search for: sgifford

...ad to the agent. I'd like to do some slightly more sophisticated things, like run an external application with System(). When I was using normal extensions and routing the call to one person, I could do something like this: exten => 3772,1,Ringing() exten => 3772,2,System(/home/sgifford/ircsay sgifford "Call for ${EXTEN} at ${DATETIME}") exten => 3772,3,Wait(2) exten => 3772,4,Dial(SIP/sgifford) to run an external application and wait 2 seconds while the caller still heard ringing. Is there a way to do something similar when a queued call is delivered? M...

...contains about .06 seconds of silence. If I talk for another minute, this file will get up to 200 bytes or so. In my queue configuration, I have: [testq] monitor-format = gsm monitor-type = MixMonitor ... I can see what looks like MixMonitor starting and stopping at the right time: -- IAX2/sgifford-3 answered Zap/1-1 == Begin MixMonitor Recording Zap/1-1 -- Hungup 'IAX2/sgifford-3' == Spawn extension (incoming, 3772, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' == End MixMonitor Recording Zap/1-1 I have tried turning debugging up very high (like 5...

...ers Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting MixMonitor options from Queue I know this is not what you need, but you might postprocess recordings to raise the volume level. I know this is not optimal but it's a start. l. 2010/1/21 Scott Gifford <sgifford at suspectclass.com> Hello, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonito...

Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.

...e, but I get these messages in the logs: [Jan 17 11:21:07] WARNING[2413] chan_zap.c: Unable to request echo training on channel 1 [Jan 17 11:21:13] WARNING[2413] pbx.c: Zap/1-1 already has a call record?? [Jan 17 11:21:35] WARNING[2413] /home/sgifford/src/agx-ast-addons/app_txfax.c: Transmission loop error When I send a fax to a line that's busy, I get: [Jan 17 11:40:29] NOTICE[2439] pbx_spool.c: Call failed to go through, reason (0 ) Call Failure (not BUSY, and not NO_A...

Hello, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonitor application, we would just add the "v4" option to make it much louder. I don't see a way to set this option when

HI, Anyone knows the current status of JIAXclient? I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. I tried to e-mail the author but seems that his account is no longer valid. I in need of a java IAX client that could be loaded as an applet. I know that is a lot of viable SIP alternatives, but due to NAT/Firewall

...91234567, 'WaitTime' => 20, 'Action' => 'Originate', 'Application' => 'txfax', 'ActionID' => '1248244247.1814', 'Priority' => 1, 'Data' => '/home/sgifford/prog/faxscripts/testfax4.tif', 'Variable' => '' }; RESPONSE: { 'Message' => 'Originate successfully queued', 'ActionID' => '1248244247.1814', 'Response' => 'Success'...

Hello, I ran into an interesting problem earlier today. I have a Unix machine I maintain in a largely Windows shop. They use Windows Backup for their backups, and so I created a readonly share of the entire filesystem with one user, "backup", who is an admin user. This lets them back up the entire Unix machine by attaching to the "backup" share, but nothing can be changed.

I was just thinking how they could be easily supported. This would work, right? : imap stream tcp nowait root /usr/sbin/tcpd /usr/local/libexec/dovecot/imap-login imaps stream tcp nowait root /usr/sbin/tcpd /usr/local/libexec/dovecot/imap-login --ssl imap-login would try to connect to master process using some named socket. If it couldn't, it would create the master process itself. Master

Hello, We've got an Asterisk server with a Digium TE110P card, connected to a Tadiran Coral Flexicom IPX 500 switch using a T1 card. We are having echo problems on the lines coming in from the Digium card. I was wondering if anybody is successfully using a Digium card and Asterisk with a Tadiran switch, and if so whether they could share some configuration information? Thanks! ----Scott.

Hello, I have an application that needs to record outgoing calls. It's running on Asterisk 1.4.18, with CDR data stored in MySQL. Outgoing calls are recorded based on their uniqueid. When outgoing calls are placed, there is a line like this on my extensions.conf: exten => _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm) For regular outgoing calls, this works fine. The

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

Hello, I'm trying to implement multi-party calls according to these instructions: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO They are almost working, except that the Goto at the end of [dynamic-nway-start] doesn't seem to work. When I turn verbosity up a bit, I get something like this in my error log: == Channel 'SIP/SWG-0085a180' jumping out of macro