similar to: Newbie can't dial out to pstn

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

I've put my fire proof suit on ready to ask a question ;o) Redhat 9 Kernel 2.4.20-8 AVM Fritz PCI Problem 1... i have followed instructions from:- http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI (Plus many many other sites found via google over the last 16 hours) after modifying the makefile & defs.h,

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

Hi all, <p>once again i came up with my conferencing stuff. On a conference with more then two people it's a waste of bandwidth, that every entity send it's data to every entity. Since there is only one audio line, the audio must be merged on the server. Here are my questions: - How many audio chanels can a server process (let's say a 3GHz machine) in this way: decode all

Hello: I would like to know if there's a way to request free chanels from remote asterisk servers ? My idea is to make an agi returning a dial to inter-asterisk connected servers when there's not enought chanels on local server, maybe like a ping to all of them or maybe requesting to a central server where all the *s send and request information about available chanels each 2 or 3

Hi all, I want to check if this is a bug for which I should file a report. I am using R2.5.0 on OS X 10.4.9. When I invoke the data editor and when I change the values of individual cells, it seems to work as intended. However, when I try to delete/add a row/column, R.app crashes. I've attached the crash log. Best, -Nathan -------------- next part -------------- An embedded and

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

Hi I am using inAccess channel driver. Compiled, installed. This is what I get when I am trying to start * --------------------------------------- [chan_oh323.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource): libh323_linux_x86_r.so.1.12: cannot open shared object file: No such file or directory WARNING[16384]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so

I installed TDM400P and X100P pci cards in a system running mandrake 10.1 official, kernel 2.6.8.1-12mdksmp. I can compile zaptel, libpri, asterisk and modprobe (zaptel, wcfxs, wcfxo) without errors. Except that running ztcfg and asterisk fails. [root@asterisk asterisk]# ztcfg Notice: Configuration file is /etc/zaptel.conf line 3: Unable to open master device '/dev/zap/ctl'

On Thu, Jun 19, 2008 at 12:32 AM, Duncan Sands <baldrick at free.fr> wrote: > Hi, > > On Wednesday 18 June 2008 15:08:46 Edd Barrett wrote: >> Holger Schurig wrote: >> >> With 3.3.5 my first test took 5 times to produce a non "bus >> >> error" build. There were no 'make cleans' in between. >> >> >> >> What is

All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: > From: Ben Witso <benw@bgwcomp.com> > Date: Mon Jun 21, 2004 7:28:42 PM US/Central > To: Asterisk-Users

Hi, Please let me know if you have any ideas - I am taking wild guesses now.... Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP

hey McFlow, this theme actually important for me now )) Me and my friend have the same problem in Command & Conquer 3 Kane's Wrath if you still there (or somebody else) please, tell us how did you fix it??? :( I have XP on my PC, my friend using Linux Gentoo. we make a chanel in hamachi, enter it. We pinging each other good but in the game I don't see him... and he sees me and my

Version: 2.2.9 & 2.2.15 ( http://ppa.launchpad.net/malte.swart/dovecot-2.2/ubuntu ) OS: Ubuntu 14.04 Server LTS I installed Ubuntu 14.04.2 LTS newly on my new server... And I try to configure the mail server considering both my system account and the virtual mailboxes, by seeing https://www.exratione.com/2014/05/a-mailserver-on-ubuntu-1404-postfix-dovecot-mysql/ But when I configured by

hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. ----- if i use analogue phones in the above case ( we have analogue phones already )

Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the

Core Dump(?) from /var/log/apport.log on Ubuntu Server 14.04 LTS... ---------------------------------- ERROR: apport (pid 29249) Mon Mar 9 01:26:31 2015: called for pid 29239, signal 11, core limit 0 ERROR: apport (pid 29249) Mon Mar 9 01:26:31 2015: executable: /usr/lib/dovecot/auth (command line "dovecot/auth -w") ERROR: apport (pid 29249) Mon Mar 9 01:26:31 2015: Unhandled

Hi, On Wednesday 18 June 2008 15:08:46 Edd Barrett wrote: > Holger Schurig wrote: > >> With 3.3.5 my first test took 5 times to produce a non "bus > >> error" build. There were no 'make cleans' in between. > >> > >> What is going on? > > > > You mean you used your bsd-ports-provided gcc to compile LLVM and > > you've