search for: sdnglobal

Displaying 6 results from an estimated 6 matches for "sdnglobal".

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2004 Sep 30
1
Queue Setup almost got it
...ailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup extensions.conf: ; ; Standard Tech Support Calls exten => 1590,1,Answer exten => 1590,2,Wait(1) exten => 1590,3,SetVar(QUEUE_PRIO=0) exten => 1590,4,Queue(nocc|t||) exten => 1590,5,Playback(sdnglobal/no1avail-leavevm) exten => 1590,6,Voicemail(s1500) exten => 1590,7,Wait(1) exten => 1590,8,Hangup ; Network Down Emergency Queue exten => 1501,1,Answer exten => 1501,2,Wait(1) exten => 1501,3,SetVar(QUEUE_PRIO=50) exten => 1501,4,Queue(nocc|t||) exten => 1501,5,Playback(...
2004 Sep 30
2
Queue Setup
Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was
2003 Oct 30
2
critical problem
About every 10th call coming into my x1000p is not getting the audio it should. You can see the messages scrolling on the console as they usually would, playing the thankyou, then and menu messages. internal phones ring, but when answered there is no audio. The caller gets a full volume echo with about 1/2 second latency. At first I thought it might be related to using the aggressive
2007 Mar 20
1
SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
...and I see repeated INVITEs being sent without any acks. OPTIONs are being sent and acked. The remote SIP phone is 'eden-1000a' and the voicemail extension is 9990. *This worked just fine up until the upgrade.* Does this ring a bell with anyone out there??? Tim McKee <tmckee at sdnglobal dot com> SDN Global ============================================== pbx*CLI> sip debug peer eden-1000a SIP Debugging Enabled for IP: 10.253.4.50:5060 pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@hostname.company.domain;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.253.4.50;bra...
2007 Apr 24
1
E&M Wink start problem
Attempting to talk to an Eagle Telephonics switch at a disaster exercise. Didn't think a plain old E&M wink start T1 would be this much of an issue. We finally got the Eagle to accept a call from *, but whilst I can hear the person on the Eagle, they can't hear me. When they initiate a dial out I only get the first 2 digits from their switch... Does anyone have decent
2010 Mar 23
0
Strange Meetme disconnects
Running * version 1.6.1.17. My meetme conferences automagically disconnect users approximately 5-15 seconds after the user is connected. This occurs regardless of whether music on hold is active or not. [Mar 23 11:34:36] -- Executing Macro("SIP/SDN_TMCKEE-000000e9", "confroom,1808") [Mar 23 11:34:36] -- Executing [s at macro-confroom:1]