similar to: Hangup half a call?

I'm in search of useful ACD type statistics from the queues. Ie talk time, ratio's, dropped calls etc. The flat file queue_log is nice, but more useful would be the data in Postgres or Mysql. Unfortunately the queue module does not yet support ODBC DB logging (yet). In the meantime this quick and dirty hack gets the job done. Replace the flat file with a unix named pipe.

Anybody ever put a Sangoma and a Digium card in the same server? Specifically a four port card from each company? -bill wlloyd@slap.net

I too have been having SIP problems the last couple of days, I get the same message as Louis-David but in this setup only: PSTN >-ISDN-> AS5300 >-SIP-> * I can make outbound calls no problem but inbound calls seem to stall, according to 'sip debug' it just says 'Ignoring this request' but I cant establish why .... > -----Original Message----- > From: William

I've got a Zhone 24 port FXS to configure. The configuration is beyond stupid. The people that designed this unit should be chased down and fired. I'm going around in circles frigging with all the options. Does anyone have a config file for this unit that I can use as a starting point? -bill wlloyd@slap.net

I'm trying to get a patch tested for inclusion in CVS. Anyone that is running E&M on a T1 and had to fool around with emdigitwait could you please try this. This patch removes the need for the emdigitwait parameter and speeds up dialing. This situation is mostly interfacing a legacy PBX/Key system with Asterisk. I've been running a couple of systems with this patch for

I'm sure this is easy and I'm missing something, but how can I add an extension into the directory command (ie get's it's list of extensions from voicemail.conf) yet have no voicemailbox for the extension? Basically I have an extension that gets forwarded onto somebody cellphone where they use their own voicemail. I'd like to be able to list them in the company

I've been trying to work with the dynamic feature support.. IE adding codes like *2 to features.conf that can trigger a dialplan application to run. I've been unable to get "goto" to work properly. "AGI" also seems to not function correctly if called as a feature. Anyone else playing around with this feature might have some insight? -bill wlloyd@slap.net

Hi, I found out that GoTo in applicationmap is not working. OK, LOCAL is working. but I expected that applicationmap is using the DIAL option tT. But it doesnt, it works without tT Option, so also callee can use internal functions if callee knows the code. Any workaround avaiable? best regards Thomas

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Hi - I'm trying to implement the applicationmap stuff in features.conf, and I can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom IP500s and Snom190s. My features.conf looks like this: [general] parkext => 700 parkpos => 701-720 context => parkedcalls parkingtime => 240 transferdigittimeout => 2 ;courtesytone = beep

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

hello, I use trixbox.I had define a feature code testfeature: [applicationmap] #include features_applicationmap_additional.conf testfeature => *3,callee,Macro,vote [featuremap] blindxfer => ## ; Blind Transfer disconnect => ** ; Disconnect Call automon => *1 ; One Touch Record atxfer => *2 ; Attended Xfer testfeature => *3 here is my macro-vote: [macro-vote] exten

I'm having trouble with both my Polycom IP600 and IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a phone call that lasted 1hr 39mins which is my record so far. I cannot figure out what is causing the call to terminate and I am hoping somone on this list can help me. In this example both the phone and the asterisk server have public IP addresses so NAT shoul not

Hi everybody, I would like to set asterisk up such that to use the call parking feature, instead of transferring a call to the extension set up in features.conf, you just dial a code (e.g. *3) and this then parks the call. The main reason for this is that a number of the phones I use have transfer buttons that I can't reprogram to use Asterisk's own transfer functions, therefore you

Hi Kevin. Thank you again for help me! In my case, in the final application for smartphones or in a softphone for PCs, there will be a button on the GUI and the user will have just to touch it, and the door or gate will open. I mean, during an ongoing call, the callee will see a button in the interface of its SIP application. For example, we can use the lib of Linphone and implement a GUI over

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able

Anyone know how (or if it's possible) to get a logo on the screen of a Polycom 501? I've been looking around for hints on how to do it but so far nothing would indicate Polycom supports doing it. -bill

Problem resolved. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife. Now all she has to do is pres *3 and it is done. No transfers. No hanging up. No dial back. extensions.conf [context] exten =>

Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller => #9,callee,goto(521|1) show features - Dynamic Feature Default Current ---------------

> Hi Kevin. > > Thank you again for help me! > > In my case, in the final application for smartphones or in a > softphone for PCs, there will be a button on the GUI and the user > will have just to touch it, and the door or gate will open. I mean, > during an ongoing call, the callee will see a button in the > interface of its SIP application. For example, we can