[asterisk-users] TDM400P usada?
Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. ___ --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] TDM400P usada?
Chile. No hay listas en español, y si lo enviè en español es justamente porque si alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias por la amabilidad de traducir mi correo. saludos, bye bye On 5/5/07, Tom Rymes [EMAIL PROTECTED] wrote: On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He is expecting a price that would correspond with a used card. (In other words, cheap) Rodrigo: 1.) ¿Donde estás? ¿Cómo podria alguien dar un precio sin saber donde tendria que mandarlo? ¿España? ¿Puerto Rico? ¿Argentina? 2.) Si no hablas Inglés, seria mejor buscar una lista de Asterisk en Español, porque la mayoria de las personas aqui no hablen Español. Tom___ --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] Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-
I need information about integration of Asterisk and Avaya 70XX, if you have a configuration this sistems, please send me the informatión. Sorry for my english, i speeck spanish. Atte, Rodrigo Mercado Sepúlveda. On 7/20/06, Maxx Lobo [EMAIL PROTECTED] wrote: An update:I've found that I can leave the TE110P card in the server, unload themodule and issue an 'amportal restart' - this brings the IVR/meetme/internal directory voice prompts all back again.So it looks like the issue is directly related to the TE110P module(wcte11xp) in kernel 2.6.9.34-0.2 with CentOS 4.3. Anyone elseexperience this issue or have any suggestions based on this new information? Thanks---MaxxMaxx wrote: I've developed an issue I was hoping someone on this list might be able to shed some light on. I have two identical asterisk servers: CentOS 4.3, single TE110P card. I've also got a SIP trunk configured with TelaSIP for in/out calling. Both boxes have been running the same versions of Asterisk, Zaptel, etc. In short, no difference between the two. One box is the primary system, all of the SIP clients on the network associate to it. I'll call this one PBX01. The other box, PBX02, is a test platform to install new releases on, and to make sure everything works as before before any upgrades take place (and new bugs get introduced ;-)) The plan was to get the system up and running using a VoIP trunk, then cut the T1 line over to the Digium TE110P card and keep the VoIP trunk as a backup for emergencies. Originally both boxes were running [EMAIL PROTECTED] 2.7, and there were no issues. I upgraded to [EMAIL PROTECTED] 2.8, and got hit by the spinlock.h kernel issue as a result of a yum update. No big deal, ran the fixes, everything seemed to work okay. VoIP trunk still in place. There were no errors with zaptel, and the module for the TE110P was detected on boot. Absolutely no asterisk errors either. Then I noticed that any voice menu-related items were broken - IVR menu was broken when outside callers called the VoIP trunk number, Meetme conferencing calls were also 'broken' in that there was nothing but silence when I called a working conference number, and internal directory calls to were also receiving dead air. I tried a number of things, none of which worked - so on a whim, I removed the TE110P. Amazingly, everything worked fine when the box restarted. Meetme conferencing, external calls, internal directory, everything. Thinking that the card may have developed hardware issues, I tried the TE110P from the standby server (PBX02). The same issues returned right away. So then I tried swapping servers, to make sure there weren't some hardware issues cropping up. Same result again - worked fine without the TE110P, did not work with it. Now I tried upgrading the test box to Trixbox 1.1, and upgrading the zaptel to 1.2.7 from source. Same issues all over again. I've seen the suggestions on Kennonsoft http://www.kennonsoft.org/2006/07/trixbox-v10-quickie-how-to.htm about the meetme conferencing directory permissions, and I've made all of those changes. It didn't work before the changes or after - same problem all along. The modules are getting loaded fine as evidenced by lsmod: [EMAIL PROTECTED] asterisk]# lsmod ModuleSizeUsed by wcte11xp 304960 zaptel1967403 wcte11xp crc_ccitt 60811 zaptel No asterisk errors in the logs, either. No build errors when rebuilding zaptel-1.2.7. I've tried both kernel 2.6.9.34-0.1 and 2.6.9.34-0.2 (currently running the latter on both servers). I can post the full logs if necessary. The servers had sound cards in them, I removed these and made sure they were removed from both modprobe.conf and by kudzu. I didn't think this was the issue, but it was sound related and I am grasping at straws. Please tell me I'm missing something basic and what I can do to fix it. Without the TE110P cards I can't cut things over as I intended to, and I'm all out of ideas on this. Thanks- --Maxx ___ --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
[asterisk-users] Asterisk / Avaya 70XX
I need information about integration of Asterisk and Avaya 70XX , if you have a configuration this sistems, please send me the informatión. Sorry for my english, i speeck spanish. Atte, Rodrigo Mercado S. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding to mobile phone
I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank sorry for my english, i speek spanish only. atte,Rodrigo M On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote: is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Lito LampitocSent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] call forwarding to mobile phone Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanksLito ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk v/s other Telephonic plants
On 7/18/06, Rodrigo Mercado [EMAIL PROTECTED] wrote: I need information / documents or configurations of asterisk with other Telephonic head offices(plants), for your help , thank sorry for my english, i speek spanish only. atte,Rodrigo M On 7/18/06, Lito Lampitoc [EMAIL PROTECTED] wrote: is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network. On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Lito LampitocSent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] call forwarding to mobile phone Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanksLito ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
