similar to: IAX registration keep alives

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have successfully configured an IAX trunk between 2 asterisks, calls can go through both ways without any problems, NAT in the middle of course (iptables) Now , leave them for a while , and make a call from the external server , it doesn't go through, Dial from the internal one, everything works fine again.. Now , it is clearly a problem in the NAT engine,

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

Hi, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4 Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I've tried every combination and I cant get it to work. I'm assuming it must work otherwise it wouldn't have been listed but for 60 seconds more work it would be a bigger benefit to the asterisk

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Hi all I been looking into the whole code strugture of chan_iax and i see there is a option to specify the refresh rate of registrations: But there is no code to actually load this from the config file thus i changed the setting in chan_so.h, and recompiled. But still my refresh rate is 60 sec. I need to get this down to 15 sec (nat /pat firewall issue) any ideas? thanks Liaan

Hi all Do any of you know i can force asterisk to lookup ip addresses for peers and trunks everytime it tries to make a call. One of the peers has a dynamic ip and is using DynDNS to register host. Now i need to reload asterisk everytime i want to call it thanks liaan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL:

I know this has come up before but... Has anyone done anything to get an IAX client built on Windows? I thought someone had started one, but I haven't heard anything about it since - and that was months ago? Anyone have any idea what the status is? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming,

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

On Mon, 14 Feb 2005, Reid Spencer wrote: > isn't necessarily tied to LLVM. Anyway, lets cross that bridge when we > get there. Sounds good. -Chris -- http://nondot.org/sabre/ http://llvm.cs.uiuc.edu/

Tanu Sharma wrote: > Hello, > > In an attempt to randomise the basic blocks in a function, is it > possible that I can randomise the entry block as well? And maybe insert > some instructions in the pass to call entry block while running the > program ? > > Is it feasible? > > What does entry block consist of ? The entry block, by definition, is the first basic

I need to setup quota on //server/home for user a to 1GB and on //server/client (for all users on this share) to 10GB May you please let me know the exact command to be used for this? Also, do I need to add any other parameter in smb.conf in order to make quotas work? I have compiled samba with quotas option. Thank you. --------------------------------- Do you Yahoo!? Yahoo! Search presents

Hello, In an attempt to randomise the basic blocks in a function, is it possible that I can randomise the entry block as well? And maybe insert some instructions in the pass to call entry block while running the program ? Is it feasible? What does entry block consist of ? Thanks Tanu --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second

Hello, Is it possible to display the data structure graphs created by the Data Structure Analysis using a visualization tool? Thanks --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup a SAMBA + OPENLDAP server following Samba-3 By Example, but I have run into several problems. All the tests described in the Chapter 6 such as pdbedit -Lv, slapcat, and ldapsearch -x -b give the desired results. But, getent can't read passwd or group information from ldap backend. I have done my best to solve the problem, but it just doesn't work. I have compiled the nss_ldap

Dear All, . After installing, testing and like many other I found that Asterisk is reliable and a great Open source telephony solutions for a professional use. I would like to create a business with *, offering it as a IP PBX solution to customer, as a server, whatever with some Digium HW and SIP phone, .. etc. Asking for something that might have been asked. What are the implications for