similar to: Is iaxtel.com down for 700 #'s?

Hello, I just started using asterisk, and have a question. I have setup two asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1 FSX modules) and is connected to the PSTN. B has same, but is NOT connected to PSTN. I want to configure B to call A via iaxtel, and connect to the PSTN using A's line. How can I configure iaxtel dial plan for B in extensions.conf? I want to be

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

Hello I am still having problems with IAXTELL and FWD configuration. I get the following when I dial 17009965342 which is set as an example to dial to FWD people. 1+700+99+ 5 digit number. I have placed XXXXX where my passwords are. CLI> Executing Dial("Zap/14-1", "IAX/abatista:xxxxxx@iaxtel.com/917009965342@iaxtel") in new stack -- Calling using options

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

I'm having trouble configuring these services the way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will use it. If I don't allow GSM Voicepulse will use ILBC. Does anyone know how to

Hi there, do I have a local problem, or is registration at IAXTEL impossible at the moment? "iax2 show registry" permanently shows a TIMEOUT for 69.73.19.178. Philipp

In taking a cursory browse at Vonage's site today, I noticed they are now offering a soft phone. Has anyone had any experience using this? And does this possibly open new opportunities for using Vonage with Asterisk? Just thinking outloud on the list, soliciting thoughts and experiences from others. AJ

Hi all I've been trying to make * work with IAXtel to no avail, all seems ok in the config but am not getting anywhere This is what I'm getting from console (user/pass/dest # changed for obvious reasons): DEBUG[1133735216]: File chan_sip.c, Line 3841 (check_user): Setting NAT on RTP to 0 DEBUG[1133735216]: File chan_sip.c, Line 4891 (handle_request): Check for res for phone1

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

G'day list, I am getting a lot[1] of traffic on my Internet link, ICMP messages from 69.73.19.178 telling me UDP port 5036 is unreachable (this IP address belongs to iaxtel.org). I see from the wiki that IAXtel supports only IAX2 from December 2003. Fine, however it looks like my * still wants to try and register using IAX1, and I can't find how to turn this off. This situation is

When I try to call iaxtel it goes to codec ADPCM even though I have define in iax.conf gsm Call accepted by 69.73.19.178 (format ADPCM) -- Format for call is ADPCM My settings: [general] port=4569 register => xxxx:xxxx@iaxtel.com bandwidth=high jitterbuffer=no tos=lowdelay [voipjet] type=peer host= xxx.xxx.xxx.xx secret= xxx auth=md5 notransfer=yes context=incoming disallow=all ;

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

Hi, I've been setting up * recently and slowly getting to grips with it, however I'm getting rather confused with all the different configs for IAX calls, FWD calls iaxtel etc etc. What I think I need it a basic understanding or even a comparison of these different voip systems (if thats what they are?) I'd like to be able to make calls to other voip users, both in the UK and abroad

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

Hi all, I spent most of the last weekend testing and trying to diagnose some mostly incoming call issues. I'll start with the easy one in the hopes it might have a positive impact on the others. First- I have an account with IAXTel. I can place calls to other IAXTel subscribers and also through IAXTel to landline toll free numbers and all works great. iax2 show registry shows I am

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't