similar to: Meetme Admin menu

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

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

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

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

Hi, I try to use oh323 package from inaccessnetworks for asterisk, but after make and make install that package, I have this WARNING message hwen a try to launch asterisk from shell command line...asterisk -vvvc... [liboh323wrap.so]WARNING[1024]: File loader.c, Line 235 (ast_load_resource): No load_module in module /usr/lib/asterisk/modules/liboh323wrap.so

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and

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, 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, 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 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

I'm having a problem involving Outlook and .pst files and a lock that is getting "stuck" I believe. Once Outlook crashes in the fashion it does, it is unable to reopen the file, claiming it is already in use. Explorer also does not let me access the file. Rebooting the workstation does not fix it. smbstatus does not show the lock, however hwen I show byte range locks, there is one,

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, 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

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 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

Hi all I see the rsync source and rsync makes hashing table and search hashing table tag_table to find the index of array struct sum_buf , which is a element of struct sum_struct. According to the source code, variable last_i is used to encourage adjacent matches allowing the RLL coding of the output to work more efficiently. Why last_i makes more efficiency? I can't understanding what

I am trying to get my latest installation of Centos (4.4) to see videos when I am in www.cnn.com. I installed mplayer using yum and that went well. But I still get the screen from microsoft saying I need to install their plugin.. What more can I do to get these videos to work when I use either mozilla or konqueror.. Thanks..