similar to: Dropped Calls between Sip and Zaptel

Hey, I've done some searching for this and never really found a concrete answer. Is there a specific reason or solution why just in the middle of a call Asterisk will drop it and I'll get dial tone again? Anyways, this is the output from /var/log/asterisk/full at the time of disconnection: May 13 08:37:13 DEBUG[5379]: Stopping retransmission on

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk started. Just before things go down, the log shows the following error: ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500 at which point a "show pri spans"

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of

From c|net News: "On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 --

Greetings list, Wondering if anyone's come across this before. I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server: -- Called

We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca

We're working on an application in which it appears it would be far more efficient to share data between Asterisk and external applications by simultaneously accessing and updating astdb. While the current asterisk/db.c code uses ast_mutex_lock and unlock pairs to protect the integrity of astdb from multiple Asterisk threads, this of course does nothing to protect astdb from external (i.e.

Attempts to perform a "make all" in /usr/src/asterisk/channels/h323 fails with countless errors of the form: /usr/src/pwlib/include/ptlib/ptime.h:152: macro or `#include' recursion too deep In file included A "make all" using the stable branch builds with the same pwlib code but of course the h323 code in the stable branch doesn't work. So it seems those of us who

>Date: Mon, 07 Feb 2005 02:22:07 -0800 >From: George Pajari <George.Pajari@netVOICE.ca> >Subject: [Asterisk-Users] Remote MWI via IAX? >To: Asterisk Users Mailing List - Non-Commercial Discussion. > <asterisk-users@lists.digium.com> >Message-ID: <4207414F.10807@netVOICE.ca> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed> > >We have a

We're having a problem with an E1 trunk in Mexico into an IVR server and would appreciate any suggestions. Hardware: Digium TE110P jumpered for E1 zaptel.conf: span=1,1,0,ccs,hdb3 # clear=1-30 bchan=1-15 bchan=17-31 dchan=16 loadzone = us defaultzone=us Circuit status is fine: Status: Provisioned, Up, Active Calls are accepted by Asterisk without any

We're trying to use some AudioCodes MP104 FXO units as gateways to Asterisk but cannot get them to reliably detect DTMF. Some landline calls get most digits but some are duplicated. Some cell phone calls get 0% DTMF recognition. Anyone with experience with these units have any suggestions? ABP Technical Support has been unable to diagnose the problem and is now sending random guesses and

Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to pass some calls to another using IAX and attempts to use the Dial command results in multiple messages "Out of idle IAX2 threads for I/O, pausing!". Since this server needs to support IAX I'll have to back out this version and find another idle server to use to play with the T.38 code. g. -- George

Please tell me the obvious mistake I'm making here. (And yes, I well know about NAT and one-way audio problems in general.) I want to try the new T.38 passthrough stuff, downloaded it, built it, tested it with an SPA-2100 and can hear announcements fine but echo test shows no audio outbound (i.e. SPA to Asterisk). Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in

I wrote earlier: > Please tell me the obvious mistake I'm making here.... The problem was a lack of sleep. Sorry to have troubled the list. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk

"Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith." - George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102

The Vancouver Linux User Group is holding a "Virtualization Round Table" Monday (Feb 18) evening at the BC Institute of Technology discussing some of the different approaches to server virtualization. I'll be speaking about using OpenVZ to provide virtual servers used to host multiple instances of Asterisk (the technology behind our Virtual Private Asterisk Server or VPAS

Hi, I'm trying to autotune a good sequence of IR optimization passes and I seem to run into segfaults in opt (in LLVM5) with certain pass orderings. Is this expected behavior? If so, what would be the recommended way of determining pass dependencies so that I can encode them into the tuner? The test program can be found here: https://gist.github.com/kavon/92d153cdd54ce9b77162af3af47d4c95

Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P (same problem with various previous versions; same problem with different TE120P cards). The customer has a partial (10 B-Channel) PRI that when it is busy (eight or more B channels in use), tends to fail as shown below... [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown error 500 [Jan 26 23:00:31]