similar to: FW: DID to hunt group?

Asterisk 1.4.2 I have spent much of today trying to make a DID (from SIP GW) ring to 4 extensions in a hunt (roll-over) group. Results from searching docs and forums seem to indicate it is doable and so trivial no one includes an actual example. I can make all 4 exts ring at once with the like of exten => _1655,1,Ringing() exten =>

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

Can someone get this guy off the lists? ---------- Forwarded message ---------- From: postmaster@prebit.net <postmaster@prebit.net> Date: Nov 1, 2006 3:22 PM Subject: Benachrichtung zum =?unicode-1-1-utf-7?Q?+ANw-bermittlungsstatus (Fehlgeschlagen)?= To: joakimsen@gmail.com Dies ist eine automatisch erstellte Benachrichtigung +APw-ber den Zustellstatus. +ANw-bermittlung an folgende

Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and

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We are the marketing specialists www.host4bulk.com that provide cheap bullet proof bulk email friendly hosting for your website ($400 for one month of bullet proof hosting) and cheap bulk email campaigns ($200 for 1 million emails sent) As you may already know, many web hosting companies have Terms of Service (TOS) or Acceptable Use Policies (AUP) against the delivery of emails advertising or

Hi there, I'm urgently searching for a bulletproof NON-X mp3 player for high bit rates that can be remotely controlled - if possible. (Server-Client?!) Right now I'm using mpg123 and this program not always starts flawlessly with streams. All ideas are appreciated. HERB RadioOnDemand.com --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage:

> > I think there are too many packages that would need changes under this > scheme. There would be zero if the registration of options is not required for packages first uploaded on CRAN before the feature is implemented. If an option is not registered no validation is triggered and nothing breaks even if we opt in the behavior. > If those functions could be made simple enough and

Hi there, I'm tring a basic asterisk settings. I have a asterisk 1.2.7.1 running on a I have a net with two computers and a router. The router IP in the local net is 192.168.1.1, The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux. the second pc has IP: 192.168.1.4 name fissun . SO GNU Linux. On datile3, it runs a softphone kphone. From this I want to call the external world. on

Hello, I get some problem installing asterisk + ekiga on my debian etch: ii asterisk 1.2.13~dfsg-2 Open Source Private Branch Exchange (PBX) ii ekiga 2.0.3-4 H.323 and SIP compatible VOIP client $: asterisk -U asterisk -vgc give me some WARNING like : ,---- | WARNING[21806]: res_musiconhold.c:852 moh_register: Unable to open | pseudo channel for timing... Sound

Anyone else on the list using these tools? I grabbed the latest iteration using yum from Dag's repository, but I get random segfaults on a fully-patched 4.4 system on dual Opteron 275's from some of the utilities. Perhaps, it's some 64-bit vs 32-bit-ism? I'm not much of a programmer (actually, I'm a terrible programmer) so any suggestions would be very much appreciated.

On 3/4/2015 12:45 PM, Dave McGuire wrote: > There is. But I already have a firewall, running on bulletproof > hardware that doesn't depend on spinning disks. I don't want to add > ANOTHER firewall when I already have a perfectly good one. Besides, my > mail server is built for...serving mail. Not being a firewall. You can implement whatever type of security you are

The following gotcha caught me off-guard just now. I have two matrices, a and b: a <- matrix(1,3,3) b <- matrix(1,1,1) (note that both "a" and "b" are matrices). I want them in a list: > B <- NULL > B[[1]] <- a > B[[2]] <- b > B [[1]] [,1] [,2] [,3] [1,] 1 1 1 [2,] 1 1 1 [3,] 1 1 1 [[2]] [,1] [1,] 1

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

On Fri, 2018-03-23 at 10:44 -0400, lingpanda101 via samba wrote: > > Andrew in another topic discussed the issue. See below. > > "This is the failed re-index thing we had brought up by a number of > other posters late last week. I'll be working on that today. > > Run the sambaundoguidindex script and stay with 4.7 for now until I can > fix it. A patch for