similar to: Strange Call problems on some numbers

While I'm frantically scouring this list, does anyone have any information about getting BLA (busy line appearance) working on Cisco 7960? The last I heard was that this was unsupported in Cisco's SIP firmware

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]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I've been using for a long time asterisk-perl-0.08 for prepaid card applications, and I've identified a problem with the last releases of asterisk-1.2, installed with Trixbox. The command get_variable() raises a signal SIGPIPE when it is called (whatever the variable to get). I made tests with Asterisk 1.2.20, 1.2.21 and 1.2.22, and I

Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller

Hello, I'm trying to put together a low cost - low powers PBX appliance for several customers. I have purchased a couple of the soekris net4801 boards and have asterisk up and running on them fine but they just don't quite cut it in the processing power department. I've been able to get about 10 simultaneous SIP calls with simple ulaw (no encoding decoding). While this might be OK for

Hi, I'm using an Asterisk 1.2.18 box with a remote Snom 360. My Snom always rings but sometimes (it happens randomly!) no voice is passing thru (2 ways). Asterisk CLI shows this warning: Jan 10 10:03:26 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown at xxx.xxx.xxx.xxx> I have already set localnet and

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

Hi, I have Asterisk-1.2.18 install with FreePBX & more than 75 extnsion, daily I come accross an issue & try resolving them its either user learning curve or my ignorance. But, I dont know what to say regarding this issue. I have my Dial Plan for internal users to have a 3 Digit Extensions. So instance my Ext is 239 & someone dials the main #, its gets the

Olly Betts <olly <at> survex.com> writes: > > On Thu, Jan 14, 2016 at 11:04:29AM +0100, Jean-Francois Dockes wrote: > > Olly Betts writes: > > > On Sun, Jan 10, 2016 at 02:53:14AM +0000, Bob Cargill wrote: > > > > I will look into the bug you listed to see if it might be related. If there > > > > is anything else that I can do, please

Hello, We're in the process of moving to a PRI circuit for our asterisk switch. Can anyone point me in the right direction as far as PRI Cards are concerned? Thanks!

Hello, Can anyone recommend the 'best' kernel and zaptel versions to use with asterisk? we're currently running trixbox and are having numerous call quality issues(disconnects, echo, garbled speech) and I'm considering wiping the asterisk box and installing a virgin copy of centos, compiling asterisk myself and installing freepbx on it's own.. Is there anyone who can

I discovered, while trying to set up Tcl bindings for Notmuch (https://notmuchmail.org/), which uses Xapian, that flintlock was not being locked (I had lost updates). I then found that opening a Xapian database for writing directly via the Xapian Tcl bindings also silently fails to lock flintlock. I have taken a copy of flint_lock.cc to play with, and I find that it locks the file when called

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

Got a wierd one that's reminding me of a problem mentioned in an earlier post but for the life of me, I can't find it. So... Inbound calls via a Voicetronix interface on my Asterisk box are being properly detected and routed to my dialplan as expected. It's a simple plan right now that rings a few internal Voicetronix and SIP stations. When the inbound line rings, it's ringing

Hello, We've released another update to our astGUIclient suite: 2.0.3 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk

Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is