similar to: How to "busy out" zap channels

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

Well I'm stunned no one has suggested a webcast option. I mean we aren't talking a bunch of people unable to grasp the concepts of chat/voice/vision sessions with a log in/remote display capability. If you think this is an option let me know I have someone who has some software they wouldn't mind stress testing as a trial. Cheers, Dean > -----Original Message----- > From:

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

Hey, I've come across two interesting problems today. First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that

Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordings for no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2

I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the "start, answer, end" information in it. Below is the insert statement that MS Profiler shows being sent. As you can see those fields are missing. INSERT INTO

Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -------------- next part -------------- An HTML attachment

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of

Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug

Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried

At the risk of sounding like an newbie, is is possible to build the RPMS for all architectures on the same box at the same time? I would really like to automate this, so that I can keep track of the RPMS's and build them into my own future repo. (That's another project, I'm sure I'll come to the list for that one!) -- The Solo System Admin - Follow me - I follow you

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

Asterisk Users, I am running Asterisk-1.4.11 on a Debian "Etch" system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions,

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

I''ve this small test app where I''m saving a user profile picture in a subdirectory under "public" folder. Now when a user uploads a picture - a unique directory is created under his id and then his/her image is stored there. Now while retrieving the image I only get the path name and nothing else. How can I get the image shown in the browser...HELP PLEASE!!! --

On 30.01.2016 16:59, poma wrote: > ... > > https://sourceware.org/bugzilla/show_bug.cgi?id=19538 > > Mister Anvin, care to share what's the status of the "ld?" problemo, is anyone working on it? It would be maravilloso if syslinux can continue to build and test with latest and greatest.

Good Evening I just wanted to ask a question here, is it possible to build a static binary that I could sep to a client machine, and have it do it''s first run to connect to a puppet master? My plan is to use a static binary as a sort of installer for particular systems, as I don''t really want to have to install puppet client on 70 systems by hand! These are all production

Let say I have a server located in Europe and one in North America. The 2 servers are connected together with iax2. Both server are connected to phone lines in there own country. If I want that when a user call a north american phone number from the server in Europe it use a zap channel on the server located in North America and also if someone in North America dial an European phone