similar to: ** Asterisk Sunday Morning News: Contribute to the community

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

Hi, I'm very new to Asterisk and I have the following scenario. 1. Let's say I have a number of 1-222-222-2222 from my SIP service provider (VoicePulse). 2. I point my phone, Verizon wireless cellphone (1-111-111-1111), voicemail to the number provided by SIP service provider (1-222-222-2222). 3. I use another phone (1-333-333-333) to call 1-111-111-1111 and leave a voicemail message.

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

Hi all, most of the freedesktop.org servers just got new kernels, so they'll be rebooted on Sunday morning UTC. Sorry for any service disruption, downtime should be minimal. Regards, -- Tollef Fog Heen UNIX is user friendly, it's just picky about who its friends are

>From somewhere in Cyberspace to all points on the compass its the On Computers Radio Show. Sunday's show will be all tech talk with the team. Peter Kastner will start off the show by talking about this week's hot news stories then the gang will continue on with tech talk. During the second hour, we will be joined by Kerry Garrison of the <http://geekgazette.com/> Geek Gazette

Hopefully I'm not just missing some little detail here. We're trying to set the HANGUPCAUSE on SIP channels to have our softswitch play the proper recording instead of answering the call on Asterisk to play the message. It appears that no matter what the HANGUPCAUSE is set to, Asterisk always just sends "603 Declined". I looked through the source code briefly and it appears

Spring is back in the Stockholm area. After a few day's worth winter re-runs, the sun is back and night-time temperature is at least 5 degrees celsius. Time to move out all my annual flowers and prepare the garden for summer. Sweden is famous for our annual five week holidays - by law. From june to late august, it's almost impossible to make any business decision, since there's always

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

> > Shot in the dark here: > > What are you using for the sharenfs value on the ZFS filesystem? Something like rw=.mydomain.lan ? They are IP blocks or hosts specified as FQDNs, eg., pptank/home/tcrane sharenfs rw=@192.168.101/24,rw=serverX.xx.rhul.ac.uk:serverY.xx.rhul.ac.uk > > I''ve had issues where a ZFS server loses connectivity to the primary DNS server and

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

> > Offlist/OT - Sheer guess, straight out of my parts - maybe a cronjob to > rebuild the locate db or something similar is hammering it once a week? In the problem condition, there appears to be very little going on on the system. eg., root at server5:/tmp# /usr/local/bin/top last pid: 3828; load avg: 4.29, 3.95, 3.84; up 6+23:11:4407:12:47 79 processes: 78 sleeping, 1 on

On Sunday, August 18, 2013, the Asterisk community services listed below may have intermittent availability due to network maintenance. This maintenance will begin at approximately 2:00 PM CDT[1] and should last no longer than two hours. The affected services are: * All community services provided by Digium, including, but not limited to: * bamboo.asterisk.org * code.asterisk.org *

This Sunday several community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 10:00 AM CST[1] and should last no longer than six hours, ending around 4:00 PM CST. The affected services are: * JIRA issue tracker (issues.asterisk.org) * Proxy for various community services, including but not limited to: ** wiki.asterisk.org **

Tonight a few community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 10:00 PM CST[1] and should last no longer than one hour, ending around 11:00 PM CST. The affected services include but are not limited to: * bamboo.asterisk.org * reviewboard.asterisk.org This Sunday, several other services will undergo maintenance as well. This

Sunday night all asterisk.org services may have intermittent availability due to maintenance. This maintenance will begin at approximately 10:00 PM CST[1] and should last no longer than one hour, ending around 11:00 PM CST. The maintenance is expected to last only a few minutes. The affected services are: * Everything in the asterisk.org domain. Thank you for your support! [1]:

Tomorrow, Sunday, community services may have intermittent availability due to maintenance. This maintenance will begin at approximately 2:00 PM CDT[1] and should be complete before 8:00 PM CDT. The affected services include at least: * git.asterisk.org * issues.asterisk.org * crowd.asterisk.org * downloads.asterisk.org * downloads.digium.com * packages.asterisk.org * asterisknow.asterisk.org

This Sunday several community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 10:00 AM CST[1] and should last no longer than six hours, ending around 4:00 PM CST. The affected services are: * JIRA issue tracker (issues.asterisk.org) * Proxy for various community services, including but not limited to: ** wiki.asterisk.org **

Tonight a few community services will have intermittent availability due to maintenance. This maintenance will begin at approximately 10:00 PM CST[1] and should last no longer than one hour, ending around 11:00 PM CST. The affected services include but are not limited to: * bamboo.asterisk.org * reviewboard.asterisk.org This Sunday, several other services will undergo maintenance as well. This