similar to: Sipura 2100 and Asterisk and Fax

Hello Guys I have 2 azacall 200 as extensions for a Internet public asterisk box , I just want to connect point A with point B I have tried canreinvite= yes / no Nat=yes/no Nat=1/0 commented canreinvite commented Nat, I have no audio with other azacall or to other sip device in the same asterisk. Note: other devices like grandstream , siipura, Virbiage, and xtenworks perfectly ,, just

I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

I tried google and sixtel.com - couldn't find a web page for Sixtel - would some kind soul point me in the proper direction? ============================================================================ ====== Date: Mon, 13 Dec 2004 16:51:47 -0800 (PST) From: Steve Edwards <asterisk.org@sedwards.com> Subject: Re: [Asterisk-Users] IAX.cc / Sixtel? To: Asterisk Users Mailing List -

Hi, How is iax.cc / sixTel to be configured as a termination provider in asterisk@home? The iax.cc / sixTel instructions tell you to do this: iax.conf: ------------ [sixTel] type = friend host = iax2.sixtel.net context = inbound secret = mypassword allow = all extensions.conf: -------------------------------- exten =>

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hi, I've tried to make toll-free DID work for the last 2-3 weeks. Apparently only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody else just gets a busy signal. It takes for them about 5-6 business days to respond to my request. It seems they are looking in the once a week to the High priority tickets. It also seems that they are permanently closed ! No matter when I

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

I am noticing that calls that come from our IAX pstn gateway provider and terminate to our Asterisk IVR do not receive ringing when an extension is dialed. For example: 1. An inbound PSTN caller calls our number 2. Asterisk answers and provides greeting 3. PSTN user dials extension of internal SIP phone 4. No ringback is heard from PSTN callers perspective 5. SIP user picks up or the

I noticed while reading some posts that people were looking for a complete NPA-NXX list for all area codes and prefixes. We happen to have the entire database. So I am making it available to the public. Help is available at: http://download.sixtel.net/npa/help.txt (Caution, 20meg files) Mysql Insert for this is available at: http://download.sixtel.net/npa/npainsert.txt CSV data for

How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm

I want to setup a phone system that can allow the caller to type in a faxback number, and automatically be sent a fax file. Is this possible with Asterisk? Or using Asterisk & a Linux fax program? I've seen stuff about incoming faxes, but nothing about outgoing. This is over a Zap channel, not IP, btw. Thanks.

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ? What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

I am wondering if anyone has sample XML config for the Sipura 2100 ATA. We have been autoprovisioning our 2002s with success and the 2100's take the same XML that we have come up with, but I am not sure of the syntax for specific things that I need these boxes to do, such as turning T.38 on. If anyone is willing to share their xml for autoprovisioning Sipura 2100's, it would be much

I have an analog phone connected to a Sipura 2100 which in turn connecteds to * over a 100mbps LAN. When I do ChanSpy on a bridged call, it causes massive jitter. When I attempt ChanSpy with a Grandstream GXP-2000 the monitored call is clear. Has anyone had this happen? Any suggestions? ScriptHead

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com