similar to: Vegastream clockslip problems

Displaying 20 results from an estimated 3000 matches similar to: "Vegastream clockslip problems"

2006 Feb 12
2
dual TE410, both span 3 is broken
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
2005 Aug 20
0
Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
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
2005 Aug 20
0
1.0.9 - can't get link up, 1.0.7 works fine.
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
2006 Apr 18
5
Remember the incoming context?
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
2005 Feb 28
3
Digium E1/T1 card with mgetty+sendfax
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
2005 Jul 24
1
Caller ID, Called ID and Forwarded ID
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
2006 Feb 16
1
Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it
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
2006 Apr 03
2
Blocked channels, according to our telco... leading to CONGESTION status
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
2005 Aug 01
2
TDM400P REV I issues - ProSLIC vs TDM400P
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
2005 Feb 11
0
/var/run/asterisk.ctl configuration
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
2007 Jan 03
0
Re: asterisk-users Digest, Vol 30, Issue 4
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
2007 Feb 20
0
Tipping Point IPS blocking Asterisk SIP quaility messages
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
2002 Aug 01
0
openssh-3.4p1.tar.gz on ftp.openbsd.org changing rather than frozen (fwd)
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:
2007 Apr 20
0
RAD IPmux8
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
2008 Feb 04
8
AGI: Not getting answers from get_data in a call-file call
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 =
2007 Aug 14
0
Maximum retries for seqno 102 when re-inviting.
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
2005 May 11
0
Vegastream assistance?
I wonder if anyone can help me? Am trying to terminate to H323 Vegastream. I'm using OH323 with little success. I can dial out and answer but voip end just keepings ringing and ringing. Thanks for any help. Neil Config file: [general] listenAddress=ALL listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no jitterMax=100
2006 Jun 09
3
FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According
2005 Aug 10
0
RE: Info / recommendation on either Audiocodes or Vegastream gateways
> > I am looking for "how to" information / references on use of either Audiocodes MP104 or 108, or Vega 50 Gateways for interconecting Asterisk to the PSTN via FX0 interfaces. > > Any info of references / personnal experiences would be appreciated > > Stratus? THE WORLD'S MOST RELIABLE SERVERS * > Richard C. Sparacino > Telecom Technology Manager
2006 Feb 20
6
Incoming Calls Getting Crossed - Weird
Hey, I got a weird one for you guys, I am running vanilla 1.2.4 and have all incoming calls come in as SIP from teliax. Twice over the past week 2 callers who have called in around the same time end up talking to each other instead of going through the ivr or at some point during the IVR. One said, yeah i was talking to another patient and we had a convo. I have double checked the dialplan and