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...talk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? > >Message: 10 >Date: Fri, 02 Mar 2007 19:07:41 +0200 >From: Cosmin Prund <cosmin@adicomsoft.ro> >Subject: Re: [asterisk-users] gtalktovoip and Asteirsk >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <45E859DD.8080103@adicomsoft.ro> >Content-Type: text/plain; charset="iso-8859-1" > &gt...

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

...lk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? > >Message: 10 >Date: Fri, 02 Mar 2007 19:07:41 +0200 >From: Cosmin Prund <cosmin@adicomsoft.ro> >Subject: Re: [asterisk-users] gtalktovoip and Asteirsk >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <45E859DD.8080103@adicomsoft.ro> >Content-Type: text/plain; charset="iso-8859-1" &g...

Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking

There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

..., immediately followed by two tones, and then they would be bridged to the client. Perhaps you're running MeetMe() with those join tones disabled? Check out the docs for MeetMe. I think it's option capital i, as in Iberia. On Mon, Jun 23, 2008 at 12:19 AM, Cosmin Prund <cosmin.prund at adicomsoft.ro> wrote: > Hello. It's been a while since I last posted (probably because my "*" works > just fine). I'm working on something to replace call queues in my own > application-specific way and I'm using MeetMe rooms to bridge agents and > clients and do other thi...

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

Hello everyone. Is it possible to do some very basic voice recognition from within Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I want to dial from my mobile phone. Dialing digits on my mobile phone while driving is not all that safe... Thanks for any input, Cosmin Prund

Hello everyone. I'm working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? I already tried using the local channel for dialing (so I can put in

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

The way I understand things, there's no way for a analog line to "reject" a call (give the caller an busy tone) if the line is not actually busy. Would a digital EURO-ISDN line give me this ability? Thanks, Cosmin Prund

Hello Gurus, here's my problem: I downloaded and installed Asterisk@Home (the version sporting Asterisk 1.2.4) and now I've got the following problem on the CLI output console (Alt+F9): Most text is fine, except something that looks to me like parameters. I don't really know how to explain this so I'm going to give an example: It show text like: <sample> Goto

I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hello everyone. What's better for Asterisk: have 2 distinct 100Mb network cards in the system, one on the "internet" and one on the "local net" OR have one 1000Mb network card with 2 separate VLAN's set up? It's a difficult decision because 2 cards are using 2 IRQ's etc but a single 1000Mb card might generate more PCI interrupts and get me into different

My Trendnet 26 port managed switch gave up on me so I'm shopping for a new switch. I learned the hard way NOT to trust marketing material from anyone so now I'm asking the list: what am I looking for in a managed, VoIP switch? P.S: For those that don't understand WHY I can't trust marketing material, let me tell you something about the Trendnet switch that's fast becoming

Hello everyone. I recently boght a Nokia BH900 headset and USB bluetooth dongle and I'd like to use them to make calls from a sofphone. I managed to this with boxe XTen-Lite and the Zoiper - but they both see the device as a simple sound card through the BlueSoleil drivers. While this is allmost usable, the headphone seems to be kept in "transmission" mode all the time and I get

Hello everyone. What I'm doing: I've made a replacement for app_queue that uses MeetMe to connect the calling party with the agents. When the call comes in it gets put into a MeetMe room with a nice AGI_BACKGROUND so the calling party can listen to music and announcements until an agent becomes available. So far everything works fine. Now I want to give the calling party an