similar to: QueueCallerAbandon is not triggering after 1.8.3.3...

Alex, Thank you so much for your response. I've been so consumed with other business that I only just now getting back to this issue. We have implemented your suggestion which is perfect. Thank you again. I've never asked a question of the community before and I'm extremely happy with the rapid response I received. Somewhat related to this initial problem I have an additional

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA

Hey all, I'm in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I'm writing this in C# but language isn't important. I'm not looking for a hand out on code, what I'm really interested in is

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and

Hey all! I'm trying to figure out a way to manipulate a call's caller ID based off of the caller's caller ID. Basically, I've got a situation where anything that goes through an Nortel Opt11's IVR comes out with the caller ID 400 (the Opt11's IVR's ext). When the call goes out the trunk that the call is destined for, I'd like to grab the 400 caller ID and

Hello. I'd like to know if Samba can be installed on Solaris. If so, what are the steps to follow? If my question is not well defined please let me know. Thank you in advance for your precious time, Juan Diego Guti?rrez Gallardo jgutierrez@teleserver.es

I've been trying to compile 2.9.2 under solaris 10 the last couple days without success. configure runs fine and I'm using GNU Make. We're trying to build under 64 bit (and I'm wondering if this might be part of our problem). Here is the output from the make... ld: warning: file /usr/local/lib/libiconv.so: attempted multiple inclusion of file Undefined

Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

Hi All, I have a query about time based sequences. I know such questions have been asked a lot on forums, but I couldnt find the exact thing that I was looking for. I want to create a time-based sequence which will mimic the trading window AND would span multiple days. Something like below: "2011-01-03 09:15:00 IST" "2011-01-03 09:15:01 IST" .... .... .... "2011-01-03

Hi everybody! I have asterisk 1.8.3.3 with pickup deadlock avoidance patch applied in production. I'm use next dialplan construction exten => *,1,PickUP(queue-number) exten => *,2,PickUP() Can anyone tell me how can I stop dialplan execution on the first priority if that priority's pickup was successfull? I tried *i* priority, but no luck. Does pickup app returns any value in

Hi List, I'm running into trouble, if I perform a 'yum update' on my AsteriskNOW. Currently I'm running Asterisk 1.8.3.3. I have the following problem, if I do the update to the actual 1.8.4.2. There are several commands on the CLI which are not working or even not present like core show uptime (not working) core restart (not present) core show version (not present) my Skype

Hello, We see some strange behavior with phone calls, we use Asterisk 1.8.3.3. SIP clients (all behind NAT at different locations, so not a single NAT solution is used): - x-lite - linksys pap2t - polycom kirk (multiple type numbers) - polycom (multiple type numbers, hardware phones) Our Asterisk servers stays in between (some calls are recorded). Asterisk is running on a physical server (no

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/c3bb9124/attachment.htm>

Folks, My controller is being given the session id as part of the POST payload. Is there a way to manually trigger the session restore process once I extract the session id? Can someone point me to the files where this is happening? Thanks, Joel -- http://wagerlabs.com/