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...e being searched in the current working directory. Please guide me on how to tackle this situation. Is there any entry in the windows registry which is set when the software is installed on the windows system and some such equivalent is missing in wine configuration settings in linux. Thank you, Deepesh

...of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh

Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test => **,caller,Macro,testflash Is it possible to do this action on hook flash? -------------- next part -------------- An HTML attachment was

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

Hello, I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to

...are getting the following error: wine : creating configuration directory '/home/raj/.wine' wine : chdir /tmp/.wine-501/server-302-1736bf : No such file or directory wine : wineprefixcreate failed while creating '/home/raj/.wine'. Kindly help me in resolving this issue. Thank you, Deepesh

...Spot Virtual Machine Error, Internal Error # Please report this error at # http://java.sun.com/cgi-bin/bugreport.cgi # # Java VM: Java HotSpot(TM) Client VM (1.4.0-b92 mixed mode) # # Error ID: 5448524541442C4F43414C33544F524147450E435050003A # Kindly help me in solving this problem. Thank you, Deepesh

Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to

Greetings and Merry Christmas, We're trying to implements a queue and agents login mechanism on our Asterisk. After going over the documentation, we're unsure if we got it right. We wish to setup a "hotdesk" mechanism, where an agent comes to a station with a PC & IP phone (that is defined as a sip friend user in sip.conf), dials a certain number (agent login extension),

Hello, I have been trying to setup asterisk (1.6.2.0) to receive fax. I am able to receive faxes sent from a zoiper softphone connected to asterisk. I have some DID numbers (with T.38 support) forwarded to my asterisk pbx. I am not able to receive faxes from these numbers. The error I get on the asterisk console is "phase_e_handler: Error transmitting fax. result=49: The call dropped

Hello, I have been trying to setup asterisk 1.6.2.0 to receive fax. I have two SIP trunks connected to asterisk. One of them is a VoIP service provider and the other is an audiocodes gateway connected with pstn and fax lines. I am able to receive faxes on the DID numbers provided by the VoIP service provider, but I am not able to receive fax through the fax lines connected through audiocodes.

Hello, I am using asterisk manager interface (http) for originating calls. How can I get the name of the channel which is created by originate? I want to use this channel for other manager commands like Atxfer, Monitor, Hangup etc. If I do action=originate, channel=SIP/200 then it creates a channel like 'SIP/200-0865ff80' which I can see in the asterisk console using "core show

Hello, I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP settings in sip.conf in this version also. I am facing a problem when a SIP client makes a call. When a SIP client registers to asterisk its status shows 'OK' and it is able to receive incoming calls. But as soon as this client make a call, its status becomes 'UNREACHABLE' and it cannot receive any

Hello, How do I use multiple postgresql databases using res_config_pgsql? I tried creating multiple contexts in res_pgsql.conf, but asterisk is only using the 'general' context. My res_pgsq.conf is [general] ;; Connect to mydb on localhost dbport=5432 dbname=mydb dbuser=pgdbuser requirements=warn [pgwritedb] ;; Connect to mydb2 on another host dbhost=<IP