search for: mixvoip

...................................................... I am using asterisk 1.4 and zaptel-1.4.12.1. I deployed quite alot of those devices but I do not see what I am doing wrong with this one. The only difference is that it has 2 USB ports for failover I guess. Best regards, Lo?c. -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

...t sn1:~# zaptel_hardware pci:0000:0b:08.0 wcte12xp+ d161:8000 Wildcard TE121 root at sn1:~# ztcfg -v Zaptel Version: SVN-branch-1.4-r4309 Echo Canceller: MG2 Is MG2 the correct one that I am supposed to use? Isn't there something better? Best regards, Loic Didelot. -- Lo?c DIDELOT MIXvoip S.a. ldidelot at mixvoip.com http://www.mixvoip.com

Hello, I played with the externalIVR application. So far I am able to read digits and play sound files. But how can I leave the application and continue in the dialplan so that I can execute other actions like going to voicemail, or ringing users etc.... Best regards, Lo?c. -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

.../bla7&SIP/loicvoip1_1&IAX2/loicvoip1_1&SIP/loicvoip1_1_a1 I tested on several asterisk 1.4 versions like 1.4.21*. Is this a bug or something like working as designed? Is there another possibility to monitor a bigger number of channels? Best regards, Loic Didelot -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

...ovided by http://www.api-digital.com - -->> - -->> > -->> > asterisk-users mailing list -->> > To UNSUBSCRIBE or update options visit: -->> > http://lists.digium.com/mailman/listinfo/asterisk-users -->> -- -->> Lo?c DIDELOT -->> MIXvoip S.a. -->> Tel: +352 20 3333 20 -->> Fax: +352 20 3333 90 -->> ldidelot at mixvoip.com -->> http://www.mixvoip.com -->> -->> -->> _______________________________________________ -->> -- Bandwidth and Colocation Provided by http://www.api-digi...

...ZAP-wedekindpbx en passthrough root at pbx1:~# /usr/share/zaptel/xpp_fxloader load --------- FIRMWARE LOADING: (load) [0 devices] Got all 0 devices Got all 0 devices --------- FIRMWARE IS LOADED Any help is appreciated. Best regards, Loic Didelot. -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

...ement manualy or to reserve some (trick and see), sometimes it's possible through some dip switches on the mother board, sometimes nothing todo except to change the MoBo... > > Good luck ! > Francois > > -----Message d'origine----- > De : Loic Didelot [mailto:ldidelot at mixvoip.com] > Envoy? : mercredi 26 ao?t 2009 12:49 > ? : f6hqz-m at hamwlan.net; Asterisk Users Mailing List - Non-Commercial > Discussion > Objet : Re: [asterisk-users] PRI worked fine for months, nowit stopps > working after 2-3 hours > > > Hello, > seems that the IRQ is sha...

Hi, I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 It says "Install libTiff >=3.8 and <4.0" Should you really use this agx-ast-addons to get app_rxfax and app-_txfax running with latest 1.4.22 ? If positive, should you take this libtiff warning into account ? If positive, where can you find such libtiff version as Debian repository (I didn't check alternate

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

...ed transfer. Of course, I mean an attended transfer by using the phones functionality, not something defined in asterisks features.conf. Any idea on how to achieve this, or any technical document from Siemens on on how this is support to work would help. Thanks in advance, Lo?c. -- Lo?c DIDELOT MIXvoip S.a. Tel: +352 20 3333 20 Fax: +352 20 3333 90 ldidelot at mixvoip.com http://www.mixvoip.com

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

...ovided by http://www.api-digital.com - -->> ?- -->> ?> -->> ?> asterisk-users mailing list -->> ?> To UNSUBSCRIBE or update options visit: -->> ?> ? ?http://lists.digium.com/mailman/listinfo/asterisk-users -->> ?-- -->> ?Lo?c DIDELOT -->> ?MIXvoip S.a. -->> ?Tel: +352 20 3333 20 -->> ?Fax: +352 20 3333 90 -->> ?ldidelot at mixvoip.com -->> ?http://www.mixvoip.com -->> -->> -->> ?_______________________________________________ -->> ?-- Bandwidth and Colocation Provided by http://www.api-digital....

...S-01/UNIT-1: Continue '/usr= /share/zaptel/init_card_1_30' Dec 30 21:42:36 wedekindpbx1 init_card_1_30: XBUS-01/UNIT-1: Ending '/usr/s= hare/zaptel/init_card_1_30' Dec 30 21:42:37 wedekindpbx1 kernel: [ 3775.622144] INFO-xpp: XBUS-01/XPD-1= 0: Initialized: FXS -- Lo=C3=AFc DIDELOT MIXvoip S.a. ldidelot at mixvoip.com http://www.mixvoip.com ----- Original Message ----- From: "Tzafrir Cohen" <tzafrir.cohen at xorcom.com> To: asterisk-users at lists.digium.com Sent: Tuesday, December 30, 2008 7:37:43 PM GMT +01:00 Amsterdam / Berlin /= Bern / Rome / Stockholm / Vienna...

I'm trying to use a Xorcom Astribank wth Ubuntu 8.04, but got no success. I can see the channel bank with lsusb, but when I tried to use zaptel_hardware, or when I try the /etc/init.d/script, they don't see my Channel Bank. I compiled the latest Zaptel 1.4.10, with Astribank's dependecies, fxload and libusb-dev. Anyone have a similiar experience ? Best Regards, -- Guilherme Loch G?es

Hello, I am having some troubles with Snom phones and maybe someone can help me. Let me say this: BLF and pickup works great with Polycomes and Grandstream etc... So I think my problem might not be Asterisk related but I am not 100% sure. The snom phones subscribe to my extensions (hint priority) as expected. The light blinks (ringing) or is turned on (in the call) as expected. My problem is to

Hello, I have BLF working on Snom phones. Ringing state (blinking) or "on the phone" state (solid) are working well. So the buttons are configured as "BLF" in the Snom webinterface. Now I would like to add another state for unavailable or dnd. In fact I would like to turn the LED red in the case the phone is not registered or the user pushed the DND button. So I though

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' I defined the extension s,h,i,t,T etc... in my context. Any

Hello, I am trying to update to dahdi 2.2.1. But somehow the xorcom (2BRI) device does not want to work. I pasted below the information I have. pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader reset pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[4937]: --------- FIRMWARE LOADING: (usb) [1 devices] ..Got all 1 devices 'xpp_fxloader'[5003]: ---------