similar to: timeout with queue

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

I have setup several Calling Queues, each setup with RoundRobin strategy. When I call the queue, the first member/agent phone rings. Great! I call it again, the second member/agent rings?? I thought that was the RRMemory strategy, but it seems RoundRobin is also doing it. Anyone know what I can do to my queues, in order to force each call down the ordering of my members list? --------------

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf I have in /etc/asterisk/capi.conf 's section "[interfaces]" the following directive context=isdn and the following directive in /etc/asterisk/extensions.conf in

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

I have set up an operator queue for our receptionist. That way, if she takes a break or is out, by logging out of the queue, calls to the "Operator" can be handled by other agents. I have set strategy = roundrobin in queues.conf. According to "the book" ATFoT, roundrobin always starts with the first agent in the queue. This is the desired result. I want all calls to start

I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

it is possible to use musiconhold. i added exten => s,5,SetMusicOnHold,default exten => s,7,dial,SIP/michael&SIP/frank|15|m to the extension conf. the logfile looks good, i think -- Called michael -- Called joern -- Started music on hold, class 'default', on CAPI[contr1/9999] -- SIP/192.168.100.150:5060 is ringing -- Stopped music on hold on CAPI[contr1/9999] --

Hello, I am implementing a small call center with 1 to 4 agents. For some reason I don't understand, if an agent is busy, and it is his/her turn in the queue round, asterisk gives an "all destinations are busy" message and hangs up the call. Agents are SIP lines registered with an audiocodes MP108FXS which registers each line independently. Ringing strategy is RoundRobin (most of

Hello list, I have been researching a bit into the way the queue app works and how different timeouts play together, and have prepared a short tutorial on understanding queue timeouts - see http://www.oinko.net/astrecipes/index.php?n=118 - any suggestion, error found or correction is welcome. While I was at it, I came across a strange bug: imageine you have three callback agents

Hello all, I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA) on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one B channel. But when I open a second B chan, the sound is choppy, with too long gaps, and the CPU load is too high (~50%). On the Asterisk's console I get these messages: -- Executing Dial("H323:4478",

I am trying to have the ring types different for internal and external incoming calls. I have followed the guide on the wiki, the SetVar executes, in extensions.conf I have it as s,1, Yet it doesn't work? When the phone rings, the ring type is the one I chose on the phone, it rings same tone for both when I test. Using Asterisk Stable. Anyone got this working and can

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

I''ve been having a mess of a time using this module, which stinks because its behavior is EXACTLY what I am looking for... whenever I bootstrap new rabbitMQ nodes I want to add them to our HAProxy instance. Here''s my relevant site.pp entries: node /^rabbit.*/ inherits basenode { @@haproxy::balancermember { $fqdn: listening_service => ''messaging00'',

Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent -> 2nd Agent -> 3rd Agent I've actually solved that by defining penelty for the accounts,