similar to: Regarding Call Hold

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

Hi, I have installed a TDM400P in a PC which runs RedHat Linux 3.2.2-5. There is an FXS module connected to the first slot of TDM400P and an FXO module connected to the fourth slot. This card is detected by the operationg system, which I was able to make sure using the output of lspci command. I modified the configuration files zaptel.conf, zapata.conf and extensions.conf. When I execute the

Error in file(file, "r") : unable to open connection In addition: Warning message: cannot open file 'a:/project.txt' please help --------------------------------- [[alternative HTML version deleted]]

Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to

This might sound lika a "small" issu, but here it goes: I'm a long time unix user and my shell history usage and editing is configured to use vi keybindings; it's something that's already built into my fingers and using different bindings, like the arrow keys to fetch previous lines, really blows me !... :-( Is there any way to setup the asterisk cli to use such

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected ("estimated hold time is less than 2 minutes ..."). Now the caller gets an announcement of their sequence in the queue

Hi, Im looking to put together hopfully a more useful api documentation site, more like php.net does anyone know where I can get hold of the api as a database dump thanks scott -- Posted via http://www.ruby-forum.com/.

I''ve got a belongs_to / has_many relationship working, however I don''t know if I can still use content_columns (or any other shortcut) in my view that displays my object. E.g., Employee belongs_to Department I''m displaying an Employee. Will content_columns contain employee.department.name? Or how else could I do this so I don''t have to hard-code column

When we set/clear the dyn_features for an inode we hold the ip_lock. So do it when we set/clear OCFS2_INDEXED_DIR_FL also. Signed-off-by: Tao Ma <tao.ma at oracle.com> --- fs/ocfs2/dir.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/fs/ocfs2/dir.c b/fs/ocfs2/dir.c index 07d8920..8e0f370 100644 --- a/fs/ocfs2/dir.c +++ b/fs/ocfs2/dir.c @@ -2463,8 +2463,10 @@

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

Hi! I use Bria and eyebeam and it seems that asterisk doesn't send RCTP keepalives when a SIP channel is on hold. This is a known issue as is described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Bria This gets very annoying because very often people are put on hold longer than 30 seconds (the phone's default.) In a company with more than 100 soft phones

Hi, when I put a call on hold from my Nokia E51 (SIP client), the other side does NOT hear music on hold although sip debug / wireshark shows that the E51 tells the asterisk that it now holds the call. Canreinvite is set to "no". Also, when parking a call (features.conf), the parked caller does not hear music on hold. In queues, when using "#" and when using the hold

I want to enable on hold reminder function on polycom 430 phones. I have enabled it in sip.cfg using this setting <hold> <localReminder call.hold.localReminder.enabled="1" call.hold.localReminder.period="60" call.hold.localReminder.startDelay="90"/> </hold> But still if the call is on hold the phones does not remind about the on

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

Hi List, How to put multiple call on hold by dialplan in asterisk? -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110517/0396c10f/attachment.htm>

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

Hi All, Is it possible o know how many call are into hold ? who are on hold ? By whom these extension are on hold ? And after 60 sec asterisk will call them automatically as Call Parking do? I wan to make this concept to my PBX system... Thanks in advance -- ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment

I am a novice with network fuctions! I have been exploring the network function in the statnet package, but haven't been able to figure out how to hold vertices in position while varying edge features. Can anyone advise on whether this is possible, and if so, how to do it? Thanks! -- Matthew Bakker, Ph.D. Department of Plant Pathology University of Minnesota 495 Borlaug Hall 1991 Upper Buford

Hi all, Does asterisk (I'm using 1.4.19) sets any channel variable with the holded chan when it does an atxfer? I tried to see if it does on the source, but i didn't find any clue, neither enabling console debug. Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: