similar to: Incoming calls coming into default context

Folks! I dont know whether anyone has done this exercise before of putting together a Wish-List of things that you want to do, if you have all the gadgets you need and have a client base that needs Asterisk's Features and more. Here are some of the scenarios I am playing out that I will do, once I have enough time. Can anyone add to this list of Scenarios with or without using other gadgets,

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _____ From: Mani Nair [mailto:mnair at nvloisp.com] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk Hello Guys, I am unable to make calls to outside number from some of my extensions.

Hello, I want to implement a perl logon script which would map network shares depending on group membership. This way when I move a user to or from a group it automatically gets the new shares. The PDC is Samba 2.2.1 or 2.2.2 on BSD and clients are NT workstation and 2000 Pro. Perl is latest activestate for win32 and is intended to run on Win32 clients. The big trouble is getting user Group

Is there anyone with experience how to integrate Sysmaster's VoiceMaster? Please can you share your experience. Thanks. Habiyakare Aimable Voice Services Terracom Communications Tel :(250)08435550 SIP:04400104@voice.terracom.rw E-mail:aimable@terracom.rw MSN:aimable@terracom.rw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Hi folks Is it possible to turn off the native bridging on Asterisk? I've been hacking about app_disa.c to support account & pin numbers, that tag the calls depending on who logs in..... It all works fine, then dials the destination number they requested. My setup is as follows [ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN) If i dial

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

On machine A I have something like the following in extensions.conf: [iax-extensions] exten => _9.,1,Dial(IAX2/machineB/${EXTEN:1}@mycontext) exten => _9.,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => _9.,3,Hangup On machineB I have something like this: [mycontext] exten => 2002,1,Dial(SIP/2002,60) exten => 2002,2,NoOp(DIALSTATUS=${DIALSTATUS}) exten => 2002,3,Hangup If I use a

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

Per the wiki: Logging off 1. call the extension for AgentCallbackLogin 2. enter your password followed by # 3. when asked for the extension number just press # But if your exten=> is this: exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext) How do they logoff per the wiki's directions? If you use ACBL as above, it never asks you for the extension number because you have

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Sorry to resurrect an old thread, but I finally got around to testing this approach (round tripping through bitcode in memory) and it works beautifully - and isn't that much slower than cloning. I have noticed however that the copy process isn't thread-safe. The problem is that in Function, there is lazy initialization code for arguments: void CheckLazyArguments() const { if