the sofhophone and codec from: http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/firefly-thirdparty.exe http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/g729.zip To enable the g729: 1.- Install firefly-thirdparty.exe; 2.- close firefly program; 3.- extract g729.dll from g729.sip to c:/program files/firefly; 4.- start firefly, setup a new account and enable the g729 check box; -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] softphone with g729 codec
On Wed, 2007-07-11 at 08:32 -0400, Maximo Villamayor wrote: you can prove this www.portsip.com You can use the older version of firefly that supports IAX2/SIP protocols and g729 codec. Get the sofhophone and codec from: http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/firefly-thirdparty.exe http://razametal.is-a-geek.org/asterisk/Softphones/Windows/firefly/g729.zip To enable the g729: 1.- Install firefly-thirdparty.exe; 2.- close firefly program; 3.- extract g729.dll from g729.sip to c:/program files/firefly; 4.- start firefly, setup a new account and enable the g729 check box; Regards, Gordon Henderson wrote: On Mon, 2 Jul 2007, jonny hashem wrote: Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? eyeBeam - the commercial version of X-Lite: http://www.counterpath.com/index.php?menu=Productssmenu=eyeBeam Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Edit ulaw file
On Tue, 2007-07-10 at 10:24 -0400, Gary Chen wrote: I recorded some sound files using Asterisk record() app as ulaw file. I need to edit these sound files. What kind of audio editor can I use to edit these files? You can use audacity, works on GNU/Linux and windows and is free software (free as in freedom). Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Que on A2Billing
On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote: Thanks man... So far everything worked as expected... How can I make internal calls stay within the PBX. For example, when one SIP-Friend tries to call another SIP-Friend without sending the call out on Trunk and receive it back. Same like dialing from one extension number to another extension. My SIP-Friends are using US DID numbers and I would like to keep the local calls within the network. Right now when I try to call other SIP-Friend, I get a message saying The number you have dialer is currently not available... while the SIP-Friend is registered. Try dialing the number 9 before the sip/iax2 friend number. Regards, Cheers, Nitesh -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Upgrade cisco SIP phone 7940
On Sun, 2007-06-17 at 13:45 +0100, Adrian Marsh wrote: According to the procedures, I should be able to upgrade, but once the phones loaded and reboots it says it downgrades again and reboots, then the cycle starts again. Try disabling all the tftp boxes on the cisco IP Phone except the tftp used to upgrade to SIP. Maybe you have any other tftp config that downloads another firmware. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Que on A2Billing
On Fri, 2007-06-15 at 12:19 -0400, Nitesh Divecha wrote: Thanks everyone, OK, I got everything working... I manage to create a SIP Customer with a real DID number and configured an ATA with the DID number. ATA can login and can make calls out without any issues. But incoming calls are failing... As soon as the call hits Asterisk, A2Billing script runs and ask for PIN Number... I checked the context for my DID it shows context=a2billing and under sip.conf context=a2billing. If I change the default context under sip.conf to context=default, then the calls are failing... meaning I do not get any response back, but on *CLI debug show that its failing to look for the DID number. Well, I know this is due to my DID is in context=a2billing. Anyone can suggest how can I fix this... I want to ring my incoming to that ATA which has DID assigned. You need to setup the DID on the DID section of a2billing. First create one SIP/IAX2 configuration for your DID provider and assign the context a2billing-did. Later on the DID section, add the DID Provider, add the DID number and asign one destination to the DID (your ata card number) or any SIP extension enabling the voip call radius button. Try it. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Que on A2Billing
On Fri, 2007-06-15 at 14:20 -0400, Nitesh Divecha wrote:
You said change the context for SIP Customers to
context=a2billing-did, do I have to create this context or
A2Billing
will generate by itself?
The a2billing package comes with some examples, you must have to create
the a2billing-did context :
[a2billing-did]
exten = _X.,1,NoOp,${CALLERID(all)}
exten = _X.,2,DeadAGI(a2billing.php|1|did)
exten = _X.,3,Hangup()
This will be the context for your DID provider and not for your
customers.
Check this link for more information:
http://forum.asterisk2billing.org/viewtopic.php?t=1784
Cheers!
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 985 5138
e-mail : [EMAIL PROTECTED]
www : http://www.manta.telconet.net
http://www.telcocarrier.net
Linux User: 255902
Beat me, whip me, make me use Windows!
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting
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Re: [asterisk-users] Que on A2Billing
On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote: Thanks man... That really helped me to move couple of steps. Now I see the incoming calls are going in proper direction... I know I am still missing a small piece here... I did ADD the Destination as a SIP/2486543210, assigned the card number, enabled VOIP_CALL, and enabled Active. 2486543210 is your card number? When I dial the DID number, on the *CLI it shows the following: - a2billing.php|1|did: bug -- AGI Script Executing Application: (DIAL) Options: (SIP/2486543210|60|HL(360:61000:3)) -- Limit Data for this call: -- - timelimit = 360 -- - play_warning = 61000 -- - play_to_caller= yes -- - play_to_callee= no -- - warning_freq = 3 -- - start_sound = UNDEF -- - warning_sound = timeleft -- - end_sound = UNDEF Destroying call '[EMAIL PROTECTED]' Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) I think that 2486543210 is not a customer, card number or SIP/IAX2 friend, maybe is PSTN number. To redirect the call to any PSTN number you must need to set voip call to inactive and set the destination number to 2486543210. I bet I am missing something in extension.conf correct? I dont see any examples in my package. The context is fine don't worry about it. Any suggestion... Thanks once again... Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Que on A2Billing
On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote: When I call from my cell to the above DID, it hits on the Asterisk and I see A2Billing trying to call SIP/2486543210, but it fails because Asterisk says Unable to create channel of type 'SIP' (cause 3 - No route to destination) . I know it, but the error is saying that you don't have one 2486543210 user registred. Show us the output of: sip show peers Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Que on A2Billing
On Thu, 2007-06-14 at 14:46 -0400, Nitesh Divecha wrote: Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the IVR Prompt... Meaning from my XLite dialer I want to dial directly and let A2Billing do the billing part. Right now is something like when I dial any number from XLite, A2Billing script is invoked and it will announce You have XXX amount, please enter the number you wish to call followed by #. And then I have to enter the number again and then the call is initiated... Its kinda annoying to do that every time you want to call. Is there anyway to modify config some where, so it will do the billing in background when the phone call is hangup. Yes, is possible using the a2billing.conf file in the right way. I don't have the v1.3 installed, but in the previous release 1.2.3 you must have to modify : use_dnid=YES number_try=1 say_balance_after_auth=NO say_balance_after_call=NO say_rateinitial=NO say_timetocall=NO Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Voip-info.org
On Wed, 2007-06-06 at 11:21 -0400, Justin Moore wrote: On 6/6/07, Ed Nuñez [EMAIL PROTECTED] wrote: Is anyone else having trouble going into voip-info.org today? Yep. Dead for me too. Dead from Ecuador too. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] real time billing system
On Fri, 2006-09-29 at 11:12 -0500, Pato Valarezo wrote: Hi, sorry for the question, i've been searching for a real time billing system for asterisk with zap/sip support, for use in post paid systems like locutorios, do you know of or use any ? Give a try to StarshopOSS: http://www.starshop-online.com/howto/how_to_setup_voip_calls_in_your_cybercafe_with_starshop_3.htm Regards, thanks -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] ooh323 hang up after the call is answered
Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556 ERROR:Failed to open audio channels. Clearing call.(incoming, ooh323c_1) 10:57:40:556 Sent Message - EndSessionCommand (incoming, ooh323c_1) 10:57:40:556 Sent Message - ReleaseComplete (incoming, ooh323c_1) 10:57:40:562 Received EndSession command (incoming, ooh323c_1) 10:57:40:562 Closing H.245 connection (incoming, ooh323c_1) 10:57:40:562 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:577 H.225 Release Complete message received (incoming, ooh323c_1) 10:57:40:577 Release complete reason code 12. (incoming, ooh323c_1) 10:57:40:577 Cleaning Call (incoming
Re: [asterisk-users] ooh323 hang up after the call is answered
I fogot, the H.323 device is one Antek networks INC with two fxo ports. Regards, On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote: Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556 ERROR:Failed to open audio channels. Clearing call.(incoming, ooh323c_1) 10:57:40:556 Sent Message - EndSessionCommand (incoming, ooh323c_1) 10:57:40:556 Sent Message - ReleaseComplete (incoming, ooh323c_1) 10:57:40:562 Received EndSession command (incoming, ooh323c_1) 10:57:40:562 Closing H.245 connection (incoming, ooh323c_1) 10:57:40:562
Re: [asterisk-users] ooh323 hang up after the call is answered
Solved... installed chan_oh323 :) http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 I don't know why ooh323 does not work. Regards, On Fri, 2007-02-23 at 11:21 -0500, Guillermo Salas M. wrote: I fogot, the H.323 device is one Antek networks INC with two fxo ports. Regards, On Fri, 2007-02-23 at 11:07 -0500, Guillermo Salas M. wrote: Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial(OOH323/Telconet Mantaer-c5f8, SIP/666|30|TtrwWC) in new stack Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on RTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Setting NAT on VRTP to 524288 Feb 23 10:57:32 DEBUG[6096] chan_sip.c: Outgoing Call for 666 Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Called 666 Feb 23 10:57:32 DEBUG[6096] channel.c: Driver for channel 'OOH323/Telconet Mantaer-c5f8' does not support indication 3, emulating it Feb 23 10:57:32 DEBUG[6096] channel.c: Prodding channel 'OOH323/Telconet Mantaer-c5f8' Feb 23 10:57:32 DEBUG[6096] channel.c: Scheduling timer at 160 sample intervals Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6079] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Feb 23 10:57:32 DEBUG[6068] channel.c: Avoiding initial deadlock for 'SIP/666-098cde60' Feb 23 10:57:32 VERBOSE[6096] logger.c: -- SIP/666-098cde60 is ringing Feb 23 10:57:36 DEBUG[6079] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Acked pending invite 102 Feb 23 10:57:40 DEBUG[6079] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Feb 23 10:57:40 DEBUG[6079] chan_sip.c: build_route: Contact hop: Guillermo Salas M sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Feb 23 10:57:40 VERBOSE[6096] logger.c: -- SIP/666-098cde60 answered OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 WARNING[6096] src/chan_h323.c: Don't know how to indicate condition -1 on ooh323c_1 Feb 23 10:57:40 DEBUG[6096] channel.c: Scheduling timer at 0 sample intervals Feb 23 10:57:40 DEBUG[6096] channel.c: Didn't get a frame from channel: OOH323/Telconet Mantaer-c5f8 Feb 23 10:57:40 DEBUG[6096] channel.c: Bridge stops bridging channels OOH323/Telconet Mantaer-c5f8 and SIP/666-098cde60 Feb 23 10:57:40 DEBUG[6096] chan_sip.c: update_call_counter(666) - decrement call limit counter Feb 23 10:57:40 DEBUG[6096] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'dial' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' in macro 'exten-vm' Feb 23 10:57:40 VERBOSE[6096] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Telconet Mantaer-c5f8' And the h323 log: 10:57:32:717 Created a new call (incoming, ooh323c_1) 10:57:32:753 Received SETUP message (incoming, ooh323c_1) 10:57:32:753 Tunneling disabled by remote endpoint. (incoming, ooh323c_1) 10:57:32:753 Enabled RFC2833 DTMF capability for (incoming, ooh323c_1) 10:57:32:754 Sent Message - CallProceeding (incoming, ooh323c_1) 10:57:32:754 Sent Message - Alerting (incoming, ooh323c_1) 10:57:40:475 Cmd connection accepted 10:57:40:476 Processing Answer Call command for ooh323c_1 10:57:40:476 Creating H245 listener 10:57:40:476 H245 listener creation - successful(port 12031) (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:476 Sent Message - Connect (incoming, ooh323c_1) 10:57:40:476 H.245 Listerner socket being monitored (incoming, ooh323c_1) 10:57:40:501 H.245 connection established (incoming, ooh323c_1) 10:57:40:501 Sent Message - TerminalCapabilitySet (incoming, ooh323c_1) 10:57:40:502 Sent Message - MasterSlaveDetermination (incoming, ooh323c_1) 10:57:40:538 Sent Message - TerminalCapabilitySetAck (incoming, ooh323c_1) 10:57:40:542 Master Slave Determination received (incoming, ooh323c_1) 10:57:40:542 MasterSlaveDetermination done - Slave(incoming, ooh323c_1) 10:57:40:542 Sent Message - MasterSlaveDeterminationAck (incoming, ooh323c_1) 10:57:40:556 Opening logical channels (incoming, ooh323c_1) 10:57:40:556 ERROR:Local endpoint does not have any audio capabilities (incoming, ooh323c_1) 10:57:40:556
RE: [asterisk-users] freepbx with ASTERISK 1.4
On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote: it's possible to configure freepbx 2.2 with asterisk 1.4? Look here for the archives: http://lists.digium.com/pipermail/asterisk-users/ Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0. You'll find EXACTLY what you're looking for. :-) Look at: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.amportal.user/5377 Regards, Stefano ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Softphone on Linux
On Fri, 2007-02-09 at 09:21 +, Tim Panton wrote: On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote: On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. [..] It's all in the graphics libraries etc. If you are already running firefox, the plugin isn't a huge extra overhead. Xten or Kiax will have a full set of their own .so which almost certainly won't be shared with anything else that is running. If you are already running firefox give a try to moziax: http://moziax.mozdev.org/ It's a firefox extension for using as IAX2 softphone. MozIAX is free software :) The only way to know for sure would be to try it on a sample system - fire up the browser, and click on: http://click.mexuar.com/webuser/click/145/userurl/Westhawk And give me a call (in UK office hours). [..] ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Detect hang-up
On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote: I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! Maybe my zapata.conf can help you. I've one X100P working for almost 2 years :) [channels] language=es context=from-pstn signalling=fxs_ks rxwink=300 ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=no hidecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no relaxdtmf=yes inmediate=yes busydetect=yes busycount=6 callprogress=yes musiconhold=default echotraining=400 rxgain=-4.0 txgain=4.0 group=0 callgroup=1 pickupgroup=1 TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Softphone on Linux
On Mon, 2007-02-05 at 22:37 +, Gordon Henderson wrote: On Mon, 5 Feb 2007, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. Any recomendations? Idefisk for the softphone. I agree idefisk. Is light and supports IAX2. Lynx for the browser ;-) Dilo or switfox [1] ;) [1] http://getswiftfox.com/releases.htm Gordon ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Software callcenter
On Mon, 2007-01-15 at 20:37 +0100, Lenz wrote: [..] Hello everybody Anyone know a software for callcenter, with statistics and reports and work with asterisk? Try MOR from www.kolmisoft.com Regards, Regards -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 to a Huawei softX3000 problem has already been solved ！
Solved :) Added at sip.conf : silencesuppression=yes Regards, On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote: Kevin, contributes with the list, somebody can have this problem and you it can help. The list is here for helping, but also we must contribute with it. :) Best Regards Josue 2006/12/13, kevinho [EMAIL PROTECTED]: Asterisk to a Huawei softX3000 problem has already been solved ！ msn:[EMAIL PROTECTED] _ Windows Live Safety Center 为您的计算机提供免费的安全扫描服 务。 http://safety.live.com/site/ZH-CN/default.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 to a Huawei softX3000 problem has already been solved ！
On Fri, 2007-01-12 at 15:46 -0800, Rehan Allah Wala wrote: What about Huawei to Asterisk ? Is it the same problem with that ? I get a weird error, call comes in, i answer and it disconnects. The problem is with the silence suppression. Try disabling it on asterisk. Regards, Rehan Subject: Re: [asterisk-users] Asterisk to a Huawei softX3000 problem has already been solved ï¼ From: Guillermo Salas M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Organization: Telconet S.A. Date sent: Fri, 12 Jan 2007 18:37:35 -0500 Send reply to: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com asterisk-users.lists.digium.com mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Solved :) Added at sip.conf : silencesuppression=yes Regards, On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote: Kevin, contributes with the list, somebody can have this problem and you it can help. The list is here for helping, but also we must contribute with it. :) Best Regards Josue 2006/12/13, kevinho [EMAIL PROTECTED]: Asterisk to a Huawei softX3000 problem has already been solved ！ msn:[EMAIL PROTECTED] _ Windows Live Safety Center 为您的计算机提供免费的安全扫描 服 务。 http://safety.live.com/site/ZH-CN/default.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! MSN: [EMAIL PROTECTED] Skype: Rehan33 First they ignore you, then they laugh at you, then they fight you, then you win. By Mahatma Gandhi. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 to a Huawei softX3000 problem has already been solved ！
On Wed, 2006-12-13 at 21:51 -0300, Josué Conti wrote: Kevin, contributes with the list, somebody can have this problem and you it can help. The list is here for helping, but also we must contribute with it. :) Best Regards I have the same problem.. any one know can I solve it? Best resgards, Josue 2006/12/13, kevinho [EMAIL PROTECTED]: Asterisk to a Huawei softX3000 problem has already been solved ！ msn:[EMAIL PROTECTED] _ Windows Live Safety Center 为您的计算机提供免费的安全扫描服 务。 http://safety.live.com/site/ZH-CN/default.htm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Trouble compiling asterisk 1.2.14
Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops -O6 -march=k8 -fPIC -DSASR -DNDEBUG-DWAV49 -I./inc src/add.c cc1: error: bad value (k8) for -march= switch cc1: error: bad value (k8) for -mcpu= switch make[2]: *** [src/add.o] Error 1 make[2]: Leaving directory `/usr/src/asterisk-1.2.14/codecs/gsm' make[1]: *** [gsm/lib/libgsm.a] Error 2 make[1]: Leaving directory `/usr/src/asterisk-1.2.14/codecs' make: *** [subdirs] Error 1 ruidoso:/usr/src/asterisk-1.2.14# uname -sr Linux 2.6.8-12-amd64-k8 Maybe I'm forgotting to install any dependency? Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Trouble compiling asterisk 1.2.14
[SOLVED] On Thu, 2007-01-04 at 21:33 +0200, Tzafrir Cohen wrote: On Thu, Jan 04, 2007 at 01:42:57PM -0500, Guillermo Salas M. wrote: Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 [..] gcc 3.3 and gcc 3.4 (which are availble on Sarge) don't support that, I believe. Try setting those values explicitly in the makefile. Or try reproducing the build of the deb package from the pkg-voip buildserver: http://pkg-voip.buildserver.net The problem is solved using the pkg-voip packages. Asterisk is installed and running now. 1.- Edit /etc/apt/sources.list 2.- Add the line: deb http://pkg-voip.buildserver.net/debian sarge main 3.- aptitude update 4.- aptitude install asterisk Thank you for your help. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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: [Announce] Web-MeetMe 3.0.0 released
On Wed, 2007-01-03 at 16:55 -0500, Steven wrote: Any screenshots available? I do not want to even test this without having any idea what it is or how it works. The brief description on sf.net is not enough. I'm testing the 2.0 version on asterisk 1.2 . What do you want to know about the application? Best regards, -- -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] FXO USB that works with Asterisk?
On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] FXO USB that works with Asterisk?
On Thu, 2006-12-07 at 18:17 +, jose luis peche baldera wrote: Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,. WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native formats* is 8 (*read/write* = *64/64*) Alguna sugerencia Please, make a new message for a new question, do not reply a thread with a different topic, and finally, use english. Regards, __ From: Guillermo Salas M. [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] FXO USB that works with Asterisk? Date: Thu, 07 Dec 2006 12:46:44 -0500 On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 __ Éxitos, grandes clásicos y novedades. Un millón de canciones en MSN Music. ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Switching from FreeBSD to Linux - which distro?
On Tue, 2006-12-05 at 18:47 -0500, Mike Garey wrote: I recommend debian, been using it for years now, it was a no brainer to choose this for my asterisk deployments.. A few other people I know have used debian with asterisk with no problems either. Choose Debian, is easy to maintain.. apt-get rocks ! On 12/5/06, Phil Finkler [EMAIL PROTECTED] wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards, Phil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Billing software with reseller accounts
Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Voxee lag problems ?
On Fri, 2006-11-10 at 17:29 -0800, Tom Lynn wrote: Add me to the list. Not only lagged, but also failures to register. AND, apparantly Paypal won't automatically authorize payments to them anymore. I'm not recharging my account anymore. Is working fine from me. You can reach the payments page and send the money via PayPal to the e-mail noted in red words as an announcement in the same page. Regards, On 11/10/06, Tim Panton [EMAIL PROTECTED] wrote: On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here - wrote an email to support. They claim that their servers are fine and will get back to me in a day or two... Now there is a definitive case of a 'lagged' communication channel! :-) Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Java Web Phone
On Wed, 2006-11-01 at 16:05 -0500, Vladimir Montealegre Estailes wrote: Hello list partners you know about a softphone made in java attachable in a web page? GNU! I'm using JIAXClient [1] to permit to any user to join one meetme room [2] with the IAX2 protocol, works very great for me, and is very easy to install and modify to your needs. [1] http://www.hem.za.org/jiaxclient/ [2] http://www.rmsenecuador.info/jiaxclient/index.html Thaks in advance! __ Visita www.tutopia.com y comienza a navegar más rápido en Internet.Tutopia es Internet para todos. Web Bug from http://www.tutopia.com/bannerserving/banman.asp?ZoneID=0BannerID=3494AdvertiserID=776CampaignID=2739Task=GetMode=TEXT ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Running asterisk with 'sudo'
On Thu, 2006-11-02 at 17:12 +0100, Asterisk wrote: Hi guys, I'm using RedHat and am trying to configure my sudo to enable user 'testuser' to run Asterisk. However whenever I try to run 'sudo asterisk' as 'testuser' I get prompted for password. This is the line in my sudoers configuration file that I thought should do the trick, but it doesn't: testuser ALL=NOPASSWD: /usr/sbin/asterisk Does anyone know how to configure the sudo so that 'testuser' will be able to run the asterisk? Use the visudo to make changes to /etc/sudoers , to make the sudo stop asking for a password you need a line at /etc/sudoers like (take note on the space after the = ): testuser ALL= NOPASSWD: /usr/sbin/asterisk Thanks, Alex ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem
On Tue, 2006-10-31 at 09:55 -0500, Zeeshan Zakaria wrote: Anybody knows why ARI gives this error message when I enter extension number and password. Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed to open stream: Permission denied in /var/www/html/recordings/modules/voicemail.module on line 525 Are you sure about the file permissions? The file /var/spool/asterisk/voicemail/default/222/INBOX/msg txt must be permissions for the apache user or group. Try changing the ownership of the file. Using Debian will be like (apache group is called www-data): chown asterisk:www-data /var/spool/asterisk/voicemail/default/222/INBOX/msg Regards, It doesn't show the voicemails, although it shows that there is 1 or 2 voicemails in the INBOX. -- Zeeshan A Zakaria ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Billing Solution ?
On Mon, 2006-10-30 at 18:31 +0100, Noc Phibee wrote: Hi what is the best billing solution for Asterisk ? With WWW manager interface for user can see the real invoice... I'm using a2billing and works like a charm for me :) Regards, Thanks bye ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] SIP v IAX2
On Thu, 2006-10-26 at 13:14 -0400, Henry.L.Coleman wrote: As I understand it the main advantege IAX has over SIP is the number of port it uses and therefore its ability to traverse router/switches and firewalls Also the higher number of simulatanious SIP calls travelling through these devices adds a higher overhead than IAX with it's single port. Personally I like IAX but I there simply isnt enough hardware out there to use it exclusively. What about the bandwidth used for both protocols? Is IAX using less or more bandwidth than SIP? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Thursday, October 26, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote: with SIP qualify, I can specify, what time in delay I will accept, with sip and setting qualify=3000 I can circumvent this anoying messages (bacause delay in reply is about 2000ms, and I accept 3000ms) with iax, qualify is working different, so setting qualify=3000 will ping peer every 3s, quite inconsistent, imho So are you saying that in your world two different things, created by totally different people, must have the same configuration settings. - You will find DUNDi configuration a lot easier with IAX, although you can use SIP. - If you use SIP to route calls between Asterisk boxes, you will lose your caller id as SIP uses the From: number to authenitcate with. You will have to store the original caller id in an extra SIP header, and then pluck it out an the other end, if you want to preserve caller id. Yuck. IAX doesn't have this problem. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 softphones
On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] MOR and MCC - billing solutions for Asterisk released
Hello, On Tue, 2006-08-29 at 15:52 +0300, Mindaugas Kezys wrote: Hello, Kolmisoft: http://www.kolmisoft.com released new versions of MCC and MOR - Billing solutions for Asterisk PBX MOR - is new product, it's MCC v2. Rewritten to support MySQL and is based on Ruby on Rails. When can I find the user guide for start using MCC v1.5 or where can I download the MCC v2 ? At the kolmisoft.com site I can only see the v1.5. Best regards, ChangeLog for MCC: DB: New fields: providers.enable_cid_prefix varchar providers.disable_cid_prefix varchar providers.min_time integer default 1 providers.increment integer default 1 cids.enable_to_provider boolean default true cids.nat boolean default true cids.voicemail boolean defaut true cids.voicemail_psw charvar default '' rates.connection_fee double precision default 0 calls.rate double precision default 0 calls.user_connection_fee double precision default 0 calls.rate_connection_fee double precision default 0 users.first_name varchar users.last_name varchar users.min_time integer default 1 users.increment integer default 1 Fixed 386 code from Slovakia to Slovenia APP: Fixed bug with transfers - thanks German Aracil - suspended, needs more testing Changes to support CID manipulation - sponsored by Imre Csaba Varasdy Changes to support connection_fee based on rates(destinations) Rate, user_connection_fee, rate_connection_fee now are added to calls data Min_time and increment for billed time now taken from db, not conf file GUI: Fixed bug with email exists message Added Spanish translation - thanks German Aracil Added Hungarian translation - thanks Imre Csaba Varasdy Added German translation - thanks Inga A. Added Albanian translation - thanks Arben Myrtaj Now possible to assign connection_fee for rate(destination) User name split into First Name and Last Name Voicemail support in autoconfiguration, reachable by *98 for VoIP users User/admin can change cid/nat/voicemail/voicemail password for user's every CID (which supports autoconf.) under his details and when registering Possible to change call's status from processed to not (Changes color in GUI) and hide 'processed' calls in invoices. Possible to hide calls shorter than 'x' seconds. User can see his payments When registering, possible to set address like: http://mcc.company.com/register.php?ref=27, then referrer's field will be filled automatically Register authentication with noisy picture to prevent bot-registering Registration process reworked, check more here Reseller's CID's moved to new section - Devices Now various billing options (1/1, 6/6, 30/6) could be set per user basis - sponsored by Patrick Cardozo ASR (Average Success Rate) / ALOC (Average Length of Call) counting - sponsored by Patrick Cardozo New window to check CIDs and Extensions New values to define.php $USE_PROCESSED_CALLS $REG_ADDITIONAL - Additional info in registrtion page (like come visit our VoIP store) Regards, Midnaugas Kezys ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] real time billing system
On Fri, 2006-09-29 at 16:48 -0500, Pato Valarezo wrote: Chapeti wrote: Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que me parece mas fácil es que te hagas uno propio, echale una ojeada a lo que hay en http://www.voip-info.org/wiki-Asterisk+manager+API, lo único que haría falta sería un poco de conocimientos deVB 6 y de como trabajar con sockets ( cosa que no es nada del otro mundo ). Saludos. mmm... bueno no estaba buscando precisamente algo que sea abierto, simplemente algo que me ayude a instalar un pequeño locutorio con telefonos sip y con 4 salidas zap. Para lo que me comentas del AMI, muy interesante, la verdad que se pueden hacer maravillas... aunque no lo hiciera en VB, mas bien en algo mejor como python!. Voy a buscar que encuentro y si no hay nada adaptable me pondré manos a la obra con esto. Que tal Pato :) I'm using a2billing at my cybercafe and works very well. You can use starshop-oss as well the setup instructions are at: http://www.starshop-online.com/howto/how_to_setup_starshop.htm I preffer a2billing because is giving me more features like having two or more providers for the same destination and LCR. Saludos, gracias por la información. saludos -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] G729 Softphone
On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote: Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Give a try to Eyebeam at www.counterpath.com , it supports video and voice with g729. BOL Siphone is freeware that supports video/voice and uses de g723.1 codec you can download it at http://www.bol2000.com/download/sipphone/ Thanks, Daniel ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 settings Net2Phone
On Tue, 2006-05-09 at 11:37 -0300, Vinícius Bossle Fagundes wrote:
Hi,
I´m looking for settings to configure net2phone carrier in my
asterisk. I found this configurations, but it´s not work. I don´t
known if this configuration is for voice line or voice access account.
Anybody can help me, with other configuration?
I've some net2phone accounts working with Asterisk.
Thanks.
sip.conf
[general]
useragent = X-Lite release 1103m
register = PHONENUMBER:[EMAIL PROTECTED]
---
sip.conf
---
[general]
useragent = Cisco ATA 186 v3.1.0 atasip
register=NET2PHONEACCOUNT:[EMAIL PROTECTED]
[net2phone]
username=NET2PHONEACCOUNT
useragent=Cisco ATA 186 v3.1.0 atasip (040211A)
type=peer
secret=PINNUMBER
qualify=no
nat=yes
insecure=very
host=sip.net2phone.com
fromuser=NET2PHONEACCOUNT
fromdomain=net2phone.com
canreinvite=no
allow=g723
[net2phone]
type = peer
host = sip.net2phone.com
username = PHONENUMBER
secret = PASSWORD
fromuser = PHONENUMBER
fromdomain = net2phone.com
context = incoming
insecure = very
canreinvite = no
extensions.conf
[outgoing]
exten = _9NXXNXX,1,Dial(SIP/net2phone/${EXTEN:1})
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Re: [Asterisk-Users] Net2phone on asterisk
On Sun, 2006-05-21 at 19:45 -0500, Daniel wrote: Has anyone setup a n2p account into asterisk? Yes, check http://lists.digium.com/pipermail/asterisk-users/2006-May/152317.html Regards, Guillermo. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 to SIP
On Sun, 7 May 2006 19:58:26 +0500, Farhad Ibragimov [EMAIL PROTECTED] wrote: Thanks Try reading this URL (spanish language): http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 With the page instructions I can call from and to H.323 to every registred SIP/IAX2/H.323 device with my Asterisk box. Good luck, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I dont have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo V. Salas M Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 1er Piso Teléfono: 262 8071 Celular : 09 985 5138 Manta - Manabí - Ecuador ___ --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] the best billing tool for Asterisk
On Fri, 2006-04-14 at 13:08 -0700, Mindaugas Kezys wrote: Hello, You can try: http://www.paskambink.lt/mcc Or can try http://www.asterisk2billing.org/ it supports postgresql Regards/Pagarbiai, Mindaugas Kezys -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, April 11, 2006 9:55 AM To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] the best billing tool for Asterisk On 4/11/06 8:14 AM, Joao Pereira [EMAIL PROTECTED] wrote: Hello to all I would like to know some opinions of people that are using billing tools for Asterisk. Can you please advise me in wich billing tool to I use? Thanks Joao Pereira ___ --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 Lots of people whip together their own solution as there is no billing solution out there for Asterisk that fits all. Usually you end up making tweaks here and there even if you do use a prebuilt solution. ___ --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] [EMAIL PROTECTED] and H323
On Tue, 2006-03-07 at 12:08 +0200, Viktor Tatianin wrote: Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': If you have asterisk 1.2.4 version you must have to compile oh323 as in http://www.oinko.net/astrecipes/index.php?n=40 but replacing the versions from: http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib-Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh323-Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz Please help for resolve this problem Viktor Tatianin ___ --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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Problem getting two x200p cards working on 1.2.4
Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface This is the cli output for zap show channels : My /etc/zaptel.conf : # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCFXO/1 Generic Clone Board 2 fxsks=1 # Span 2: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us My /etc/asterisk/zapata-auto.conf ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCFXO/1 Generic Clone Board 2 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1 context=from-pstn group=0 channel = 1 ; Span 2: ZTDUMMY/1 ZTDUMMY/1 1 This is the corresponding 'lspci -vv -n' for my two cards: 02:01.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 201 Region 0: I/O ports at b800 [size=256] Region 1: Memory at feaff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 02:03.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 177 Region 0: I/O ports at b000 [size=256] Region 1: Memory at feafd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- And dmesg shows: NET: Registered protocol family 10 Disabled Privacy Extensions on device c0340020(lo) IPv6 over IPv4 tunneling driver divert: not allocating divert_blk for non-ethernet device sit0 eth0: no IPv6 routers present Freed a Wildcard Unregistered Tormenta2 Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: Echo Canceller: KB1 Registered Tormenta2 PCI Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) ACPI: PCI interrupt :02:01.0[A] - GSI 22 (level, low) - IRQ 201 Failed to initailize DAA, giving up... wcfxo: probe of :02:01.0 failed with error -5 ACPI: PCI interrupt :02:03.0[A] - GSI 19 (level, low) - IRQ 177 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Any ideas? -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Problem getting two x200p cards working on 1.2.4
On Mon, 2006-03-06 at 14:11 -0900, Mojo with Horan Company, LLC wrote: in zaptel.conf, you have fxsks=1 -- this only allocates the first card. try fxsks=1-2 instead. Done, but still with problems: [EMAIL PROTECTED] ~]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Changing signalling on channel 2 from Clear channel to FXS Kewlstart ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? dmesg is showing and error with one of the cards, some realted with DAA: Freed a Wildcard Unregistered Tormenta2 Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: Echo Canceller: KB1 Registered Tormenta2 PCI ACPI: PCI interrupt :02:01.0[A] - GSI 22 (level, low) - IRQ 201 Failed to initailize DAA, giving up... wcfxo: probe of :02:01.0 failed with error -5 ACPI: PCI interrupt :02:03.0[A] - GSI 19 (level, low) - IRQ 177 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) And asterisk cli is only showing one card: mail*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-pstn en 1from-pstn en mail*CLI Guillermo Salas M wrote: Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface This is the cli output for zap show channels : My /etc/zaptel.conf : # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCFXO/1 Generic Clone Board 2 fxsks=1 # Span 2: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us My /etc/asterisk/zapata-auto.conf ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCFXO/1 Generic Clone Board 2 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1 context=from-pstn group=0 channel = 1 ; Span 2: ZTDUMMY/1 ZTDUMMY/1 1 This is the corresponding 'lspci -vv -n' for my two cards: 02:01.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 201 Region 0: I/O ports at b800 [size=256] Region 1: Memory at feaff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 02:03.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 177 Region 0: I/O ports at b000 [size=256] Region 1: Memory at feafd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- And dmesg shows: NET: Registered protocol family 10 Disabled Privacy Extensions on device c0340020(lo) IPv6 over IPv4 tunneling driver divert: not allocating divert_blk for non-ethernet device sit0 eth0: no IPv6 routers present Freed a Wildcard Unregistered Tormenta2 Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: Echo Canceller: KB1
Re: [Asterisk-Users] Problem getting two x200p cards working on 1.2.4
On Mon, 2006-03-06 at 14:13 -0900, Mojo with Horan Company, LLC wrote: forgot to add this -- after the change in zaptel.conf, you would want to put a channel = 2 in zapata.conf right after the channel = 1 at the bottom. Now asterisk can actually see it. Done, but still having the error: Mar 6 18:21:48 VERBOSE[7394] logger.c: -- Registered channel 1, FXS Kewlstart signalling Mar 6 18:21:48 WARNING[7394] chan_zap.c: Unable to specify channel 2: Device or resource busy Mar 6 18:21:48 ERROR[7394] chan_zap.c: Unable to open channel 2: Device or resource busy here = 0, tmp-channel = 2, channel = 2 Mar 6 18:21:48 ERROR[7394] chan_zap.c: Unable to register channel '2' Mar 6 18:21:48 WARNING[7394] loader.c: chan_zap.so: load_module failed, returning -1 Mar 6 18:21:48 VERBOSE[7394] logger.c: -- Unregistered channel 1 Mar 6 18:21:48 WARNING[7394] loader.c: Loading module chan_zap.so failed! It's like the card is not being recognized or something like it, but the cards are the same and was working at another computer without troubles. Guillermo Salas M wrote: Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface This is the cli output for zap show channels : My /etc/zaptel.conf : # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCFXO/1 Generic Clone Board 2 fxsks=1 # Span 2: ZTDUMMY/1 ZTDUMMY/1 1 # Global data loadzone= us defaultzone = us My /etc/asterisk/zapata-auto.conf ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCFXO/1 Generic Clone Board 2 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 1 context=from-pstn group=0 channel = 1 ; Span 2: ZTDUMMY/1 ZTDUMMY/1 1 This is the corresponding 'lspci -vv -n' for my two cards: 02:01.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 201 Region 0: I/O ports at b800 [size=256] Region 1: Memory at feaff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 02:03.0 Class 0780: e159:0001 Subsystem: 8086:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 177 Region 0: I/O ports at b000 [size=256] Region 1: Memory at feafd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0 +,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- And dmesg shows: NET: Registered protocol family 10 Disabled Privacy Extensions on device c0340020(lo) IPv6 over IPv4 tunneling driver divert: not allocating divert_blk for non-ethernet device sit0 eth0: no IPv6 routers present Freed a Wildcard Unregistered Tormenta2 Zapata Telephony Interface Unloaded Zapata Telephony Interface Registered on major 196 Zaptel Version: Echo Canceller: KB1 Registered Tormenta2 PCI Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) ACPI: PCI interrupt :02:01.0[A] - GSI 22 (level, low) - IRQ 201 Failed to initailize DAA, giving up... wcfxo: probe of :02:01.0 failed with error -5 ACPI: PCI interrupt :02:03.0[A] - GSI 19 (level, low) - IRQ 177 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone Registered tone zone 0 (United States / North America) Registered tone zone 0 (United
Re: [Asterisk-Users] Re: a2billing without IVR
On Fri, 2006-02-24 at 10:58 +, Barry Flanagan wrote: Asterisk Sales wrote: mailto:asterisk-users@lists.digium.com Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destination followed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. Yes, this is all configurable from /etc/asterisk/a2billing.conf If you set use_dnid=YES then a2billing will pick up the destination from the number the user dialled. Set the following to turn off the IVR stuff: ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth=NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call=NO ; Play the time the user can call (values : yes - no) say_timetocall=NO Hope this helps. Thank you, is working for me right now :) -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Skype vs. an Xlite registered to Asterisk
On Sun, 2006-02-26 at 15:50 -0600, Michael Graves wrote: I second this...as a road warrior myself I find too many places where SIP clients just won't work. So I rely on Firely over IAX2 which has been 100% reliable. I'm using idefisk softphone with iLBC on my Debian roadwarrior laptop and works very nice. Also, John Todd has been using the PSGW SkypeSIP gateway software in new and different ways. Perhaps that's an option. On Sun, 26 Feb 2006 18:30:07 +0100, asterisk wrote: Hi I have good results in using, the old very (free) of firefly (IAX2), with g729! rgds Jesper Langpap hugolivude wrote: I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets clipped to static if they speak too loudly. Many of these users have migrated to Skype – much to my chagrin! I'd like to get them back using a SIP client so they can take advantage of all Asterisk can offer. Anyone else had trouble with voice quality with Xlite? Any work arounds? I was thinking about trying an Xlite client that can support G729. Anyone had experience with that? Does it significantly improve voice quality? I also read that SJ Phone is better than XLite, but is it really the client application that makes the biggest difference or the codec? Perhaps it's a combination or something entirely different? Anyone with experience with an SJ Phone and G729 codec? Any suggestions welcome! Yours, Hugh P.S Asterisk 1.2 on Redhat 9.0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Anyone using LG LIP-100 ip phone
Hi, Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this phones but seems to work only with net2phone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTPmodelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Problems with gnugk, asterisk, and ooh323
On Thu, 2006-02-09 at 21:27 +0100, Joe wrote:
Greetings to All,
I hope someone has already gotten this working. I spent all day today trying
to get ooh323 and gnugk to run on the same box. After a lot of tweaking to
get everything compiled, I got both up and running.
I've it working. Can call from H.323 to SIP or IAX and from SIP or IAX
to H.323. The H.323 devices can use the SIP or IAX trunks to call
outside.
I can make calls IAX to H323, but cannot make calls in the reverse
direction. I have tried many different configs on the GK, but always come up
with the same error. It appears to me that asterisk successfully registers
with GK as I can see the aliases and the e.164 numbers, but when the h323
softphone tries to call my IAX softphone, I get this:
I'm not using gatekeeper on th oh323.conf. The H.323 devices are using
the asterisk ip as gateway (not gatekeeper) and to call to H.323 I need
to specify the number of the device followed by the ip
([EMAIL PROTECTED]).
admissionRequest {
requestSeqNum = 2
callType = pointToPoint null
endpointIdentifier = 9 characters {
0033 0032 0039 0037 005f 0065 006e 0064 3297_end
0070 p
}
destinationInfo = 1 entries {
[0]=dialedDigits 100 no IP ???
}
srcInfo = 2 entries {
[0]=h323_ID 8 characters {
0073 006f 0066 0074 0070 0065 0065 0072 softpeer
}
[1]=dialedDigits 123
}
bandWidth = 1
callReferenceValue = 4096
conferenceID = 16 octets {
ee 87 f1 90 9f 46 8d 40 94 7e 5e 75 87 e5 c0 15 [EMAIL PROTECTED]
}
activeMC = FALSE
answerCall = FALSE
canMapAlias = TRUE
callIdentifier = {
guid = 16 octets {
e1 87 ea ec d6 83 ff 4a bd 0c f5 b9 93 6e 32 df ...J.n2.
}
}
cryptoTokens = 2 entries {
[0]=cryptoEPPwdHash {
alias = h323_ID 8 characters {
0073 006f 0066 0074 0070 0065 0065 0072 softpeer
}
timeStamp = 1139496380
token = {
algorithmOID = 1.2.840.113549.2.5
paramS = {
}
hash = Hex: 64 71 36 5e 9b 4e a8 64 c4 fe bf 5d dd 6e 22 00
}
}
[1]=cryptoEPPwdHash {
alias = dialedDigits 123
timeStamp = 1139496380
token = {
algorithmOID = 1.2.840.113549.2.5
paramS = {
}
hash = Hex: 06 d8 7b d1 ff 54 e3 bb c9 66 49 c2 4d cb 38 94
}
}
}
willSupplyUUIEs = FALSE
}
2006/02/09 15:47:02.165 1 RasSrv.cxx(343) RAS ARQ Received
2006/02/09 15:47:02.166 3 RasSrv.cxx(1948) GK ARQ will
request bandwith of 1280
2006/02/09 15:47:02.181 2 RasSrv.cxx(388)
ARJ|195.27.242.114:3775|100:dialedDigits|softpeer:h323_ID=123:dialedDigits|f
alse|calledPartyNotRegistered;
2006/02/09 15:47:02.181 3 RasSrv.cxx(231) RAS Send to
195.27.242.114:3774
admissionReject {
requestSeqNum = 2
rejectReason = calledPartyNotRegistered null
I have to assume here that the called party is Asterisk, but I cannot find
any information regarding using a username and password for Asterisk GK
registration.
When a call goes from IAX to H323, the destinationInfo has 2 entries, e.164
number, and an IP address.
If anyone has gotten this to work, I would love to hear how.
Regards to all.
Joe
P.S. Running *1.2.4 (crashes quite a bit by the way with ooh323) and ooh323
from add-ons 1.2.1
Joe
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Re: [Asterisk-Users] help with oh323
On Fri, 2006-02-10 at 00:45 +0500, Hussain Umair wrote: hi ive been tryin to get oh323 to work and installed it without any problems but it gives me the same error all the time this is the third time ive installed it..please if anyone can kindly help me out thanks in advance... [chan_oh323.so]Feb 10 00:35:29 WARNING[4891]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZN11PTimedMutexC2Ev Feb 10 00:35:29 WARNING[4891]: loader.c:440 load_modules: Loading module chan_oh323.so failed! I had the same problem before. Have you checked the variable ASTERISKINCDIR in the Makefile from the oh323 path? It must be pointing to the path containig the include source of your running asterisk version. # # Set ASTERISKINCDIR variable to the directory containing the include files of # Asterisk PBX. # #ASTERISKINCDIR=/usr/src/asterisk/include ASTERISKINCDIR=/usr/src/asterisk-1.2.4/asterisk-1.2/include Regards, Umair. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problema calling from elesign h.323 to iax
On Wed, 2006-02-22 at 21:44 +0200, [EMAIL PROTECTED] wrote: Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb 22 14:27:11 VERBOSE[22106] logger.c: -- IAX2/911-12 is ringing Feb 22 14:27:17 VERBOSE[22106] logger.c: -- IAX2/911-12 answered OH323/[EMAIL PROTECTED] Feb 22 14:27:18 VERBOSE[22105] logger.c: H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9', exception CTRL_ERROR (Capability Exchange [Rejected]). Feb 22 14:27:18 VERBOSE[22073] logger.c: -- H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9' cleared, reason 24 (Call ended with Q.931 cause [31 - Normal, unspecified]) Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'IAX2/911-12' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'dial' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'exten-vm' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'OH323/[EMAIL PROTECTED]' Any ideas? I'm using the channel_oh323.so module. I've another h.323 device tha works without problems. Best regards, have you tried playing around with fastStart ; h245Tunnelling ; h245inSetup It's working now :) ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no Thank you very much for the help ;) it helps to change and see what happens -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Problema calling from elesign h.323 to iax device
Hi, i'm using an elesign voip gateway esc1700 to call to one iax sofphone (idefisk) and sip device (sipura spa-2002), the trouble es when I make the call using the esc1700 the communication is dropped, this is the log portion: Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Call accepted by 200.93.220.21 (format ulaw) Feb 22 14:27:11 VERBOSE[22069] logger.c: -- Format for call is ulaw Feb 22 14:27:11 VERBOSE[22106] logger.c: -- IAX2/911-12 is ringing Feb 22 14:27:17 VERBOSE[22106] logger.c: -- IAX2/911-12 answered OH323/[EMAIL PROTECTED] Feb 22 14:27:18 VERBOSE[22105] logger.c: H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9', exception CTRL_ERROR (Capability Exchange [Rejected]). Feb 22 14:27:18 VERBOSE[22073] logger.c: -- H.323 call 'ip $201.218.10.58:30010/18733-f4b26fe9' cleared, reason 24 (Call ended with Q.931 cause [31 - Normal, unspecified]) Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'IAX2/911-12' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'dial' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' in macro 'exten-vm' Feb 22 14:27:18 VERBOSE[22106] logger.c: == Spawn extension (macro-dial, s, 10) exited non-zero on 'OH323/[EMAIL PROTECTED]' Feb 22 14:27:18 VERBOSE[22106] logger.c: -- Hungup 'OH323/[EMAIL PROTECTED]' Any ideas? I'm using the channel_oh323.so module. I've another h.323 device tha works without problems. Best regards, -- ___ --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] calling from SIP to a h.323 device with oh323
On Mon, 2006-02-20 at 17:04 +0100, Marc Patino Gómez wrote: Hi, Can you post your working config, I'm wasting my time to config h323-sip Is working now :) I'm using asterisk-oh323 0.7.3 on my asterisk 1.2.4 box. I've to configure in oh323.conf with gatekeeper=DISABLED and the context of my sip clients. The H.323 device is configured to use the asterisk ip address as gateway. With this config I can use SIP/IAX2 trunks to call outside from the h.323 device and can call from SIP/IAX2 to H.323 and from H.323 to my SIP/IAX2 devices :) sip*CLI oh323 show conf sip*CLI Configuration of OpenH323 channel driver -- Version: 0.7.3 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported formats in pref. order: alaw0 ulaw1 gsm2 g7233 g7294 Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: tone Max number of inbound H.323 calls: 100 Max number of outbound H.323 calls: 100 Max number of simultaneous H.323 calls: 100 Max call rate (ingress direction): 1.00/30 Default language: es Default music class: default Default context: from-internal sip*CLI I've to create the h.323 extentions for the two ports of my H.323 device (ext 103 and 104 for port 1 and port 2) : [ext-local] include = ext-local-custom exten = 101,1,Macro(exten-vm,novm,101) exten = 101,hint,SIP/101 exten = 102,1,Macro(exten-vm,novm,102) exten = 102,hint,SIP/102 exten = 103,1,Macro(exten-vm,novm,103) exten = 103,hint,OH323/[EMAIL PROTECTED] exten = 104,1,Macro(exten-vm,novm,104) exten = 104,hint,OH323/[EMAIL PROTECTED] exten = 555,1,Macro(exten-vm,novm,555) exten = 555,hint,SIP/555 Thanks Guillermo Salas M wrote: Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday). = Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip (pid = 29977) nip*CLI Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] calling from SIP to a h.323 device with oh323
Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday). = Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip (pid = 29977) nip*CLI Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] No path to translate from Zap to SIP
I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with SIP/usa-e2ea It only happens with sip trunks, with iax2 trunks the calls works like a charm. Can you help to fix it? -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] No path to translate from Zap to SIP
On Fri, 2006-02-03 at 14:42 -0800, Philip Edelbrock wrote: Hmm, I'm guessing you are allowing codecs on SIP which aren't translatable. Try only allowing ulaw and alaw in your sip.conf. Thank you. Edited sip.conf and now the sip trunks are working :) Phil Guillermo Salas M wrote: I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call because I couldn't make Zap/1-1 compatible with SIP/usa-e2ea It only happens with sip trunks, with iax2 trunks the calls works like a charm. Can you help to fix it? -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] (newby) IAX Trunk on low bandwidth connection
On Wed, 2006-02-01 at 18:28 +0100, Master_PE wrote: What is a normal dealy on a satelite installation? I've 650ms right now from my router to Internet. Regards, Master_PE Op 1-feb-2006, om 13:26 heeft Garth van Sittert het volgende geschreven: Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/ bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of call center. That is, we want to get a few land-lines from our telco in different countys and bridge all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP in our area that can deliver a broadband connection for anything less then an arm and a leg, so we're considering runing an * - * connection using VoIP over a low bandwidth connection (we're considering 128kbit but we might be able to go to 256kbit). The bandwidth price is not a problem for our satelite installations, we cand get acceptably priced broadband (~256kbit) so the distant *'s will have propper connections. My question: Is 128kbit a wide enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund ___ --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 From - Wed -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] VOIP carriers and asterisk
Con fecha 28/1/2006, burak balasaygun [EMAIL PROTECTED] escribió: Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? nufone.net voipjet.com voxee.com thanks Burak Balasaygu ___ --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] Best CoDec for high network latency
Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo. ___ --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] Best CoDec for high network latency
Con fecha 28/1/2006, Jean-Michel Hiver [EMAIL PROTECTED] escribió: Guillermo Salas M a écrit : Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? There are basically three parameters I can think of when speaking of voice over ip quality: 1 - Lag. In your case, a ping from your Asterisk box is 1 to 1.5 ms. Changing codecs is not going to help you here. The lag if 1000 ~ 1500 ms 2 - Jitter. In your case, if the ping does vary between 1 and 1.5, that's 500ms ping jitter, which is high. You might want to have a large jitter buffer to compensate for it. But this increases lag even more... 3 - Packet drop. iLBC is meant to cope better with packet drop than other codecs, although in my experience any codec with too much packet drop will sound dreadful. If you have the bandwith and no packet loss, I would recommend that you bump up the jitter and stick with ulaw. While there might be a lot of lag - half duplex kind of conversations... - the audio should remain clear. I don't have packet loss, but my BW is limited. If you are having packet loss on top of this, you might want to try iLBC... At any rate, nothing is going to replace trying out some settings for yourself... BTW: How come the latency is so high? The worst I've seen so far was a link varying between 600 and 1200ms and the quality varied from good enough to pretty horrible... Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-user ___ --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
