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Asterisk 1.8.3.2 I have been getting this warning constantly on CLI in a call scenario where I use local channels to connect SIP with PSTN. I use callfile and local channel to first call a PSTN number and if answered, use local channel to call SIP phone with music on hold enabled in Dial string. If I call PSTN from SIP directly or vice versa I don't see this warning coming. On SIP I have

I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained

Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Hi group, I have a simple call center scenario set up on Asterisk. Customer calls the DID and gets placed in Queue waiting for their turn to talk to the available agent. Sometimes Customer hangs up in between and in this case I want to get the last position of customer in Queue. I know there is a variable called ${QEORIGINALPOS} that gives us original position of caller in Queue, but there

Hi Group, Is there any information available for Asterisk 1.8 dimensioning? I googled but couldn't find helpful data for 1.8. I am trying to figure out hardware configuration for following features implemented in Asterisk 1.8? (1)100 SIP clients. (2)ACD (Around 15 realtime queues) (3)Call recording for all SIP clients. (4)4 port PRI (E1). There would be around 100 concurrent calls.

Hi, Can we know the estimated wait time for a caller before sending him in a Queue? Asterisk 1.8 Thanks, --AM -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110412/4e5eae7d/attachment.htm>

Hi, wrandom strategy for Queue says...rings random interface, but uses the member's penalty as a weight when calculating their metric. So a member with penalty 0 will have a metric somewhere between 0 and 1000, and a member with penalty 1 will have a metric between 0 and 2000, and a member with penalty 2 will have a metric between 0 and 3000. Please note, if using this strategy, the member

Hi, Has anybody any idea about detecting Special Information Tone(SIT) when making utbound calls? http://en.wikipedia.org/wiki/Special_information_tone I googled for detecting SIT in Asterisk but couldn't find useful results. Thanks, --Sam