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http://www.deviantsart.com/2su095m.png marcotasto at libero.it 3/14/2015 5:01:37 AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150314/28e43fc5/attachment.html>

Hi All, sorry if I post again this e-mail but I think the first one was lost. I don't know if this is OT but I'm working in my spare time at a small hardware project that match to what's requested below. It's a board with Input/Output capabilities and 10Mbps ethernet interface. It has Microchip software TCP/IP stack on it. Being at a very beginning stage, you can see a little

Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message.

JABBER: gtalk_account OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0'> localhost*CLI> jabber show tes JABBER: gtalk_account INCOMING: <?xml version="1.0" encoding="UTF-8"?><stream:stream from="gmail.com"

http://bwa-surma.org/pndyv/jzvmnxmtdohwbfhplindsrf.mpbpyaydmvosnbmgs marcotasto at libero.it 1/25/2015 10:15:09 AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150125/be3cb925/attachment.html>

...------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ------------------------------ Message: 9 Date: Mon, 09 Mar 2009 17:08:28 +0100 From: Marco Signorini <marcotasto at libero.it> Subject: Re: [asterisk-users] Faxing success rate on PRI To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <49B53EFC.8060706 at libero.it> Content-Type: text/plain; charset=3DISO-8859-1 Thanks Doug and Lee, yo...

Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems), it sounds like it is the "bees knees" for sorting out echo problems with cards like the x100p. Has anyone

Hi All, I'm having a strange problem and I'm not able to understand what's happening. I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine. They are linked together through localhost. I've turned qualify on for the iax peer. Periodically I've this message: [Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer: Peer 'iaxfax' is now

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

...in progress call is > received. > > Any ideas? show application playback Check the "noanswer" option /O :-) ------ Check our new SIP router class: http://edvina.net ------------------------------ Message: 3 Date: Mon, 09 Mar 2009 16:13:17 +0100 From: Marco Signorini <marcotasto at libero.it> Subject: Re: [asterisk-users] Faxing success rate on PRI To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <49B5320D.7080902 at libero.it> Content-Type: text/plain; charset=ISO-8859-1 Hi Gordon, thank you for...

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to "upgrade" to feature sets. Your door openner is a piece of cake. 1. Create an option in your dialplan only in the "from-access-door" context that reads

Ciao, ho recentemente acquistato un metodo molto costoso per guadagnare su Internet. Il metodo funziona davvero!!! Ora, in un giorno guadagno pi? denaro di quanto molti ne guadagnano in un mese. Le mie statistiche per oggi. Ho guadagnato _570 in 40 minuti. Fico, vero? Questo metodo funzioner? per circa 2 mesi. Io ho 2 inviti gratis; vorrei condividerne uno con te, solo non dirlo a

Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info and old firmware versions at this

Hi List. I'm running the patlooptest program I've found in dahdi_tools 2.1.0.2. The target is a TE121P board with a loopback cable inserted on the socket. I suppose that the loopback is working fine because I'm able to see the green led on and dahdi_tool reports no errors. When I run the patlooptest I've a lot of errors (the received values are completely different than the

Hi All, I'm trying to identify a new server as a replacement for what our customer actually has (DELL PowerEdge 860). The server will mount the Digium board TE121, we already have, with echo cancel onboard. I need to know if someone could suggest a new server that's compatible with this board. With "compatible" I mean that's not having any problem like IRQ sharing, IRQ miss

Hi List. I've bought a new server for my home asterisk installation and I'm trying to install asterisk with dahdi drivers and OSLEC. To do that I've got the svn dahdi-linux trunk revision 5366 and the echo subproject from the 2.6.28-rc6 Linux kernel sources. As reported in the dadhi README file I've uncommented out the two OSLEC related lines at Kbuild file in the

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call