similar to: voicemail comprehension

If you have two asterisk systems how do you hook them up together so the users of one system can make calls onto the other system. Thanks Steve steve@17q.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040927/69058745/attachment.htm

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi All, I have several dozens of CentOS and WhiteBox servers. Most of them are CentOS 4.6. Our installation service is done in the datacenter where the servers are located. When we install a fresh clean install, we use the GUI menus, while using the KVMoverIP. That was working great with CentOS 4.x In CentOS 5.x, the installation process 'annonces" that "Hey,. I know you are

hello centos network ads on the list centos-annonces Thursday the mirrors are still not updated even the main deposit No it has never been so long someone has an explanation can be thanks -- http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x092164A7 gpg --keyserver pgp.mit.edu --recv-key 092164A7 -------------- next part -------------- A non-text attachment was scrubbed... Name:

how can i do to display a message to every wrong number ??? -- Luis Diaz - Un obsesivo con proyectos! :oP

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

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

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

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

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 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

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

My SIP-provider sends my a SIP-invite like this : INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0 Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c Max-Forwards: 70 From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e To: <sip:329298yyy6 at 80.XX.XX.69> Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68 CSeq: 1 INVITE User-Agent: SysMaster VoIP

hi all, can anyone emailme the .conf of asterisk at home, i cant download the full size tar or iso because of a network problem that fu*** every big file download.... and i just wanna learn not change my distro bye and thanks! -- Luis Diaz - Un obsesivo con proyectos! :oP

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

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002

If you remove all the 'static's from the member functions, it'd work more like IRBuilder. It would also allow you to take the LLVMContext& as a constructor parameter, so that methods like this: On Fri, Feb 4, 2011 at 6:57 PM, Talin <viridia at gmail.com> wrote: >   /// GetStruct - return a constant struct given a context and a vector >   /// of elements. >   static

In python, one can do this mydict = dict([(keyfun(x), valfun(x)) for x in mylist]) to create a dictionary with whatever keys and values we want from an input list of arbitrary size. In R, I want to similarly create a list with names/values that are generated by some keyfun and valfun (assuming that keyfun is guaranteed to return something suitable as a name). How can I do this?