similar to: Debugging not going to log file

We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors.

Hello everyone, Our company has an Asterisk server where one of the T1s is connected to an IVR. Asterisk is configured for FXO Loopstart, and the IVR is configured FXS. We can call into it, but our software doesn't expect DNIS digits on FXS, and the digits are being interpreted as user input. We can certainly change our software accordingly, but our test bed also includes other

Off topic but an interesting article http://blogs.zdnet.com/BTL/index.php?p=1306 Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050603/6e1373e7/attachment.htm

Hi All, We have built an asterisk server (asterisk - 1.4.26.2) where there would be around 322 concurrent calls going on, but I can see that full log grows rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug and its tedious even by using any commands to get the required call from the log if there is any problem. Is there any way of splitting the full log into parts

Hello What is the difference between these 2 version of Asterisk in terms of functionality. For a small office am I going to run into problems if I use the easy version... Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/311c56ec/attachment.htm

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

Hi guys, a little OT but I figured this is the place that would know. Is there a free or paid webapp where I can get inbound sms messages? I only need to receive a few inbound sms messages a month but it cant be my current cell number :-( Any thoughts? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hi all, I've compiled samba 4 from branches and it worked like a charm. I'm trying to use regshell to read (remotely) the following Windows registry key: HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Uninstall I can connect to the remote machine with no problems: ./regshell --remote=192.168.50.142 --user=Administrador%xxx HKEY_CLASSES_ROOT> HKEY_CLASSES_ROOT>

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Anybody here having trouble receiving email from the list on Gmail? I havn't received anything since Friday July 29.

Hello - I am looking at connecting 7 - 10 locations together using Asterisk and possibly some VoIP gateway appliances. I need to insure best voice quality as these trunks will be used primarily for customer calls. I am considering implementing a full T1 frame relay circuit to each location which can be done for a reasonable cost. DSL and Cable are currently at each location and setup for

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I'm trying to setup asterisk to accept call from Teliax, request the 10 digit number from user, then dial it thru the VoIPJet. If I'm not wrong I will be charged by both providers because both connection is active during conversation. So my question is can I set the things so that I pay only to VoIPJet? Specific configuration snippets will be greatly appeciated. Thank you.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Okay, so I've just put the sernet repo file in my yum.repos.d directory and a yum update will elevate my samba server to the latest version. Is there any pitfall that is out there that I can avoid before yum updating. Centos 5.3 samba3-3.0.34-37 & related packages openldap-2.3.43-3.el5 & related packages I still have my samba3-3.0.34 packages squirreled away so I can force downgrade

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

I recently setup a new server with CentOS 5.2. Everything works great on the server except when I try to join the domain it simply tells me: "The following error occurred attempting to join the domain "Domain": The user name could not be found. I have researched this error and attempted a number of fixes but have been unable to resolve the issue. I am able to connect to