similar to: Cisco 7960 BLA

I am having some really strange problems calling from 2 asterisk boxes of mine. One is version 1.2.22 the other 1.2.18. The problem is identical on both boxes. When I try to call certain numbers (8006375410, for instance) the call rings and rings and rings. Eventually the receiving end will pick up in the middle of an IVR as if I had been connected for some time already. When I call this number

After the upgrade, I was encountering a "network unreachable" error. This happens on kernel versions -> 2.6.32-573.1 .1-el6.x86_64 and 2.6.32-573.3.1-el6.x86_64. If I revert back to 2.6.32.504.30.3.el6.x86_64, the problem goes away. I've been searching frantically without much luck. I have found references to the same error on bugs.centos.org but no resolution as of yet. Has

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

Hi, I'm sure you have all seen this exception dozens of times, but I cannot figure out what it means and what I am doing wrong. As per suggestion of the list, I have put up a Mingw64 build slave and am frantically trying to get it to work. But the Waterfall display kindly informs me that "exception get_builddir" is the result of the attempt of building. I know I am doing

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

Hi all, I am trying to compile a fortran routine originally written for R from within matlab. It all worked fine, but the newest version of the Fortran code has a call to rchkusr() and I now get a message that this is an undefined symbol. I am under the impression that rchkusr() just serves to let the processor take care of interruption messages (typically, the user frantically

Xen Community, I have a team of people frantically creating manpages for the xe command and all of it''s 361 subcommands. The initial source documentation is in asciidoc but also saved as Docbook. From docbook we transform to epub, html, manpage and pdf. This is currently being hosted in a subversion server at a Seattle area college and we''d like to move it to the xen-org

I have an Asterisk server with 3 TDM400P cards. 9 FXO and 3 FXS ports. It also has 2 Astribank-8 units connected. The customer is having calls dropped at random intervals but several times a day. Could this be an issue with Interrupts with the 3 cards? I am also having a problem sending and receiving faxes when they are either connected to the Astribank or to an FXS port on the TDM

Has anyone tried runing zfs on the Intel SS4200-E [1],[[2]? Doesn''t have a video port, but you could replace the IDE flash DOM with a pre-installed system. I''m interested in this as a four disk smallish (34x41x12) portable ZFS appliance. Seems that people have got it running with Linux/Openfiler [3], and the process a Celeron 420 is 64bit. [1]

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

--- Alex Young <alex-qV/boFbD8Meu8LGVeLuP/g@public.gmane.org> wrote: ><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><><> > Have you

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris

Dear nouveau developers and users! After many years of dealing with the blob in my kernel, I decided it was time to move to an open source solution. With one of my machines, equipped with 9500GT it went almost flawlessly. I had to patch kernel to get GLXVBlank option working (only because I used the latest and greatest 3.7-rc6), but that is all, and I'm very happy with the result. Nouveau on

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

My goal here is to share a directory that people ftp files to. [global] workgroup = TELEVOX_1 netbios name = samba server string = Debian Mass Storage Device security = domain realm = TELEVOX_1.LOCAL username map = /etc/samba/smbusers #password server = * encrypt passwords = yes interfaces = 192.168.169.31 localmaster = no winbind separator = + winbind cache time = 10 idmap uid = 10000-20000

The "telco guys" probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul