similar to: Dial + disconnect

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 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

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

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, 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 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, 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 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, 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

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

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call

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

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

Many thanks to Dr Yu for updating Rmpi for R 2.6.0, and for starting to make the changes to support Open MPI. I have just built the updated Debian package of Rmpi (i.e. r-cran-rmpi) under R 2.6.0 but I cannot convince myself yet whether it works or not. Simple tests work. E.g. on my Debian testing box, with Rmpi installed directly using Open Mpi 1.2.3-2 (from Debian) and using 'r' from