Re: [asterisk-users] SPA400 and asterisk
You could use it as a usually FXO Gateway. I have tested and it works fine. 2007/6/12, MBIT Technologies [EMAIL PROTECTED]: Hi Guys I am just looking to see if you can help me. I have been investigating the SPA400 and it seems to run asterisk for the voicemail system. Does anyone know if it could be programmed to also talk to the FXO ports? ___ --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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] WiFi SIP phones
You could try, N80, N95 devices. It cost arround 300 dollars and works fine with SIP , Wifi and GSM. I have been trying for several weeks with Truphone, Gizmo, Asterisk and other providers my N80 IE, and it works perfectly Regarsd 2007/5/23, Chris Bagnall [EMAIL PROTECTED]: Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since then. What models are currently out there people would recommend I look at? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] test
ACK 2007/4/12, Razza [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 -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --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] spc.exe
Sipura Profile Compiler is only for ITSPs and agreements does not permit that Regards Andrew Joakimsen escribió: Does anyone have a copy of spc.exe they could send me? ___ --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] WRT54GP2 provisioning
If you are an ITSP provider, you could do with SPC tools (provided by Linksys to ITSPs) Regards Curt Shaffer escribió: Can anyone point me to a good source for provisioning WRT54GP2 from a central server? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Google talk and Asterisk 1.4
Check it! http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk Robert LaPoint escribió: Hello All Does anybody know where I can find information on configuring Asterisk 1.4 to work with Google talk. ___ --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] Google talk and Asterisk 1.4
What did not work? I made test under SVN Trunk and only have issues with audio behind NAT clients. You could check at bugs.digium.com Gtalk development state and bugs resolved. I did not make test with 1.4 beta 2 , so i could not help you more Regards Robert LaPoint escribió: I have already tried to follow this document but it did not work under 1.4, so I am just wondering if Google talk is even supported under asterisk 1.4 yet. Thanks Alberto -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Saturday, September 30, 2006 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google talk and Asterisk 1.4 Check it! http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk Robert LaPoint escribió: Hello All Does anybody know where I can find information on configuring Asterisk 1.4 to work with Google talk. ___ --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
Re: [asterisk-users] señalizacion te110p, signaling te110p
Maybe you could try an asterisk forum in spanish in order to get better results using your native language. DiegoF escribió: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? atentamente diego fernando güiza arce / hello to all, I have a doubt, ye I have solved some but others arrive, good since te110p had said I want to connect a PBX to one, the PBX offers señalizaciòn to me r2 European in cable rj45 or coaxial that type of signaling is used for the card te110p to me, in addition, some of those two types of connections serves to me or I must buy some adapter. I saw something that tapeworm that to use a balum, is necessary for anyone of the two connections. as type of connection they recommend to me but? I need to know something but on the PBX to form in te110p? kindly diego fernando güiza arce // -- // DiegoF // // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados // // Se han fijado que cuando estan solos...no hay nadie??? // // Cada vez que me siento a pensar, lo unico que consigo es sentarme. // ___ --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] Linksys SPA400
It has a proxy inside (asterisk), you could register to it as a regular sip proxy, so you could use it. Carlos Chavez escribió: Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. ___ --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] Linksys SPA400
Not True! You could register against it any spa product, and also asterisk. Cory Andrews escribió: It's only designed for use with the SPA-9000 (LVS-9000) product ecosystem. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, September 21, 2006 1:00 PM To: Asterisk Subject: [asterisk-users] Linksys SPA400 Does anyone know if the Linksys SPA400 is compatible with Asterisk or is it only for the SPA9000 system? It is interesting because it is a 4 FXO ATA at a reasonable price. ___ --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] Attended Transfer Asterisk 1.2.11
Im updating from 1.2.9.1 to 1.2.11 and im having a issue with attendad transfer via SPA 941 that i did not have with 1.2.9.1. I get this message on Cli log. Sep 14 16:09:52 NOTICE[5780]: chan_sip.c:6897 get_refer_info: Supervised transfer requested, but unable to find callid '[EMAIL PROTECTED]'. Both legs must reside on Asterisk box to transfer at this time. I have canreinvite=yes on all extensions, and tried with canreinvite=no, but same happens. When i press tranfer, i could talk with destination extension, but when i transfer call, it hangups both sides. Regards -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Choppy MOH (Cisco gateway)
VAD maybe was caussing this. Regards Zeeshan Zakaria escribió: Actually the problem was somewhere in the Cisco equipment, as the service provider has confirmed. Some option in their device to conserve bandwidth by compressing voice data was causing this choppyness. As they've turned this option off now, MoH works perfect. ___ --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 logging per day
You could use logrotate or you could configure your cron to send asterisk -x logger rotate, which it will do what you want. Regards Christophorus Laube escribió: Hi list, I am searching for a possibility to let my * log per day. So that a new logfile is taken every night at midnight, with the date in the file name. Is there a way to do so? Does anyone of you has tried that before? Regards, Christophorus ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Whcih phones are better for mass deployment
I prefer Linksys ones. Spa 9xx series, are great, and provisioning from Sipura/Linksys is much better than PA1628 (Unencrypted). Supports https,tftp and http. With Encryption. Vonage use it. Regards Thomas Kenyon escribió: Michael Graves wrote: Polycom Aastra are both great in this manner. Even Cheapo PA168S phones will remotely update their configurations from a simple handful of files from a tftp, ftp or http url. (which is admittedly only simple to do with 30 phones simultaneously if you use PoE). I'm very surprised if a Grandstream can't manage this. ___ --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] QUINTUM TENOR ASM200 Configuration
If you want to answer directly to him, try Reply to all, and delete [EMAIL PROTECTED] email address. It is not so had to do. FRANCISCO PEREZ-LANDAETA escribió: Hi, this message is for Steve. Sorry for replying to the digest. It wasn't my intention. I would appreciate if you can guide as to how make the tenor asm200 work with asterisk. I am using asterisk at home. I guess my problem is configuring the tenor so that it is recognized and can take calls from asterisk (both ways). If you can help me out and send me a sample config i would be very thankful. My config is an asterisk at home box, and i wish to be able to have my quintum register to the asterisk. Both devices will be in different lans. My intention is to be able to call my asterisk box (in my home), using my quintum box (in my office) and vice versa. thanks, Francisco _ Get the new Windows Live Messenger! http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ --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] Call Processing Slow 11 seconds
Yes you could script a dialplan putting ... and S0 (zero) at the end. An example : (xxS0) It will dial 6 digits directly when you enter the 6th. You could learn how to adapt your Linksys dialplan looking this wiki. http://voip.wikispaces.com/ [EMAIL PROTECTED] escribió: Yes that works. I'm using Linksys adapter, is there a code I can put in the dial plan to prevent users from putting # after the number? I have a lot of people on the server and cannot ask them all to be pushing # after every call. Thanks for the tip and any help will be appreciated. -- Original message -- From: G.Jacobsen [EMAIL PROTECTED] In case you use an adapter or voip phone: Did you try to press hash # after the number ? - then the adapter/voip phone dials immediately and doesnt wait for the next digit timeout. Cheers Gerry -Original Message *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] *Sent:* Samstag, 9. September 2006 15:15 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Call Processing Slow 11 seconds I'm having some slowness issue with Asterisk. When a number is dialed it takes 11 seconds before it rings out. I been considering using openser for the call processing and leaving asterisk for voicemail and conference bridge. I get a dialtone rightaway when the receiver is picked up but after dialing the number but within asterisk extensions and pstn numbers takes 11 seconds before ringing out. Anyone else experiencing this. I use Asterisk 1.2.3 Asunto: RE: [asterisk-users] Call Processing Slow 11 seconds De: G.Jacobsen [EMAIL PROTECTED] Fecha: Sat, 9 Sep 2006 17:20:05 + Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] How to Install H323
I think remember there is a readme on /docs that talks about chan_h323.Check it ! Anyway you could try too at voip.info dot org. Regards Wasif escribió: Hello, Could anyone tell me how to install/configure H323 with Asterisk 1.2.11 . Thanks Wazb ___ --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] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be xfer'ed via the soft button on the phone itself, it seems that if you hit the button twice in quick succession, there is no problem (effectively a blind transfer), if then I try to tell the other extension that Joe is calling to sell you a fridge and hit xfer, the calling party cannot hear what that person at the extension is saying. Sometimes the tables are fully turned, the caller can hear, but the operator can't hear a thing. One thing's for sure, if you hit the button quickly (blind transfer) it works no problem at all. This is what I see asterisk saying when I transfer the call unsuccessfully. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' I've looked at the macro with a fine tooth comb, I cannot see any problems with it whatsoever, (though that doesn't mean that my ignorance isn't getting in the way). I found some mention on the digium mantis bug tracker, here's the link: http://bugs.digium.com/view.php?id=7421 Before I try and patch the source (which I'm hesitant to do since I run the debian packages), is there another solution or maybe an unidentified issue that I haven't been able to decipher? If there's more information that I can provide to solve this problem, I'd be happy to do so. Thank you. ___ --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] Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
I use canreinvite=yes in my config files, and it does work, so maybe its a spa 941 misconfiguration. I think if nat=no sometime it has problems if you are behind NAT, but under same network it must not fail. Which firmware are you running on spas? Dan Serban escribió: Alberto Sagredo wrote: I have 30 SPA 941 phones with Asterisk 1.2.7.1 and 1.2.9 and it works fine. Are you canreinvite=yes ?. I have not been notice any problem related to transferring calls (blind and attended) Thank you for your response, it gave me a nudge to check the configuration in the sip.conf file. It seems that if I set canreinvite=no for every SIP peer, it works! And I have found no other adverse effects. Strange issue... Regards Dan Serban escribió: I have a system running Asterisk 1.2.10 (Debian packaging) and about 50 Linksys SPA-942 phones, after the initial config and mass deployment of the phones everything looks like it's configured well. When an incoming call is answered and then attempted to be xfer'ed via the soft button on the phone itself, it seems that if you hit the button twice in quick succession, there is no problem (effectively a blind transfer), if then I try to tell the other extension that Joe is calling to sell you a fridge and hit xfer, the calling party cannot hear what that person at the extension is saying. Sometimes the tables are fully turned, the caller can hear, but the operator can't hear a thing. One thing's for sure, if you hit the button quickly (blind transfer) it works no problem at all. This is what I see asterisk saying when I transfer the call unsuccessfully. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/82-006d42a0ZOMBIE' I've looked at the macro with a fine tooth comb, I cannot see any problems with it whatsoever, (though that doesn't mean that my ignorance isn't getting in the way). I found some mention on the digium mantis bug tracker, here's the link: http://bugs.digium.com/view.php?id=7421 Before I try and patch the source (which I'm hesitant to do since I run the debian packages), is there another solution or maybe an unidentified issue that I haven't been able to decipher? If there's more information that I can provide to solve this problem, I'd be happy to do so. Thank you. ___ --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
Re: [asterisk-users] Linksys SPA-3000 Administration Guide
This Guide is offered as i know only to ITSP and large distributors not to end-users. You could find a User Guide for SPA 3102 at Linksys Website. Regards Marcos Rubino escribió: Anybody have a recent copy of the Admin Guide (not the user guide) for the SPA3000/3102? The only one I was able to find was a terribly written two year old one on the Sipura site[1] and Linksys says you have to be a Service Provider to get one from them. [1]http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf I am just a humble VOIP enthusiast, can anybody hook me up? Please CC me on the reply (or respond directly), I don't actively follow this list. Thanks. Marc __ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Choppy MOH (Cisco gateway)
VAD option on Cisco Gateways maube are causing this. Check This http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Its a feature , to have Zeeshan Zakaria escribió: My service provider had issue with his Cisco hardware when it came to MoH. They were new with Asterisk at that time. I told them many times that they had problem in their system, but they never agreed, until one day when one of their engineers figured out that the Cisco hardware was compressing the MoH data to conserve bandwidth, causing choppy MoH. That was some simple feature which he switched off and I didn't have MoH problem after that. I am not a Cisco expert, but those who are, may know what I am talking about. Zeeshan A Zakaria On 7/10/06, Bill Gibbs [EMAIL PROTECTED] wrote: Yes that is correct. Bill -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Monday, July 10, 2006 12:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) On Jul 10, 2006, at 4:49 AM, Bill Gibbs wrote: And of course I just found this article http://www.cisco.com/warp/public/788/voice-qos/hissing.html#topic3 Hope this helps some other people out as well! So was the fix to reconfigure your gateway to notuse VAD? Just want to be clear... Marty ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Encrypting the Conversation
Maybe in Asterisk 1.4 SecureRTP application would do that. Regards Henry J. Cobb escribió: Hi, Is it possible to encrypt the conversation between two parties on SIP,IAX or ZAP channels? Sure, setup a VPN. You can get a Linksys VPN router for less than $100 and run whatever protocol you like over your VPN. ___ --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 h323
Im using several Asterisk Box with chanh323 from asterisk, and it works fine. Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability. A fail (crash) last month with about 600 calls per day. Regards Alberto Sagredo hakem voip escribió: You can do this by installing a h323 module. Conversion works simetimes good, sometimes not good. H323 behaviour on asterosk with my experience with kind of unpredictable. 2006/6/20, Khaled Chehab [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hi Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works . Please if u know send me how to active h323 protocol or the conversion protocol Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/ -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Hakem Voip ___ --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] SPA-941 Disable call waiting or Disable Call waiting via asterisk
It has a conceptual problem i have notified several times to Cisco-Linksys. It could not be disabled, i have the same problem with my queue extensions, and the way to resolve has been to use call-limit=1 in extensions. i hope this helps. Tommaso Calosi escribió: I'm trying to disable call waiting for Linksys SPA-941, but unfortunately as far as I have seen, there are no parameters on the web interface regarding this feature. I just want callers to hear the busy tone when the called party is at the phone. Probably I can accomplish this by using the disable call waiting in asterisk as well, but I have not been able to find any documentation for this. I have found this http://www.voip-info.org/wiki/index.php?page=PBX+Disable+Call+Waiting about call waiting, but it's quite unusefull. Thanks Tommaso Calosi ___ --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 -- Alberto Sagredo I+D Area (Asterisk // Cisco-Linksys) Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] Compiling SVN Trunk
I have the same problem on some modules. For example app_math.so [app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728 __load_resource: missing mod_data for app_math.so Any help?. I have been looking , but nothing reasonable found. Thanks -- Alberto Sagredo ___ --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] Compiling SVN Trunk
Umm. Maybe i have left some asterisk 1.2.9.1 modules.. and it has not been replaced. By i made a make install after i compiled it, so it would be replaced?. I will check it. Thanks Joshua Colp escribió: Alberto Sagredo wrote: I have the same problem on some modules. For example app_math.so [app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728 __load_resource: missing mod_data for app_math.so Any help?. I have been looking , but nothing reasonable found. Thanks Have you wiped out /usr/lib/asterisk/modules? Sounds like you have some leftover modules as that was converted to a dialplan function and app_math taken away - in trunk. ___ --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] Compiling SVN Trunk
Thanks again. Sorry. Kevin P. Fleming escribió: - Alberto Sagredo [EMAIL PROTECTED] wrote: By i made a make install after i compiled it, so it would be replaced?. It doesn't get replaced, because the new version of Asterisk doesn't have that module any longer (it's been moved to a different module). In fact, the 'make install' process prints a HUGE warning to your console when it finishes if you have old modules in /usr/lib/asterisk/modules, but apparently you did not see it. ___ --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] Wanted: CISCO 186 ATAs
Why use it? It has been replaced by other Sipura/Linksys Stuff. Do u use SIP or H323...? James Ching escribió: Greetings, I'm looking to purchase 10 Cisco 186 ATAs. Please send me quantities available, payment methods and out the door pricing (shipping + tax + unit costs). ___ --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] statistics
Check these ones. http://www.micpc.com/qloganalyzer Queuemetrics http://www.ag-projects.com/CDRTool.html Take a look on voip-info.org for more options Regards issam escribió: Hello How can we do statistics with asterisk thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] British English voice files are ready for download
If you need g729 and g723 format, let me know and i could convert it to you. Tim Panton escribió: On 19 May 2006, at 17:05, Mark Phillips wrote: Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} It is great that there is an extra (British) choice for asterisk sounds. I downloaded the SLIN and I have a couple of remarks. 1) you can't use the SLIN directly on a non-intel machine - you may have to byte swap it first (took me a while to work out why I just got pulse modulated static on my NSLU2 home asterisk! (armv5teb) ) 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. I instantly felt like a child again, being told slowly and clearly what to do. The only reason I mention this is so people don't assume that these new recordings will always be the preferred offering to systems installed in the UK, it will depend on the image that a company wishes to present. Tim. Mark -- Mark Phillips [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 Tim Panton [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 ___ --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] change dchannel number
In the TimeSlotting procedure in E1 frames, you have TS0 for Framing and Syncronization, and TS16 for Signaling. Remember 32 channels (30+2). Jose Luis Garcia escribió: Hi: I'm using a te110p E1 card in Spain. There is way to change the dchan channel number? Here in Spain there is a voice company that works with dchannel in channel number 31. I must to set channels from 1 to 30 and dchan to 31 but when asterisk loads the chan_zap.c it says that channel 16 is reserve for dchan. Can you help me? Cheers. ___ --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] ATXFER
How many times do u need to repeat it?. You could change this info via web list manager. I think you need to read how to do that before sending 20 emails with same subject. [EMAIL PROTECTED] escribió: Please change the email address of [EMAIL PROTECTED] to [EMAIL PROTECTED] Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Having Rinback tone generation issues with 1.2.7.1
Today i move our central server to 1.2.7.1 , and im having some issues with SPA Phones and RinbackTone. Without r option, it also happens. Is having anyone this issue? I think it has not been changed anything sustancially to happen this to me. It is happening between extensiones (canreinvite=yes), outbount trunking (no gets rinback tone (generated by phone). 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] 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
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 don’t 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
Re: [Asterisk-Users] Auto Logout from queue
Via dialplan maybe? exten = xxx,1,Dial(SIP/101_Queue,20,tr) exten =xxx,2,RemoveQueueMember(Comercial_Queue,SIP/101_Queue,1) Kerry Garrison escribió: Yes, that is the functionality I am looking for, just not sure how exactly to pull that off. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Tuesday, April 25, 2006 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Auto Logout from queue Use the local channel to call the agent first, and if there is no answer, log them out. _ From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Tue 4/25/2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Auto Logout from queue i have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.techdatapros.com/ http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I get a recording from a CD to my asterisk digital assistant
You will need them in one of asterisk supported formats. wav, slin,gsm, g729, g723... Davi-Ann escribió: I got someone to record the messages we want for our auto-attendant menu on a CD. All I have to do not is to upload the files into the asterisk box, however the format is not recognized by the Asterisk box. Question 1) What formats should the sound file be, so I can upload it to my asterisk box? Thanks --Davi-Ann ___ --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] Quick question
You could try chan_oh323.so and chan_h323.so. I think also ooh323 supports inband DTMFs. Regards Alberto Sagredo Tomislav Parčina escribió: Is there any h323 channel driver that supports DTMF inband signalization? Thank you for your answer! -- Tomislav ___ --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] SPA-941/942 Bulk provisioning
You could find here an xml example to provisioning them. http://www.sipura.com/support/spa941faq/index.htm Kerry Garrison escribió: Has anyone got any information on bulk provisioning of Linksys SPA-941/94s? There is an overview in the admin guide but it refers to a different provisioning guide that I haven't found anywhere. Kerry Garrison Director of Technical Services **Tech Data Pros - Orange County's Mobile IT Service Provider **(949) 502-7819 x200 - //[EMAIL PROTECTED]// mailto:[EMAIL PROTECTED] //http://www.techdatapros.com// http://www.techdatapros.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39 Fax./Fax.: +34 91 661 9460 ___ --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] TO have ringing tone instead MOH
I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. 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] SMS in Spain (it seems Protocol 2)
Capatres released some time ago a solution with an ITSP. Maybe it could help http://blogs.capatres.com/index.php?op=ViewArticlearticleId=18blogId=1 Carles Pina i Estany escribió: Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems, Spain or Italy?) If nobody is able to send, is there more people interested on it? Or any project/person/firm trying to send SMS using Protocol 2? Thank you very much, PD: some guy from Asterisk-es said to me that it seems that Telefonica wants to implement Protocol 1 too... but I don't have any information about deadlines, etc... -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --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 behind a Firewall
If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --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] Ability to put call on hold via manager?
You could park it to parking extensiones. Does it help you? Steve Totaro escribió: Does anyone know if there is built in ability to put call on hold via the manager interface? Thanks, Steve Totaro http://www.asteriskhelpdesk.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Development news :: T38 passthrough support
Really interesting Olle We are expecting :) Miguel escribió: Olle E Johansson wrote: Asterisk won't be an T.38 endpoint, but will handle T.38 calls properly, regardless if the T.38 was offered in the original call setup, or if the caller suddenly sends a fax in the middle of a call (a re-invite). The requirement is that the incoming channel and the outbound channel both supports T38. If not, the call will be declined in a proper way. When this is tested and stable, work will continue to see if we can make Asterisk an T.38 endpoint. This is a very important addition to Asterisk. There is code for testing available. If you are interested, please check this URL in the bug tracker: http://bugs.digium.com/view.php?id=5090 I think this is a big step for Asterisk. Do you? If so, don't forget to say thank you to Steve Underwood - Coppice! Have a nice weekend! /Olle these are great news Olle, this is what im waiting for, keep the excelent work Steve, thank you very much . ___ --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] Can log into the mailbox from Soft-phone , but not from Hardware Phone
I suppose you are using 1.2.4 asterisk version Maybe is not sending dtmf tones as rfc2833 and inband mode is not being detected by your asterisk box. Im a wrong? Could you try to configure dtmf tones on your softphone? John Joseph escribió: Hi I am using asterisk 1.4 on RHEL4 I am sending this mail to the mailing list , to enquire wheter any one had faced simillar problem which I am facing now I am facing a problem which is not able to solve or understand , the problem is that I cannot log into the mailbox from a VoIP hardware phone , while I am able to login to the mail box using soft-phone for the same users Has anyone faced this kind of problems for mail “ Can log into the mailbox from Soft-phone , but not from Hardware Phone “ I am using hardware phone from grandstream Budge Tone -100 and another D-Link phone DPF-140S Would like to get feed-back Thanks Joseph John ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Authenticated SIP NOtify with 1.2.4?
I have been working with authenticated notifys for auto resync my autoprovisined devices. But it seems to stop the state machine, and when Endpoint answers 401 Unauthorized, the Sip Notify command from cli, does not answer with a Authenticated Notify? Have i misconfigured something? 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
[Asterisk-Users] Limiting Sip Calls ?
Is there any way not using group count, to limit calls received by every endpoint SIP?.. Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch. Is there another command to do that? Regards Alberto ___ --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] OH323 Peer
An easy way to do that, if you do not neet to register on a gkp, its doing a dial OH323/ipgateway:port Did you try this? Abdul Lateef escribió: Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry with all feature. Please let me know how i can add H.323 GW type peer? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] R2 implementation problem
I hope this link will help you. http://zarzamora.com.mx/asterisk/17 Regards Manuel Marin Garcia escribió: I have a TE110P connected to a Telmex E1 circuit with R2 signaling. Asterisk version= 1.0.10 Zaptel= 1.0.1 Spandsp=0.0.3pre6 Unicall= 0.0.3pre8 *zaptel.conf span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 dchan=16 loadzone = us defaultzone=us *unicall.conf immediate=no loglevel=255 protocolclass=mfcr2 protocolvariant=mx,10,4 protocolend=cpe group = 1 context = telmex channel = 1-15 channel = 17-31 * *chan_unicall is compiled without any problems and when asterisk starts I see all channels in idle state. The problem is that I am unable to make or receive calls. When there is an incoming call I see the following Incoming Call (I receive 10 ANI digits and 4 DNIS digits) I suppose to receive 0875 for DNIS Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 - 0001 [1/ 1/Idle /Idle ] Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 Detected Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 Making a new call with CRN 32769 Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 1101 - [2/ 2/Idle /Idle ] Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:2865 handle_uc_event: Unicall/25 event Detected Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 - 0 on [2/ 2/Seize ack /Seize ack] Jan 31 10:08:48 WARNING[4293]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/25 1 on - [2/ 2/Seize ack /Seize ack] There is a 9 or 10 seconds pause and the I receive the following message. There is a busy tone in the caller side after the message Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 - 1001 [2/ 2/Group A /DNIS request ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 - 0 off [2/ 800/Clear fwd /DNIS request ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 1 off - [2/ 800/Clear fwd /DNIS request ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 R2 prot. err. [2/ 800/Clear fwd /DNIS request ] cause 32774 - Invalid state Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 1001 - [1/ 1/Idle /Idle ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: Unicall/27 event Far end disconnected Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:3198 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 Call control(6) Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 Drop call(cause=Normal Clearing [16]) Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/27 1101 - [1/ 1/Idle /Idle ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: Unicall/27 event Protocol failure -- Unicall/27 protocol error. Cause 32774 Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 - 0001 [1/ 1/Idle /Idle ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 Detected Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 Making a new call with CRN 32769 Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 1101 - [2/ 2/Idle /Idle ] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:2865 handle_uc_event: Unicall/28 event Detected Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 - 0 on [2/ 2/Seize ack /Seize ack] Jan 31 10:10:34 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 1 on - [2/ 2/Seize ack /Seize ack] Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 - 0 off [2/ 2/Group A /DNIS request ] Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 1 off - [2/ 2/Group A /DNIS request ] Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 - 8 on [2/ 2/Group A /DNIS request ] Jan 31 10:10:35 WARNING[4311]: chan_unicall.c:704 unicall_report: MFC/R2 UniCall/28 1 on - [2/ 2/Group A /DNIS request ] *NOTE*: I Already tried changing values for DNIS and ANIS and same problem I tried with
[Asterisk-Users] About Meetme and CDRcustom
Hi all. Does anyone know how to cdr the meetme conference number that the person who enter called?. I did not find the variable and, last_data, seems not to give me the correct info. Regards -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 757 139 Fax./Fax.: +34 91 661 9460 ___ --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] About Extensions
Im trying to detect before entering in Meetme , which dtmf has been entered. I did a Background(file) and go to a context where i define a exten = _X.,1,Meetme() I have detected that with (1.2.1) when 1 is entered and conference 1 must be created, extensions say it is not possible and gave a fail. Other case as 01,0001, 1001,etc, works fine. What could be wrong?. Is there any other way to do that. I want to to detect # as send key, but with background and exten it seems not to work. Regards Alberto ___ --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 always uses 127.0.0.1 address
Maybe you have not configured correcly your sip.conf externip=your_external_ip try this RumaTech escribió: Hi, all Can someone tell me where to tell asterisk no to use 127.0.0.1 IP (localhost)? When I am registering with VoIP providers, they get my info as [EMAIL PROTECTED] (This is SIP registration). Also, in SIP logs, when calling I am getting things like this: Executing SetCallerID(SIP/phone2-22c3, CID Name CIDNUMBER) in new stack -- Executing Dial(SIP/phone2-22c3, SIP/sipnet/84959741926) in new stack We're at 127.0.0.1 port 18900 ANy help is appreciated, Thanks, Rudolf ___ --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] Easy to Access Telephone Directory AGI
Really interesting. Thanks Hannes!! Hannes Vogel wrote: I've written myself a easy to use telephone directory which I use at home and thought it may be of interrest to others. The purpose of this agi script is to provide an online telephone directory that can be easily accessed using the numbers on the phone dial pad. You select entries by spelling out the name of the person you want to contact using the phone dial pad. Now this is normally pretty labourious so the script provides a few shortcuts to make things easier. The best way to illustrate this is by example: Say you want to phone John Smith: - You would start by typing 5, this would find all entries that start with j,k or l. - Next you would type 6 which would narrow down the selection to all tries starting with either j, k or l followed by either m, n or o. - You continue to spell out the name in this fashion (4 = gHi, 6 = mnO, etc) until either a distinct match is found in the direcotry or the number of matches is 9 or less. If a distinct match is found the number associated with the name is returned and can be dialed. If the number of matches is 9 or less you can have an IVR menu containing the matching names built on the fly and you will be prompted to select a name (e.g. Press 1 for John Smith, Press 2 for John Doe etc). Once a name is selected the number associated with the name is returned and can be dialed. Now you might think that this is still pretty laborious but in fact you usually only have to spell out the first few letter of the first name and the last name to get a good match. Other feature include: - Being able to jump to the last name without having to finish spelling out the first name (i.e. Press 0 to skip to the last name) - Multiple numbers can be associated with a name. In this case you will be prompted to select which number you wanted returned for dialing e.g. Press 1 for Home, Press 2 for Business, etc) - Undo last typed entry in case you misstyped something - Wildcard matching (Press 1 to match any letter) - IVR menus built on the fly so you do not need to prerecord anything - IVR menus cached (the more you use it the quicker it gets) - Returns the selected number in the variable DIRNUMBER The code can be found in the Digium Asterisk Users Forum (I was not sure if I should post approx 900 lines of code to this list) http://forums.digium.com/viewtopic.php?t=3727 I can also send it direct if anyone is interrested. ___ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de ___ --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] How to Unload app_rxfax.so
Yes, you could do that making some changes on modules.conf noload = app_rxfax.so Regards Alberto Nitesh Divecha wrote: Hello All, Dunno what happen but Asterisk is refusing to start... Went over the log and found out that app_rxfax.so is failing to load. Jan 7 11:57:28 VERBOSE[4320] logger.c: [app_rxfax.so]Jan 7 11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/ app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 7 11:57:28 WARNING[4320] loader.c: Loading module app_rxfax.so failed! Is there any way to bypass this module and start Asterisk... I think it was a bad idea to compile Asterisk with fax capability... Thanks, Neal ___ --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 Call Forwarding
You need to manage this variable on Asterisk DB in order to make call
forwarding.
It must be done in extensions.conf . In voip-info you could find how to
do that.
Androtech wrote:
Hi,
I would like to forward a calling
from a specific number to an extension.
The dialplan syntax should be:
exten=_*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten=_*21*X.,2,Hangup
In
my case, the phone number to forward is 3473774567, and the extension
is 105, hence the syntax should be:
exten=3473774567,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:105})
exten=3473774567,2,Hangup
but Asterisk does not forward the number
to the extension 105.
Any ideas?
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--
Alberto Sagredo
Departamento Tcnico
Peoplecall
Email : [EMAIL PROTECTED]
Blog: http://www.voip-novatos.es
Tel./Ph. : +34 91 120 5080
Tel. Dir./Dir. Ph.: 700 755 048
Fax./Fax.: +34 91 661 9460
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[Asterisk-Users] Fast AGi Variables
Has anyone an example to pass variables to a fagi script? I have succesfull made some examples with traditional AGIs, but i could not find a way to do with FastAGI. Regards -- Alberto Sagredo Departamento Técnico Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voip-novatos.es Tel./Ph. : +34 91 120 5080 Tel. Dir./Dir. Ph.: 700 755 048 Fax./Fax.: +34 91 661 9460 ___ --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] Too high volume on Music on Hold
Hi all. I have an asterisk box on gentoo , and when i try to play MOH, it get too much volume. At a point that it could damage my ear system :) If i normalize the music, decreasing the volume, it normalizes again and play at a volume that i could not use. What could it be wrong?. In other * box with gentoo too, it does not happen. Regards Alberto ___ --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 + H323 + 723
Hi, I had the same troubles too. It does not recognise correctly g723 with oh323. With h323 i have dtmf rfc2833 issues but g723 and 729 are transported correctly via H323 capabilities. So, let make a try with h323 included in asterisk branch, not the oh323 Kanishka Somaratne wrote: Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani ___ --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
