similar to: Agents/Queues - Drops call after 60 seconds

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Here is the full "transaction" -- outgoing agentcall, to agent '1001', on 'Local/3044@local-4fee,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-4fee,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-4fee,2", "SIP/3044|20|t") in new stack -- Called 3044 -- SIP/3044-ea92 is

I am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/3044@local-fd6d,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-fd6d,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-fd6d,2", "SIP/3044|20|t") in new stack -- Called 3044 --

Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in and awaiting a call via AgentCallback. Agent 2 has not logged in. An outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's phone rings and Agent1 and A talk. While they are talking, caller B calls in. Caller B is correctly placed in the queue and hears music, however this shows up in asterisk

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

The setup: Inc SIP Call -> Asterisk 1 -- IAX --> Asterisk 2 -- SIP --> phone (3044) Asterisk 1 shows the following: (1.0.3) -- Executing Goto("SIP/XX.XX.XX.XX-0819f590", "cytel-internal|3044|1") in new stack -- Goto (cytel-internal,3044,1) -- Executing Dial("SIP/XX.XX.XX.XX-0819f590",

I'm not a PRI expert and therefore don't know what this debug stuff means for PRI, so if anyone can help me here... I'm running the latest libpri and zaptel from CVS. Keep in mind that everything works fine when using the STABLE libpri and zaptel. I am NOT running CVS asterisk. I am running 1.0.6. asterisk*CLI> pri intense debug span 1 Enabled EXTENSIVE debugging on span 1 T203

I'm guessing this is a problem with the phone itself. We all have Cisco 7960s with 7.3 firmware. I can yell down the hall at 3091 and say "are you on the phone?" and she yells back "no". But when I try and call her extension, I get this: -- Executing Dial("SIP/3044-8eb6", "SIP/3091|15") in new stack -- Called 3091 -- Got SIP response 486 "Busy

Well, it is clear enough that the problem is in interpreting the documentation. However, when you claim you tested something, and found it inconsistent with tables, it would be advisable to back it up with examples! The description in the help files and in the sources is admittedly confusing. The original paper has this, rather more clear, description in the abstract: "We consider the

hi, im looking at deploying asterisk in a small corporate enviroment which will have approx. 1200 IP Phones and an average of about 100 to 200 calls at any given time. The calls will be sent out SIP to my Cisco Gateway. Im running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a calculator or a spreadsheet which could tell me about how many calls I will be able to make through *

Hello R help members. I came across today with an article on Barnard's exact test (http://www.cytel.com/Papers/twobinomials.pdf), that is supposed to give a more powerful fisher.test - Because it doesn't assume that we know the row and column totals are in advance. Any pointers to such a function ? Thanks, Tal -- ---------------------------------------------- My contact information:

Hi guys, I'm testing the sending of a URL to an XLite softphone when a call is in queue. See the output of the CLI below: -- Executing Queue("Zap/69-1", "q_sample|tT|http:// www.google.com/") in new stack -- Started music on hold, class 'default', on Zap/69-1 -- outgoing agentcall, to agent '1000', on 'Local/ 1000@agents-1b94,1'

Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name "Maximum talktime in a queue?" One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten =>

Dear R folks, We completed a matched case-control study that was analyzed using conditional logistic regression. Because of the small sample size we need to calculate exact confidence intervals. The quick solution is to purchase LogExact by Cytel. However, we'd like to do this in R. Anyone have experience with this? Many thanks, Tomas ____________________________________ Tomas Aragon, MD,

Another example: Jerry Friedman's MART is available in R from Salford for the same price as the stand-alone TreeNet, even though they don't advertise it on their web site. Andy > -----Original Message----- > From: rossini at blindglobe.net [mailto:rossini at blindglobe.net] > Sent: Sunday, August 17, 2003 9:50 PM > To: rhelp > Cc: pralay at cytel.com > Subject: [R]

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

At JSM, I spent a bit of time with old friends at the Cytel booth (makers of StatXact/LogXact). They were wondering whether it was both feasible and of interest to create a package of the StatXact compute engine for R (to be commercially licensed, not for free!), similar to what they've done for SAS. As far as I know, it's feasible, (this is not the first commercial external package,

Why are there so many attempts to native bridge? The call is actually up and working by attempt #1 so what is it doing on all those other attempts? We are only allowing G729 so it can't be codec negotiation. -Matthew -- Executing Goto("SIP/3013-3dfa", "cytel-outgoing|917034439032|1") in new stack -- Goto (cytel-outgoing,917034439032,1) -- Executing