similar to: Abandoned queue calls do not produce a CDR?

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8. I do not pass the 'n' option to any call to Queue() in my dialplan. Yet since I upgraded to 1.4.9, I have occasionally seen this on my console: -- Nobody picked up in 20000 ms -- Exiting on time-out cycle That log message "Exiting on time-out cycle" is exclusive to the logic in app_queue meant to

Hi I'm using Ast 1.0.3, and managing a Queue for ACD. Our Callcenter supervisor uses Flash Operator Panel to see status of the Queue, and logged in Agents. All calls that stay to long in the Queue gets into a Voicemail, in order to have customer leave a message. 1. I see the Queue has a Completed and Abandoned counter. I have not found a exact definition of these, besides the obvius ones.

I went to run my queue_log parser so that I could send out a monthly report to one of my customers, and I noticed that every valid call complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an ABANDON: Here is a complete-caller: 1123325015|1123325011.2|mainq|NONE|ENTERQUEUE||00110102102 1123325020|1123325011.2|mainq|Agent/21|CONNECT|5

Things like this keep showing up in my my logs. Any idea what to look for to (1) figure out why and (2) track it to a particular service? systemd: Failed to mark scope session-19.scope as abandoned : Stale file handle -- -- Steve

Howdy. I''m wondering what happens in Mongrel/Ruby when a user asks for a page but then hits Stop in the browser or just clicks a different link before the response comes back. The reason I ask is that the app I''m building does a fair amount of database work on each page. From what I can tell, when a user abandons a request, the server has no awareness of this and thus

Dear All, I have a query ,basically i use three server for own call center. The server A and B i have configure the 60-60 channel each server. Server A and B(or call transfers into server X) calls hitting into server X.Both the server have contain same CLI mean anybody call 8032(mean server A an B) call goes to Server X. In the case of Server A 8032 mapped with toll-free,it is configured with

FROM NT \\firewall.surson runs perfectly WHY??? root@firewall init.d]# ./smb status smbd (pid 12183) est? rodando... nmbd (pid 1397) est? rodando... [root@firewall init.d]# smbclient //firewall.surson/profile -Ucatena%motpock added interface ip=192.168.1.1 bcast=192.168.1.255 nmask=255.255.255.0 Domain=[SURSON] OS=[Unix] Server=[Samba 2.2.3a] tree connect failed: NT_STATUS_BAD_NETWORK_NAME

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hi Guys, I am not so familiar with asterisk and hope to get help here. I am having now some stupid errors. My goal for the first, is to create a simple pbx with different context. As long as I use only the contex 'default' everything seems to work perfect. Now I tried to add another context i.e 'internal' and the asterisk is complaining for not finding the required extension in

Dear all I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan I have setup asterisk with avaya system i have 5 avaya system on 5 location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when

Recently discovered Markaby. Before I use it on a production system, I''d like to know if it is still being maintained? According to the change log, the last change was in February of 2006. Does this mean that the project has been abandoned? Or, at version 0.3 it was considered production stable and complete? -- Best Regards, -Larry "Work, work, work...there is no satisfactory

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system.

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download