similar to: Asterisk Manager

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

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG

has any one installed this, i just tried this on a test server, i get voice but it's corrupted, i do not get the natural voice any idea why -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/af5a3bb2/attachment.htm

Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give. I use realtime asterisk. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an intermittent problem with dangling MS Access DB lock files. In a productive environment with N batch queus (each on a separate Windows XP Professional) a scheduler PC dispatches the work load to a free queue by means of modifying a simple MS Access DB file called "PRIM.mdb", which resides on a Samba 3.0.20 share. Each free queue PC polls the same MS Access DB file every 60

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, 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

A last week there was a message about using Samba with HP Jetdirect cards. The Jetdirect cards don't multithread, so when samba does an lpq- there is a "hang time" for the client. I understand that. But, why is NT smart enough to put a message in the event log saying that the printer is busy as where *unix/samba causes the client to page me or come to my desk and wine "How

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

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: