similar to: QueueMemberStatus

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

Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks

I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee

Hi everyone, I just installed a TDM02B and surprisingly, I had really no problems except one. If I place an outbound call on the Zap line (Zap/3), everything works fine except when the called party hangups before I do. I do get congestion, but that is expected. However, when I try to make another outbound call using that Zap line, the CLI shows that the call is being dialed, but nothing

Hi Everyone, I'm wondering if anyone here write AGI's in compiled binaries. I'm writing a small Cepstral AGI in Freepascal/Lazarus. I know there are some other AGI's out there, but I wanted to add some more functionality than what is available such as having the AGI determine if the "data" argument is plain text or a path to a text file and act accordingly. The

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

Announcement: We are pleased to announce that we have released AsterPas FastAGI ObjectPascal Script Server for Asterisk PBX under xxxx license. What is AsterPas? AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX call flow using ObjectPascal based scripting. AsterPas includes many built objects available from scripts such as Cepstral TTS Engine class, database access

Is the main difference between the two the full duplex speaker on the 330? -- Warm Regards, Lee

Hi all, I just finished writing the bulk of an AGI interface to FirebirdSQL databases and I noticed that when assigning a variable through AGI (I assume this also applies within the dialplan), you have to enclose it in quotes if there are any space. Does Asterisk strip off the quotes when storing the value? Just curious. Thanks, -- Warm Regards, Lee

Hi all, I just purchased a Polycom 301 for my home office and I believe I have it setup correctly as I can dial out, receive calls in, etc. However, I'm having the following issue: When calling a local number over a Zap line, I hear a lot of feed back on the line. I had a Grandstream configured with the same information before I got the 301 and never had that kind of feedback noise.

Hello Everyone, I can see that a few people are interested in SIP WIFI phones. I have tested several Linksys 300,and it is OK. More of a toy then a business tool. It a poor built in ear speaker, which makes all calls sound tinny, and the unit is known to hang. I have two Linksys 300's that are fun to play with however, I wont hand them out to users. HOWEVER- The Zultys WIP 2 is an

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Hi all, I've been doing some research on FastAGI and there are few things that I am not sure of from wiki, etc. 1. When using FastAGI are the initial env vars sent all as one packet or are they sent line by line? 2. If line by line are they also terminated with a line feed as is regular AGI? 3. This just needs verification: Passing Variables in 1.2.x. I see that in 1.2 the variables

Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel: SIP/MyProvider/9105555555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: myCallFileContext Extension: s Priority: 1 Set: myVar=MyNewValue and... [myCallFileContext] exten=>s,1,NoOp(${myVar}) ; <====== myVar is empty

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

Had an appointment for these schmoes to come out and install another line. Was supposed to be 8-12. Its now 6PM and not even call. Missed 3 sales calls waiting on these jerks. No wonder customers were jumping ship to Vonage. -- Warm Regards, Lee

Hi all, There's been a few posts looking for telemarketer torture scripts so I figured that I would write one using a SQLite db. Handling an incoming call that is flagged from the database is pretty simple. My problem is that I would like the callee on an established channel to be able to redirect the caller to a specific context where my AGI is called and handles the call by first

Hi all, Is there a way to tell which extension transferred a call in a blind transfer? Sorry if it's a basic question, but I haven't seen an answer. ${CALLERID(num)} still holds the outside party caller id (which it should), but I'd like to the extension number of the extension that transferred the call. Any suggestions? Thank you, -- Warm Regards, Lee

Hi all, I'm having an odd problem with my polycom 301. I am initiating a call to it with AMI Originate() function: Action: Originate Channel: local/112 at Management Context: to_meetme Exten: s Priority: 1 Variable: dropped_conf=111 The "to_meetme" context is very simple: [to_meetme] exten=>s,1,MeetMe(${dropped_conf},id) If I specify every other device I have to test: *

Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it