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

...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 digium's g.729 is not compatible with these codec found on cisco AS5300. Am I correct? Any advice will be helpful Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/c88a161b/attachment.htm

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

...number' right a way. I am using chan_oh323, I am close to get this thing to work (having sorted the correct codec), just the dtmf issue now. I am using digium's g729. By the way how many variation of g729 are there. I know g729a, g729b, but there seem to be others. Please help. Thanks Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/7d38e04d/attachment.htm

...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 not depend on whether there is a zaptel device. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030812/0a9392d7/attachment.htm

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20030909/eec7cd58/attachment.htm

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

...onf) iax module to listen to all interface on the asterisk server. I wonder if ther is anybody having the same problem like mine when there are 2 nic on a Asterisk server and would like to share your findings and experience. Is iax designed to handle multiple network interface in the first place? Foong

...y process the next ext only if dial app has terminated. My extension.conf are as follows: [test] exten => _0XXXXXXXX,1,Dial(H323/${EXTEN:0}) exten => _0XXXXXXXX,2,SendDTMF(PIN_NUMBER_HERE) .... I saw someone post the similiar question but with no reply. Does anybody has any idea? cheers Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/81ddeb98/attachment.htm

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

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

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030926/4d51fdd4/attachment.htm

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

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

I just wondered whether anyone actually got this working and produced a how-to ? I recently had a customer ask about embedding it into their web pages for there customers to call them with ?? To be honest I have no idea how etc.... Gary .

...${NUMTOCALL}) However, this might not suitable for you, if your user dial in manually. My situation works fine cause everyting is automated where calling number and called number is inserted into db in advanced. also, chan_h323 has proplem sending DTMF, chan_oh323 works but sound quality is bad. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030805/b6d2ecf5/attachment.htm

This looks rather interesting. They also have an IP phone which is probably low cost, but it seems to only support G.723. Has anyone used any of these products? http://www.nicstel.com/2001/e_3023w.html http://www.nicstel.com/2001/e_products02.htm -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>