similar to: Meetme "talker optimization" always on even when no "o" option present.

Hi all, I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32 machine, and can't find any problems. So, I decided to upgrade my home pbx. All went well until I tried using my S101 to talk to the IVR. Some times, the first one or two digits get through, but eventually a digit will get stuck, playing continuously until the call is terminated. I have confirmed this

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling

puppet version 0.24.8 from debian lenny-backports My class works and resource created the authorized_keys file. But puppet detect as ensure absent and added again and again my authorized_keys got fews the same key lines. I added the target => ''/home/test/.ssh/authorized_keys'', again ensure is absent. I replace the ssh_authorized_key.rb from 0.25rc1 and again ensure is

As far back as I can remember in 1.4, the option of sending a VM from voicemailmain (3-5 or 3-5-1), depending if you could use the directory has been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I do seem to remember that in 1.2, it wasn't possible to send a message to ones-self. This bug fix apparently corrects that situation. Well, I guess it would, if only it

Hi, When I build the same asterisk package that I build on i386 on x86_64, I don't get /usr/sbin/smsq. AFAIK, the two machines have the same set of installed packages. What should I be looking for in the output of ./configure to get a clue of what might be missing? TIA. -- Bill in Denver

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi,all i install MeetMe module on Asterisk 1.6.2.10. when i use MeetMe to open a conference. even without 'r' option .it will record too. is this the bug of this module? my dialplan is : [95040] exten => 95040263007,1,MeetMe(95040,sM,123) the CLI output is : *CLI> == Using SIP RTP CoS mark 5 -- Executing [95040263007 at 95040:1] MeetMe("SIP/999-00000021",

We've started to use Asterisk for conferencing and have been getting some complaints. Our configuration is that some people call in from home, but we have a physical conference room with a Polycom. When somebody was giving a presentation in the physical conference room, we were told that the remote people kept hearing him cut in and our. To me, this sounds like the talking optimization was

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

http://bugzilla.mindrot.org/show_bug.cgi?id=1168 Summary: sftp fails to HP - UX os even when pubic keys are present in HP-UX Product: Portable OpenSSH Version: 3.7.1p2 Platform: Other OS/Version: HP-UX Status: NEW Keywords: help-wanted Severity: major Priority: P2 Component:

http://bugzilla.mindrot.org/show_bug.cgi?id=1170 Summary: sftp fails to HP - UX os even when pubic keys are present in HP-UX Product: Portable OpenSSH Version: 3.7.1p2 Platform: Other OS/Version: HP-UX Status: NEW Keywords: help-wanted Severity: major Priority: P2 Component:

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

http://bugzilla.mindrot.org/show_bug.cgi?id=1167 Summary: sftp fails to HP - UX os even when pubic keys are present in HP-UX Product: Portable OpenSSH Version: 3.7.1p2 Platform: Other OS/Version: HP-UX Status: NEW Keywords: help-wanted Severity: major Priority: P2 Component:

Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [root at voip asterisk]# find /var/lib/asterisk/sounds/test -name

Hi, I'm having 2 problems: 1) MOH in realtime is not working, I have configured it but never go to look at the database, no warning or error found and I can do a query using realtime and the family from the cli. 2) I have SIP phones via realtime, if I register one of them and a call to a queue comes the call is never delivered to the phone, I have to make a call from the

Hello Asterisk-Users, For some reason my CDR records for disposition and billsec are not working correctly. I always receive a 0 for billsec and the disposition is always at "NO ANSWER', even when I grab the calls. I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22. Here is information on how I do the call: -----------------------------------------------------------------

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =

Hello everyone! I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not working. Here's what happens, if I try to call the line: bach >> P[ 1] --> !! lib: No free channel! P[ 1] --> we have already send Release_complete I haven't changed the configuration fles. Should I change something there? If you need more info, just tell me and I'll