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Displaying 20 results from an estimated 4000 matches similar to: "No subject"

2007 Dec 31
1
PRI Crapping Out Regularly
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"
2008 Feb 18
0
Vancouver - Asterisk Event Feb 18 (Monday)
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
2008 Jan 29
1
PRI Alarms, Comes Back, But Asterisk Won't Touch It!
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]
2007 May 16
5
Microsoft's Move Into IP PBX Market
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 --
2007 Jun 08
0
Replacing SX-2000 Centigram Voicemail with Asterisk?
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
2005 May 09
2
AGI - How to Make Calls and Bridge to Original Incoming
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
2006 Mar 17
1
One-Way SIP Audio with SVN Codebase (CANCEL)
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
2004 Dec 06
0
CVS HEAD h323 no longer builds?
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
2005 Aug 24
0
Distorted Sound from E1
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
2006 Jan 19
0
AudioCodes Unreliable DTMF Detection
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
2006 Mar 15
0
T.38 Passthrough testing -- IAX problem
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
2006 Mar 17
0
One-Way SIP Audio with SVN Codebase
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
2006 Mar 18
0
T38 Passthrough testing -- unknown media type error
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
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G
2007 Feb 08
2
dCAP
Hello. To someone who have done the dCAP exam. I would like to know about it: test and practises questions examples, difficulty level,... I'll be very grateful if somebody sends me an exam model. Thanks in advance
2006 Oct 16
7
tdm2400p question
Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Nov 06
2
Asterisk & OpenVZ
Hi All, I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? Also, I'm unsure how the zaptel modules come into play, could use some guidance there as well. Thanks. JR -- JR Richardson Engineering for the
2008 Feb 15
2
HPEC
Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080214/3f0580f1/attachment.htm
2008 Jan 09
2
Intercom & Paging with Polycoms
I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but
2009 Feb 05
2
hardware that can accomondate 2 TDM24
Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St.