search for: lastcalled

Hi,all, Now,I am working at make an realtime monitor for the call center based on asterisk. and ,I had search the archive and wiki.Through the return info from the management API, I can get the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get the number of incoming calls. The info like following :

I've tried every combination I could find on the net and so far there is no joy The thing is I can do this update from the command line: Maybe some new eyes might find the answer? exten => update,1,MYSQL(Connect connid localhost root password dax) exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\

Hi all, What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Thanks ! -- Warm Regards, Lee "Everything I needed to learn in life, I learned selling encyclopedias door to door."

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

Andi - I have experienced the same issue you mention and gotten no reply as to a way to fix it. I finally implemented "blacklist" into my Asterisk and added "Anonymous", "anonymous", "unknown", "Unknown", etc., into my blacklist file. When those come in with an IP address instead of a phone number but have no real name, they get the

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Hi all, Where can I find the status definitions for the status field below? Googled /web/voip-info.org and can't seem to find anything. Event: QueueMemberStatus Privilege: agent,all Queue: support Location: SIP/111 Membership: dynamic Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 Thank you! -- Warm Regards, Lee

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

The Asterisk Development Team would like to announce the release of Asterisk 13.35.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.35.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.35.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.35.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Anyone thought about something like outgoing queues? I mean, having same info that has for inbound queues but for outbound calls, and grouping members there. For example, before using dial application put an app outqueue that get all the statics. Talked time, member status, last call, completed calls, failed calls, reset statics, and maybe some more. So its possible to get more control and has

The Asterisk Development Team would like to announce the release of Asterisk 16.12.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.12.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.6.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: