similar to: T.38 Passthrough testing -- IAX problem

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

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 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 --

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 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

"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

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]

<!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

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:

Having spent the better part of an hour searching the archives and voip-info I hesitantly ask what appears to be an obvious question but one I cannot find an answer for. Using Grandstream phones it seems that the only way to support Call Parking is to enable # transfers (i.e. use T in the dial command) since pressing the TRANSFER button on the BT phone is blind and one does not hear the call

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.

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

I'm looking at switching VOIP providers, but want to ensure I move to a company with sufficient capacity. Can any Canadian VOIP users post/email me with feedback on their providers? I'll post the results for all to read...... -------------- next part -------------- An HTML attachment was scrubbed... URL:

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within