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...t and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no joinempty=strict member => SIP/adriana,1 -------------------------------- exten=> 999,1,Answer() exten=> 999,2,Queue(queue|tn) exten=> 999,3,Hangup() Thank you. Rachid

...e (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : anthony at mbdsys.com Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.com

Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : anthony at mbdsys.com Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.com

Hello I'm looking for a wayy to modify extensions.conf It seems that PutConfig AMI command is not supposed to work on extensionsq.conf Any ideas? Thanks Vadim

Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks,

Hi, I have installed asterisk-1.4.4 and asterisk-addon-1.4.1. I followed every step to configure RealTime but something is not working properly; the warning that I am geting is: WARNING[32709]: config.c:1229 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available WARNING[1359]: config.c:1229 find_engine: Realtime mapping for

Hello, I Have an agent on a queue, evry thing works normally, but after a time (about 5 minutes) my agent is pauses (agent is still regitred but can't takes calls), on Astrisk console i have the message : [Jun 20 11:55:12] NOTICE[8803]: chan_sip.c:2757 auto_congest: Auto-congesting SIP/anna-08215f68 -- SIP/anna-08215f68 is circuit-busy -- Nobody picked up in 8000 ms I think

Hi, I have a queue with maxlen=1, and when i make a call, the call enters into the queue, but he doesn't exit from it after a reject signal received from the agent?? please, have you any idea how to remove calls after a reject signal??????? Thanks. Rachid

Sorry, I'm doing a mail test since I was not able to send any mails to the mailing list for about a week... Thanks,

Hi, I hope someone in the mailing list has a good experience in server's configuration requirement since I was not able to make a large scale test to know Asterisk's configuration requirements for my application. So, I'd like to know what kind of configuration the following application should require : Asterisk'll be used as a VoIP server to handle video and audio

Hello, I wonder if somebody can help me with following: I need to acheive something similar to this: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO but with a twist. Suppose i'm already in a meetme conf call i want to dial a * for example, hear the dial tone dial the destination number which will be bridged to the original conf call and return back myself to the original conf

Hello, I'm looking for info about interconnecting asterisk to QSIG GF enabled PABX over PRI . Any information and pointers will be helpful. The very first first question: does asterisk support QSIG BC and GF natively i see that it is supported through CAPI enabled cards but what about support through librip/dahdi? Thanks Vadim

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace: