similar to: EAGI delay

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT <*prosody rate*=\'.8\' >Hello World\, this is a test\,</*prosody* >|0|1 Would I use a scape character? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/7132be4c/attachment.htm

Hi! If the caller hungs up while an eagi script is running, I can?t regiter the cdr manually at the end of the script. I tryied to trap SIGHUP but it didn?t work. I want to register my own cdr into the script because I have a lot of data that I need to put in the cdr. The 'h' option or DeadAgi aren?t a solution for me. Thanks -------------- next part -------------- An HTML attachment was

Any C gurus out there that can tell me if this code compiled ok to be used in x86_64 (Pentium Dual Core). It's for the app_conference application. Im using Centos 4.3 x86_64 kernel: 2.6.9-34.ELsmp libgcc-3.4.5-2 gcc-3.4.5-2 after the compilation part is the makefile ************begin compilation******************* [root@centos app_conference]# make clean rm -f *.so *.o app_conference.o

Do anybody could compile app_conference on x86_64??? I tryied with two versions of app_conference and got the same problem on compiling: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared recompile with -fPIC app_conference.o: could not read symbols: Bad value" ENVIRONMENT:

Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info

Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. -------------- next part -------------- A non-text attachment was

I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

jonathan blais wrote: > I'm using Linphone. I tested with Asterisk and Speex only, I created a > channel with echo and it worked. It seems to have problem when using > app_conference. If you just use app_echo, then asterisk won't be trying to decode your frames; it will just be sending them back to you. Therefore, if your client is using an incompatible packing of the

Jean-Marc Valin wrote: >Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, >Linphone just sends raw packets, as specified in the RTP draft. > > Asterisk expects speex frames to have a terminator. The phone I was referring to was the X-Ten/X-Lite phones, which seemed to be adding something _before_ the speex data to indicate the length of the frames.

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?