similar to: Telco MWI Detection on TDM400 Interface?

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

I'm trying to get the Telco MWI recognition working in asterisk-1.6-beta7. I'm told that it's supposed to work provided my telco support FSK MWI signalling. I have Verzon FIOS. I believe I have FSK MWI signaling as I can hear the standard stutter tone when I pick up a live handset in front of my asterisk connection. I have the Verizon line attached to a TDM400 card, port 4. I

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

Folks, I've tried to find a reference, but I've had no luck, and would appreciate your thoughts: I'd like to be able to monitor a telco line for Message Waiting Notification, however I cannot figure out if this capability is available. Detecting either FSK or Stuttered Dial tone would serve, but I can't find anything in the list archives or Wiki with clues as to how this might

I use a TDM400 card to interface with my telco. I used asterisk voice mail. However, if I'm on the telco line while another call comes in, obviously it cannot go to asterisk voice mail but instead bounces to the Telco voice mail. Is there any means by which I can get asterisk detect the Message Waiting Indicator (MWI) from the telco? Is there some application or variable which can be

Hello, I'm new to asterisk and trying to set up a PBX that's connected to ISDN on the telecom operator side (Swisscom, Switzerland) and has analog line on the local side. I use Digium B410PF and AEX2460EF cards. Globally, everything is working well except that I can't get the CID sent to the analog phone. I tried with all combinations of options I could think of and several asterisk

I''m having troubles with my outbound VOIP connection. I''m convinced that I don''t have QOS/traffic shaping configured properly in my shorewall linux firewall, which serves as my Asterisk VOIP server and Internet router/gateway. I don''t have a separate router box. I''ve been using VOIP for about a year now, but just recently realized that I need to

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

Many times the "speed" of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A "speed" change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

> he graphics card is emulated in Qemu, the graphics operations are > converted to VNC commands there, and you see on the VNC display what > would appear on the Windows VGA display. OK, Thanks. 2 other questions: 1. how do I connect tzo the VNC server for my domU OS? 2. Where is the QEmu code? In xend or in kernel? -jec Jean-Eric Cuendet Senior developer / Technical support