similar to: Fwd: Which ATA device do you recommend?

Hi all, I've bought a TE110P, and received it today. So i decided to install Asterisk@home 2.7 with this card. In the past i had experiencies with X100P (clone card) and it never take me so long to reboot the machine.... Machine: P4- 2,8Ghz 1GRAM TE110P What could be wrong? Best regards, Marco Mouta

I know this is slightly off topic, but I was wondering if anyone can help with a problem getting my PAP2's to connect to Asterisk. I use a provisioning file, and I recently re-wrote the files for each PAP2. I had a small typo and the PAPs logged it as a corrupt file. I corrected the file, however, Line 1 on both of the PAP2's now wont register. Line 2 works fine though. I've done the

Hi List; Is Digium the best telephony cards to be used with Asterisk? The prices are some how high, any suggestion? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

Fry's Electronics has a new Linksys 2 line ATA box for sale for $59.99 retail. They have a version with a router for $89.99. We picked the non-router version up and it seems to be a rebadged Sipura SPA-2000. The box has a Vonage service package inside as well, but it does work with other services. The box also has a "User Guide" meant for end-users that is very well written [no

We have some extensions in our dialplan that start with a star. We can dial them from Zap phones and SIP phones, but not from phones connected to a PAP2. After the user presses star follwed by two digits (our extensions are dialed with star followed by three digits) he hears a fast-busy that comes from the PAP2, not from Asterisk. This also happens with the builtin *8 (call pickup). In

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

Hi all, I've the current scenario: User "A" - Zaptel call incoming in my Asterisk to my SIP user "B". "B" gets the Call. "A" says : "B" i would like to call PSTN user "C" "B" places a call to user "C" and asks if "C" wants the call from "A". "C" says yes i want, then B needs to

Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison "Transfer" the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call This is timeout

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Hi all, I've a Planet VIP-150T VoIP Hardphone connected to Asterisk. When I'm in a call and i press Hold button, the other party starts listening Music on Hold but then when i press the button again to get the call back it doesn't work! I've checked asterisk CLI: -- Stopped music on hold on Zap/1-1 -- Started music on hold, class 'default', on Zap/1-1 --

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works?

My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume

Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta

Hi all, I've 2 * tryning to connect each other Server A is already registred on server B But server B never registers in server A I always get this: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 00018ms SCall: 00004 DCall: 00003 [XXX.XXX.XXX.XX:4569] CAUSE : Registration Refused CAUSE CODE : 29 Any tip? Best regards, Marco Mouta

HI all, My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Best regards, Marco Mouta