search for: cheefoong

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

Hello, I have a Digium TE410P card. I get the "knight rider" lights before the module (wct4xxp) loads, but after the modules are loaded I don't get any lights. I have found the following 2 posts but still could not solve the problem http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

Helo WipeOut, I have found a solution for sending dtmf after dial. I use spooling. Take a look at the sample.call file inside asterisk dir. You need to edit this file and dump it in /var/spool/asterisk/outgoing. Asterisk will precess this file automaticlly I create the sample.call do something like this: Channel: OH323/4324324324 #dial the access way MaxRetries: 3 RetryTime: 60 WaitTime: 30

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and

Hello All, I wonder is there a way where I reload asterisk on CLI without disconnect any call that is currently taken place. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030813/41f0a4ca/attachment.htm

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

Hello, Is meetme able to do the following scenario: Say a group of caller calling in to the same conference room where one of them is a conference leader. The leader can press a key which gives that person options like dial out to a particular person and tranfer him/her to the conference. The leader's password is different from the other. I think the tricky part in this scenario is how to

Hello, When I have the following extension: exten => 900,1,dial(Zap/0122740900) can I know whether 'dial' actually gets through or the called party is busy at the moment. I want to perform different action based on whether the 'dail' success or not. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

Hello, I am recently inspecting the IAX protocol.. I wonder if there away to associate a user name to a number say I have a client register to the IAX server with username 'John' and I want to associate a number say '12345678' tho John so other register users can call john by dialing 12345678. Something like the H323_id and the E164 alias in H323 protocol. Foong --------------

Hello, I have been using IAX to serve clients endpoints for a while with no problem. But recently, to increase the bandwidth to the Asterisk server, I add another network interface card to my Asterisk server which connected to a different service provider that I currently have. Both of my nic is assigned different public ip. the client will actually choose one of these ip and authenticate itself.

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hello, I try to append a URI to the SIP dial syntax, however the URI were not shown in the sip debug message. I have read one of the post in the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten

Hello all, I am using flash operator panel, when i stop iptables everthing is fine, but once iptables is started, the operator panel doesn't work anymore. Anyone know how to set up the iptable in order for to op panel to work? I am using tcp port 5038 for asterisk manager, and I have try open both tcp and udp port 5038 in iptables but without success. thanks CCF

Hello, I have see a post in the list saying that the 'daemon' command should be remove from the asterisk startup script in /etc/rc.d/init.d/ for FC2 in order for asterisk to auto restart when crash. I wonder if this should be done on FC3 as well, because my asterisk did not restart when crash. Please help Thanks CCF

Hello all, I am trying to use asterisk to call a local access gateway by dialing a fix number, after getting connected, the is a IVR prompt for pin number and finally the real destination number. I manage to use asterisk to dial to the gateway but have no idea how to send the pin number and destination number. This is due to asterisk only process the next ext only if dial app has terminated. My

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that