similar to: RECAP: 3?

Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

Hello, I have a block of 25 DIDs and have 10 phones on the network. I want when a person tries to call out for * to pick a number for the CIDN and I want to make sure that the number isn't duplicated while it's in use. Joshua __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Voicepulse has ignored four emails over the course of two weeks. Anyone have any ideas of whats wrong? - Executing Dial("IAX2/voicepulse-in-01@66.234.228.170:4569/7", "IAX2/acctname:acctpass@gwiaxt01.voicepulse.com/14109649073") in new stack -- Called acctname:acctpass@gwiaxt01.voicepulse.com/14109649073 Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi guys, I'm using asterisk 1.2.1 since a week ago or so. today I found it crashed when making a call through teliax. This is how it looks: -- Called xxxxxxxxx@teliax/17075471770 -- Call accepted by 208.139.204.245 (format ulaw) -- Format for call is ulaw Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Folks, I've been running asterisk successfully using the extensions.conf and voicemail.conf. Now that I've got asterisk happily looking up MySQL tables for the VM configuration, I decided to try out the contributed script /usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl I edited the script so that its output goes to a separate extensions_from_mysql.conf file. The

I'm new to the Asterisk software but have successfully set it up to make and receive calls using FXO cards, voicemail transfer etc. I can successfully call the Digium test IAX using the examples provided. I have signed up for an IAX tel account and got a number. The extensions have been set up as per the examples from IAX tel. However when I try to place a call this is what I get: --

Hello, I updated CVS and nobody can call me any more with my IAX number 17007591228. I can only call other number but nobody can call me. This is what I get on debug when I call myself: -- Executing Dial("SIP/1011-7424", "IAX/bartosz:password@iaxtel.com/17007591228@iaxtel") in new stack -- Calling using options

I have two units at customer locations in the Caribbean registering to a server in the US. Both units are connected to the Cable TV company's internet feed. If I run mtr to the units I see clean internet and low latency, but when I watch the CLI, I see constant problems. The audio quality is terrible, but I can't see why. Sep 3 16:42:46 NOTICE[20423]: chan_iax2.c:7014 socket_read:

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

Thanks for everyone's input. I had been under the impression and was passing that impression on to my students. The take-away here seems to be that once you start mixing official and unofficial repos anything can happen. dennisk -- "Free as in Freedom" Free Software Foundation

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 With many thanks to Gedalya and Sven Hartge Situation: Dovecot installation with userbase in a MySQL table with the same structure described in Postfix.Admin installation; filesystem permission is flat with a single user (vmail) who owns maildir mailbox files and directories. Procedure to implement last-login plugin to update mailbox table. The

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------: