similar to: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)

Nik, I'm not sure that "NOP" is correct, but I'm in the states so I'll to defer to someone who knows E1/PRI. When I run zttool I have "OK" under the alarms. Is there a way you can call the telco and confirm the settings? Make sure that framing, coding and D channels are set up on their end the same way you're set up. As for asterisk, here's what I get

hi i've configured a TE205P on asterisk at home this is my zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow bchan = 1-15, 17-31 dchan = 16 bchan = 32-46,48-62 dchan = 47 loadzone = it defaultzone = it and my zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes

Nik, Looks like you're making some progress. When I first started using A@H I had trouble getting the outbound dialing to work. I wasn't sure where to start, so what I did was skip the macros in the dial plan. I wanted to play around with exactly what digits the telco wanted to see. So I put a specific extension in my [default] context like this: exten =>

Nik, Just curious - what is your telco setup? Do you have PRI with the specified D channels? You need to make sure that your telco is set up to have the D channels on 16 and 47. When you first start Asterisk, or when you log on to the CLI, do you ever see messages stating the B channels are successfully started? Let us know. -MC -----Original Message----- From:

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put

Hei! I'm new to Linux and Wine. I'm using Linux Fedora 7 and Wine 0.9.58. On my computer there's another operating system ? Windows XP. Now I'm trying to install Microsoft Visual Studio 2005 Professional on Fedora. And I can't do it. When the installation process begins there's an error message - ?A trial edition is already installed on this computer. You must uninstall the

Hi All, First time caller - tried WINE today for the very first time, on OpenSolaris no less. Good news - was shocked and stunned how easy it was to get an old copy of Unreal up and running - everything (almost...) "just worked" with practically no configuration, got lovely smooth rendering using my crappy built-in Intel X3100 adaptor. However, I cannot get any audio output

Hi all, Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal: 1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk application associated with it. 2. Receiving a fax at asterisk works fine, at least appears to be working

Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas

dear rsyncists, rsync is superb! but did i actually find a bug? i did: $ rsync -vRHSPa rsync://rsync.mirror.ac.uk/updates.redhat.com/7.3/en/os/SRPMS . into an empty local directory the result was 140 bogus hardlinks of the same file: -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24 Canna-3.5b2-62.7.3.src.rpm -rw-rw-r-- 140 g g 6147582 2002/12/04 16:55:24

Brian Willoughby wrote: > Hi Justin, > > Are you saying that 24-bit support does not work for you? Which > version of FLAC are you using? What platform? What kind of files are > you trying to compress? > We're using FLAC 1.1.2, and we're compressing any kinds of PCM audio. The software that we make that uses it is REAPER, www.reaper.fm. We sent some messages

Hi Forum, I am new and learning this research tool. was trying to run some tests on a censored dataset using the NADA library. Everything is hunky dory except 'censummary' throws an error and I am trying to figure out a reason for this. Please help! below you will see the error statement... thnx --nk > library(NADA) Loading required package: survival Loading required package: splines

Hola Dale, Bumping it up to 3 bits makes everything work hunky dory on the PC side of my project. Would you like me to commit that? Also, after speaking with some VStudio folks, it looks like that issue will likely remain for VS2k8 as well. Another trick that was suggested to me was to try something along these lines: Enum E : unsigned int { A, B, C, D }; ... Class ... { E e : 2; Unsigned

Casper.Dik at Sun.COM wrote: > > I would suggest that you follow my recipe: not check the boot-archive > during a reboot. And then report back. (I''m assuming that that will take > several weeks) > We are back at square one; or, at the subject line. I did a zpool status -v, everything was hunky dory. Next, a power failure, 2 hours later, and this is what zpool status

Nik, This definitely helps! Please check your dial command. You've got "Dial(Zap/0/mynumber)" and I think you might possibly want it to be something like this: Dial(Zap/1/mynumber) or Dial(Zap/g0/mynumber) I don't recall there being a zap channel zero, but it is common to have a group zero. I would recommend trying Zap channel 1 - Dial(Zap/1/mynumber) - before trying the

Hi, I'm confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

hey, I upgraded a server to 5.3 from 5.2 today, its a mail server, and its using the rpmforge clamav milter... was getting errors in my mail log like... Apr 6 01:39:32 freescruz sendmail[12304]: n368dWxW012304: Milter (clamav): error connecting to filter: Connection refused by /var/clamav/clmilter.socket Apr 6 01:39:32 freescruz sendmail[12304]: n368dWxW012304: Milter (clamav): to error

Dale Johannesen wrote:- > You might argue with that one if you didn't feel like changing your > compiler, but 9 .6p4 is quite explicit: > > ... If the value of an enumerator is stored into a bitfield > of the same enumeration type and the number of bits in the bitfield > is large enough to hold all the values of that enumeration type, the > original enumerator value and

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>