similar to: zt_rbs errors!?! never seen before.

All of the stuff I've googled for and read on wiki all relate to "Outlook". Has anyone been successful in getting "Outlook Express" to do click to dial? -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the

Anyone used the combination above? We are and it sounds like crap. The audio drops out in regular intervals which suggests that someone's g729 isn't doing its job correctly. I'm blameing XTen cause when I make a ulaw call that gets converted to 729 using digium's 729, calls sound fine. Anyone else? Similar experiences? -Matthew --

I posted this to the -dev list and have recieved no reponse so I'm gonna try the -users list. I'm trying to bring ast-ax-snmp up to date with current CVS. It's a small patch against chan_zap and pbx, but also has a loadable module component. That module is compiled against some NET-SNMPD headers/libraries. But net-snmpd also has module support and they have a function called

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

We have asterisk running on a quad processor dell. The kernel has been compiled with SMP. However, asterisk seems to only use 1 processor. 3 of the 4 always stay at 100% idle. Is it pointless to have a multi-proc machine? I was going to buy a new dual 3.6Ghz Xeon server but if nothing will take advantage of the other proc... Perhaps my conception of multi-proc/threaded is warped. If asterisk is

Before you jump ahead, yes I do have chan_zap.so loaded.. Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI -- Accepting AUTHENTICATED call from 22.22.22.22: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing

Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 "wanpipe1 card 0" span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 "wanpipe2 card 1" span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI> == Using SIP RTP

Hi All, I am from the packaging team in RBS. We use R DCOMServer Manager 2.0, this application uses RGUI 2.3.0 as one of its components. We are planning to upgrade to the latest version of RGUI 2.8.0 but found that R DCOMServer manager is not compatible with the latest version. Can you direct me from where can I download the compatible version of DCON server Manager? Thanks, Smita Kumar RBS

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status:

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

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Hi, I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four port T1 card. Only one RBS T1 plugged into it right now. I have been getting complaints about random hangups. Endpoints are all remote, but I track very closely the latency (by graphing the output of "sip show peers") which normally shows me when a peer is having connectivity issues. Several that I have

I am trying to read a csv file using the code; contol <- read.csv("RBS.csv") This is the error message I got; Error in file(file, "r") : unable to open connection In addition: Warning message: In file(file, "r") : cannot open file 'RBS.csv', reason 'No such file or directory' Where was the mistake? -- OYEYEMI, Gafar Matanmi (Ph.D) Senior

I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, "the other Mark" in a truly Digium moment in Rostock, Germany on their way to listen to the sea shanties. http://tr.im/rawhide - be afraid, be very afraid (Adhearsions' Jason Goecke is also in the picture somewhere) /r

One of my user is using asterisk 1.4 based Dialer i.e Vici 2.0. Today, when they downloaded , the CDR from the carrier site for 26th June 2010 , they see 50% calls are NEVER dialed by Dialer but it appears in CDR. Amazingly, all the call durations are of 29-30 secs. When we checked the status of the same in Dialer, lead is present there but its marked as NEW which means Dialer has ever dialed

Hello, Now that we've had our new Digium TE406P card in production for 4 days we have discovered audio drop problems that happen randomly across all channels. Here's more about our setup: P4-3.2GHz 2GB ram Slackware Linux 10.1 with custom kernel 2.4.29 Asterisk 1.2beta1 Digium TE406P quad T1 card with the following attached: - 2 x RBS D4/AMI 24 channel T1s - 1 x RBS B8ZS/ESF 24 channel

Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a HOWTO somewhere? Wiki has nothing I could find. I've got plently of public IPs I can assign to it but don't know how. Thanks, Matthew -----

Hi, I have a client that has a 24 channel voice T1 that I have been using e&m signalling over for a number of years. The local telco finally got an ISDN switch and wants to move them to PRI. I didn't see this as a big problem - I've done a few others on this particular Caribbean island without issues, but this would be the first time with a Xorcom unit involved. We tried to do

Hello, I have a question on sp package. I drew pixel heat maps of continental U.S. using insurance litigation data. If an insurance claim is litigated at a certain location, 1 is assigned; otherwise 0 is assigned. I have assigned a value between 0 and 1 to each pixel on my map using inverse-distance weighting (spatial smoothing). I need to draw two pixel maps, one for period 2001-2006 and the