similar to: RINGNOANSWER IN queue_log

Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing something really stupid which could cause a segfault under certain conditions. My chan_queue.c addition is this one line: ast_queue_log(queue, qe->chan->uniqueid, outgoing->chan->name, "RINGNOANSWER", "%d", orig);

my environment： 192.168.0.13 puppet.uc.local 192.168.0.14 puppetca.uc.local 192.168.0.15 report.uc.local 192.168.0.16 fileserver.uc.local 192.168.0.17 agent01.uc.local i want run a master as fileserver (fileserver.uc.local) the puppet.uc.local and fileserver.uc.local use one ca.pem on puppet.uc.local, i wrote a class for test class test { notify { "hello

Hello All, I have a queue with only one agent logged in al the time, but if for some reason the agent cannot pick up the call for 2 full ring, the phone does not ring the 3rd time and all the call in the queue get stuck. Below is my agents.conf [general] persistentagents=yes multiplelogin=no [agents] wrapuptime=5000 musiconhold => default updatecdr=yes agent => 1001,4321,Agent01 While

Hi, I have a problem..I have 3 agents active in my queue..How can I get list off all my active agents who was available when a call come? in my extension.conf : (1) exten = 6501,n,Queue(${EXTEN},ntT,,,1) (2) exten = 6501,n,Queue(${EXTEN},ntT,,,25) after the first one running, if i set like that in my extension.conf, in queue_log I can see ringing agents before entering the second queue

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently

Hi, I've noticed that when I set a phone on DND (phone-side DND, meaning it rejects calls with a busy status, SIP 486 response code I believe) the queue keeps on trying the phone over and over again. This creates issues in terms of CDR entries - in a scenario where there is only one phone on DND, and a delay between attempts of 1 second, the queue will attempt to ring the single phone

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is

? Hi, From asterisk 1.8, the CDR table is not logging the unanswered or extn busy calls which hit while in the queue. I am talking about this setting in the cdr.conf : ; In brief, this option controls the reporting of unanswered calls which only have an A ; party. Calls which get offered to an outgoing line, but are unanswered, are still ; logged, and that is the intended behaviour. (It also

Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David

Hello Team, Can you help me on this? I have attached here the queue logs of my asterisk. I've searching a parser for this. I do not know what are the meaning of that logs. Thank you so much. Your response is highly appreciated. Regards, Cristina -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Asterisk Development Team would like to announce the release of Asterisk 15.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0. This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this beta:

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.3-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.3-cert1 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following

I have a Bridged call with 2 parties. I want to redirect one party to a conference room and the other party to an outside number. I tried doing that with a dialplan. I used ChannelRedirect in the dialplan and redirected the first channel to the conference room. however, the second channel disconnects. Reading thru the mailing list i understand that this expected. However, I don't understand

Hello, I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got