similar to: Asterisk @home with IAX termination...

I'm setting up a small office with about 8 SIP phones. Incoming and outgoing lines will be through IAX. We would also like to use an analog line for 911. Is the TDM01B a good option for this kind of configuration? Are there gotchas I'm missing? Finally, we would like to be able to use analog fax machines in the office. Would it make more sense to purchase the TDM400 card with 1 FXO

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Hi, I am looking for voip termination all over the world especially based on IAX or SIP. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040124/25e16736/attachment.htm

Hi, I have a client who has asked me to look into the delivery of 30 second audio messages to a list of opt-in customers. Probably looking at about 5,000 messages a week over a 6 week period. I know that this would be a piece of cake to have someone develop but I thought I would ask here first if someone is already doing this and what they would charge to take this on as a hosted solution

Hey guys, I'm actively trying to get the "big" picture on how all this works and relates to each other. I've gone through some basic examples from the book and from the sample files just fine. Now, I've setup an account with a VOIP provider which does IAX termination (exgn.net) After getting an account and following their steps, I can make calls out using my IAX (cubix) and

I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination provider. Im tring to start a small Pre-paid long distance service. Thanks Erick

Hello, If I use an IAX termination service to connect outgoing VoIP calls to a PSTN will I have answer supervision so that my script won't initiate too early? Jason Wolfe jasonwolfe@bellsouth.net c (770) 561-6956 This e-mail transmission may contain information that is proprietary, privileged and/or confidential and is intended exclusively for the person(s) to whom it is addressed. Any use,

Hi: I am practicing with the attached shapefile and was wondering if I can get some help. Haven't used 'rgdal' and 'maptools' much but it appears to be a great way bring map data into R. Please take a look at the comments and let me know if I need to explain better what I am trying to accomplish. library(rgdal) library(maptools) library(ggplot2) dsn="C:/Documents and

Does anybody know of any commercial providers of IAX termination with DIDs in the Seattle, WA area? I believe the area codes are: 425, 206, 253 Failing any commercial providers, is there anybody in the seattle area running Asterisk with a PRI coming in who might be willing to sell me an IAX trunk with a DID in Seattle? -- ____________________________________________________________ Muiz

Does anybody know anyone who offers encrypted IAX termination at reasonable rates? I googled, searched the WIKI, but didn't find a whole lot of information. Thanks, David

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

Does anyone have any experience with any SIP or IAX providers that support E911? I'd love to convert entirely to Asterisk at my house, but the lack of emergency dialing has been a major hold-up for me. Thanks in advance for any suggestions! -- Kyle Sexton

I'm looking for IAX (preferred) or SIP termination in Israel, with DIDs. Can anyone give me a price, or better yet, point me to a resource for buying/selling minutes? Thanks, Mark

anyone know a company that will terminate did 831/408 area codes in california. FYI i already checked voicepulse, negative. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

If you connect an * system to a company like VoicePulse with there Connect service you are able to get incoming calls unlimited for 7.99/mo and outgoing at .0295/minute. VoicePulse also offers an unlimited business plan. Are you able to sign up with VoicePulse unlimited plan and connect an * system to it? Also, with VoicePulse connect you're able to make unlimited simultaneous calls

How about it. Anyone use these guys? Their rates look ok. I'm looking for alternatives to VoicePulse Connect that provide DIDs in Houston TX. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones,

I've installed Asterisk from the Asterisk @home distribution. Ultimately I will be using Asterisk for voicemail for about 150 users. Calls are (mostly) handled by a legacy PBX although we do have a couple of Cisco 1760 routers that connect a remote office. I've setup a SIP trunk that routes calls from Asterisk to the 1760, and that works fine. I've also configured one of the 1760s to

Hi Gurus. I searched the lists, wiki and the rest of the web but I still do not understand this. My Setup is as follows: [ISDN via chan_capi or IAX2 DiD Provider] => [* PBX] => [IAX2 Clients (Atcom AT-320ED)] I want to get callgroup/pickupgroup and callwaiting working on the IAX phones. Some web sources told me that this was not implemented, others say that the phone has to handle