similar to: Maximum retries for seqno 102 when re-inviting.

Below the trojaned and clean md5s are given. ---------- Forwarded message ---------- Date: Thu, 1 Aug 2002 13:39:22 +0200 From: Magnus Bodin <magnus at bodin.org> To: Wojtek Pilorz <wpilorz at bdk.pl> Cc: openssh-unix-dev at mindrot.org Subject: Re: openssh-3.4p1.tar.gz on ftp.openbsd.org changing rather than frozen On Thu, Aug 01, 2002 at 09:20:29AM +0200, Wojtek Pilorz wrote:

Greetings, Somewhere on my asterisk system, a calls come in in a certain context, for example, from-sip or from-pstn. Then the calls gets routed through the dialplan, and a macro gets called, and another one and then the call needs to be redirected to another number in the same initial context. And you can use Dial(Local/number/initialcontext) for that. Oops, this initial context is lost

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78

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Last month I saw something funny which I can't reproduce anymore: A 0500 number in .au is a service phone number and are forwarded on exchange level to a real phonenumber. So if A calls B it gets forwarded to C. Very simple. Now the funny thing, on the phone of C, I saw both A and B as the "caller id". I've been asking around and trying to get it again with a private 0500

Hi, I'm looking for somebody who has managed to get their IPmux8 or IPmux11 talking to an Asterisk machine. I have it setup properly I think (the two IPmux's are talking to each other, and the zttool says that the PRI is acting okay, but I'm flooded with HDLC aborts and FCS problems. Edwin -- Edwin Groothuis | Personal website: http://www.mavetju.org

Hi, Yesterday I updated asterisk to the latest zaptel driver and today my congestion problems are gone... (see http://bugs.digium.com/view.php?id=6509), only to be replaced by: Feb 17 10:02:37 DEBUG[19225] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 26 Feb 17 10:03:08 DEBUG[19274] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 216 Feb

Greetings, Our telco called last week, saying that a lot of channels on our PRIs are blocked. And with blocked they have the following description in the Siemens exchanges: BBAC BLOCKED BACKWARD This status is set when the partner exchange has a blocking set and the signaling of the trunk (non-CCS7) is able to report this blocking in the backward direction. This status can

The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One

People who use "asterisk -r" already know that if you don't try it as root (for example webservers), you don't get access. Of course you can chmod it aech time you start asterisk, but that gets annoying after two times. I have made a patch (against HEAD and 1.0.5) which makes it possible for you (=asterisk admin) to configure the ownership and permissions of

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

Last tuesday I moved the asterisk server from 1.0.7 to 1.0.9, while leaving the zaptel drivers at 1.0.7 because it was a "lunchtime" update. This is a box with two TE405Ps in it, and all eight ports in use. Today I unloaded the 1.0.7 drivers and replaced them with 1.0.9 and oh boy... two of the 8 PRIs didn't want to come back, I got a million of FCS errors over the console and I got

We have a Vegastream 400 connected to a digium Quad PRI card in an asterisk server, for the T.38 faxing here. Problem is that there are too many clockslips on it (and they get logged by asterisk as HDLC aborts). I've double checked the configuration on both sides, replaced the cable, tried different ports etc. It all lead to no resolution for it. Is there somebody on the list who has a

On Tue, Jan 02, 2007 at 03:17:35PM -0700, asterisk-users-request@lists.digium.com wrote: > Has anyone made this combination work together? I've tried everything > and can't seem to get it work right. It all compiles fine, but when > rxfax is called, I get an unknown symbol error. From my reading, > everything points to me having multiple copies of spandsp and it's

Hi guys, Just wanted to give you a heads up, so you don't end up chasing strange issues... Since early this morning, our Tipping Point IPS is blocking the Asterisk generated SIP Quality messages (the ones which tell you how good or badly reachably a remote SIP server is) Rule 5051: SIP: PROTOS Test Suite INVITE Test Case This filter detects a test case from the PROTOS SIP testing

I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->answer(); my $i; $i = $AGI->channel_status(); $AGI->say_digits($i); $i =

I've seen other request info on this config before. I've got the system up and running, Zap/Modem has been working for several days, no errors/warnings. Simply added info for broadvoice and I can dial into my system via the remote, but dial out from my box renders the following: I get one ring on the remote line, then on the console chan_sip.c:675 retans_pkt: Maximum retries exceeded

Hello All, Got one customer and he is getting disconnection within 15 seconds when he tries to make outbound calls. Initially, it was working fine without any glitches... Other customers on the same system are working fine, its just with this customer only. This is the error message thrown by Asterisk on the CLI: - Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum retries

Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!