similar to: IAXTEL testing

Hey all, I was playing around with IAXTEL last nite and have outgoing calls working a treat. I'm sure I woke a few people up in the US with my annoying test calls. :) Anywayz, incoming calls are a different matter. I have a NAT firewall my * box is sitting behind and the server 'appears' to have registered correctly with IAXTEL. Thing is, when I try and call my 1700 number

incominglimit is already implemented for SIP. Just specify under the endpoint how many incoming connections are allowed. For example, [cisco] type=friend username=cisco secret=blah nat=yes ; This phone may be natted host=dynamic canreinvite=no ; Cisco poops on reinvite sometimes qualify=200 ; Qualify peer is no more than 200ms away

Hey all, Does anyone know if there is a way to browse the CVS tree online, via a webpage? This sounds like a stupid request, but I don't have access to a shell at the moment and wanted to go over a few things. Regards, Jamie Carl Jazz Inc. Email: me@jazz-inc.net Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: jazz@netmindz.net

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, Just noticed something that might be an issue. I have just made asterisk crash consistently by doing the following. I have a D-Link DG1102s running MGCP into asterisk and an extension *9 setup which dumps me into my inbound context to simulate calls coming in from my X100P. This usually works with no hassles

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table.

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

I have been playing around with asterisk for a week or so now and haven't had too much trouble getting things to work but one thing seems to puzzle me. I have been patient hoping that there was a configuration error on the server or that the toll-free gateway was down but nothing has changed. I have the following configuration context for IAXTEL: [iaxtel] exten =>

Does anyone know if this unit works with Asterisk? Thx. B.

After battling for days trying to figure out what was wrong with my iax.conf it was determined that I do not have any inkeys set on the digium server. Now whether that is something new or just in a few cases I am not sure. Messing around and reading on IRC and the mailing list I could get certain things to work and then break other things. Now I can dial a IAXTEL number, 800 number and FWD

<BLOCKQUOTE style="PADDING-LEFT: 8px; MARGIN-LEFT: 8px; BORDER-LEFT: blue 2px solid"> <DIV>Hello,</DIV> <DIV>&nbsp;can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf</DIV> <DIV>&nbsp;</DIV><FONT size=2> <DIV>[iaxtel-trunks]</DIV>

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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

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", "")

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

Hello all, Can someone send me a current working example of what the entries need to look like for iaxtel and the iax.conf file inclusive of the register line? It's been quite awhile since I have done anything with iaxtel and I need to do a little bit of testing. Currently it appears that I can make calls out to iaxtel fine however inbound seems to be another question. I see 2 iaxtel

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})

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