similar to: PRI Crapping Out Regularly

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 --

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

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]

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

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

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

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:

"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

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

Any opinions on the best ATA? For example, if someone was having a problem and I wanted to rule out any ATA glitches or firmware issues, what device could I give them that I could count on to always be a trouble free top performer that just plain works?

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

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

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.

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

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

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

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