similar to: Fwd: Message from iptel.org SIP admin (more register= bugs)

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

It's been down the last 5 hours at least. Anyone know what the problem is, or when it will be back up?

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

Hi all, I'm trying to connect my Asterisk@Home to iptel.org, but the only I get is Allison telling me "circuit busy now, please call again later" or some thing similar. I'm trying make it by AMP and editing sip.conf and extension.conf, and I read all about it in voip-info.org. I will appreciate your help, Thanks in advance, Sebastian e-mail:smilioto@GMAIL.com IM:

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

FYI: Asterisk puts URIs in messages which violates the SIP spec and can't be accepted by URI parsers: username includes a whitespace. See for example the From header field. Attached is example of an incorrect message and related parts of RFC3261 specification. (Who doesn't want to dig into parser details may want to realize that whitespaces are used as uri delimitors in first request

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

https://bugs.freedesktop.org/show_bug.cgi?id=58378 --- Comment #24 from Andreas Loew <awl1 at gmx.net> --- Created attachment 90764 --> https://bugs.freedesktop.org/attachment.cgi?id=90764&action=edit dmesg output on 3.13-rc3 while the issue was seen -- You are receiving this mail because: You are the assignee for the bug. -------------- next part -------------- An HTML

[sorry for cross-post here from -biz, but there are significant non- cross-subscribed audiences] I'm looking for some case studies for people who are implementing Open Source Asterisk in Federal, State, and Local governments in the United States. Please reply PRIVATELY to jtodd at digium.com with contact data, and I'll follow up. I have some requests from various directions for

[moving to asterisk-users by request] On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote: > > On Jan 26, 2009, at 7:38 PM, James Lamanna wrote: > >>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote: >>> >>>> Hi, >>>> Is it just me, or does DTMF queuing not work properly? >>>> I'm consistently faced with

So, after playing with 6.0 on the Cisco 7960 and 7940 platforms, I have the following gripes, which I've sent to a very clueful Cisco person already. Mind you, I love the Cisco 79xx series phones, and currently they are what I recommend to anyone who wants a 'real' IP phone. I just cringe - Speed dials. It's nice to now have speed dials in the line appearances that

AGI's are resulting in unusual behaviors. Can someone please tell me if this is my inappropriate use of AGI's, inappropriate use of Time::HiRes, or a bug with *: I call this script twice: #!/usr/bin/perl use Time::HiRes qw( gettimeofday ); ($seconds, $microseconds) = gettimeofday; $hirestime = sprintf("%s","$seconds$microseconds"); print "SET VARIABLE

The freenum.org project wants to use your trunks! The freenum.org project is an ENUM parallel tree, which has as an eventual goal the distribution of ENUM numbering in nations or areas which due to political or other issues are not able to get secure, inexpensive, or functional ENUM capabilities. As a preliminary round, we're putting toll-free gateways into the system for various nations.

If the 1/2 ATA186 device supported IAX I think I'd have found nirvana :-)(especially at $49.95 ;-) Lenny -----Original Message----- From: John Todd [mailto:jtodd@loligo.com] Sent: Monday, April 07, 2003 7:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] IP hard phone devices (grandstream) The device at VON would have various parts of it's LCD screen go into

I have some strange behavior happening with call flow when analog line errors are encountered. This may be due to the way that the X100P detects "busy" signals, or it may be something in the software. Could someone with more in-depth knowledge make a comment on the items below? My dialing logic says "dial local area code numbers out of the analog line, and if the analog line

So I have many Cisco 7960's that are running the latest 5.1 Cisco SIP code and I cannot get the phones to talk/RTP to each other. jtodd has had this problem in the past with the 186's. Just wondering if anyone has a reason why "Cisco sometimes poop on reinvite" is the Cisco code broke? if so we can push on Cisco to fix it. the U is a MAJOR Cisco shop so we have some puhs

ok all. I have sent the PHP web gui code to Mark at Digium but have not heard back yet. I dont know that status if he wants to CVS it or not. maybe if he does not answer in a this next week ill just upload it to a website for all to DL. we could add to this as a base and get it to work better for all again the current URL for it is http://rads.netcom.utah.edu/openconf/openconf.php Dave P