similar to: Push-To-Talk?

Hi, i'm trying to get PoC on Nokia Phones to work with asterisk. I think the store-and-forward part could easy be done in the dialplan, but i can't even get the handset to register with asterisk (authentication failed). I'd try'd to find the difference between "pure sip" and PoC-SIP, but didn't suceed. Has someone an idea how to get this to work? Regards, Jay

Hi, i have some spare (read: Boss get's a new one every few month ;)) Android Phones laying around. Does someone know a way of using them as a mobile gateway for asterisk? I could not find any SIP-Gateway in the Market, and i don't think it's possible to use the GSM Audio directly with something like chan_datacard... Regards, Jay -------------- next part -------------- An HTML

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

Hello if someone found some method to authentificate to asterisk with nokia push to talk clients please send me all your documentations and the tests results, I really need this for a project of main and i wanna dig deeper to solve this mister. Thank you guys for your cooperation. Alex i put you at cc because i know you find this interesting too and maybe meanwhile you already know more about

Hiya, SIP Messaging is implemented in asterisk-10... >The only documentation I can find talks about a patch and is pretty old:>http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging <http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging> Like anything on voip-info.org it's horrible outdated. I think there's a documentation for the message-routing in docs.... Regards

I am talking about the SIP. Now the new mobiles (Nokia, Erecson, Panasonic, iPot, ... etc) all of them support SIP capability. They are able to register to any SIP server (by giving the IP address, username and password). Fring is one of the software that can be installed on the mobile devices and can register on the SIP servers. BUT, the new mobiles currently come with built in SIP (no need to

Hi, I'm looking for a reliable, reasonably-priced, single-channel interface between * and US GSM. The VOIP GSM Gateways listed at http://www.voip-info.org/wiki-VOIP+GSM+Gateways (VoiceBlue, QUTEX) are multichannel systems, very expensive ($2500 or more). Next step down, there are various Fixed Cellular Terminal (FCT) or Fixed Wireless Terminal (FWT) devices. These typically have an FXS

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

Hi, anyone knows an open OMA DM tool that would be able to configure Nokia phones (mainly the sip-stuff of the e-series) for use with asterisk? Anything open i could find was the device manager from funambol, which was last updated in 2006 :-( Regards, Jay -------------- next part -------------- An HTML attachment was scrubbed... URL:

Maybe someone here has found a good solution to this problem. I voulenteer with a local Search And Rescue unit and I was speaking with the senior members about how they interface the command trailer PBX with the PSTN or cellular networks when they are on scene at a remote location. Turns out they don't. Thus that got me to thinking about how one would get Asterisk to interface with a

Hi All, I would like to ask for some advice how to solve following situation: We have to record and decode isdn PROGRESS message and when particular message is found call should be hang up and dialplan should continue. So far we have come up with two ways we think solve the problem and would very much appreciate to hear your opinion. 1) add special progress extension to dialplan so we can

Ventrilo in version 2.1.4 work properly good under wine. I talk with friend without any problems with sound, but there is one with "Push-to-talk" function (hotkey). When frame with ventrilo have lost focus i can't use this function. I spent a lot of time to googled about it and I found program called xbindkeys. I got idea, that I can use this program to send keystroke to ventrilo.

I been looking at www.estara.com with a product "Push to Talk", I'd really love to be able to do that with *, my daughter is an export sales rep. for a small company here in France, if I could put a facility like that in a vcard type format or on the web page..... -- Dave Cotton <dcotton@linuxautrement.com>

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

Hello all, Can asterisk be used to provide PTT feature? Anyone using or tested it? Thanks: -AEE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100911/49983cdc/attachment.htm

Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users