similar to: 1.0.9 - can't get link up, 1.0.7 works fine.

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

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

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

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

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

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

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

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:

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 =

We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of "sip debug" is funny from line

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, I am about to pull my hair out after trying to get our PRI up and working. We are switching from a Cisco gateway to an Asterisk box which provides the 23 phone lines for our office. So, because the Cisco gateway is working I can assume I have all the settings right (b8zs, esf, dms100, etc) and the PRI is live (because we are switching over). When dialing from PSTN, I get busy signal. When

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

I'm trying to install a TE410P this is what happens with compiled zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/ this is my zaptel.conf (checked with the provider the values): span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=it defaultzone=it then I modprobe wct4xxp debug=1 t1e1override=15 and the kernel says : Sep 30 16:12:40 localhost kernel: Zapata

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]