similar to: pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use

I have two Asterisk servers running on the same LAN. One starts fine, but when I start the other I get: pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use and Asterisk does not start. OK I thought, I'll just change the port in dundi.conf. I changed it to 4521 and indeed it started just fine, until I reboot that is. WHen I reboot, I get the same

Hi, Reviewing the archives I saw /2004-October/070314.html from Tim Lewis. His error is almost identical to mine i.e. when "make clean; make install" in asterisk sub dir, I get the following: pbx_dundi.c:54:18: zlib.h: No such file or directory pbx_dundi.c: In function `update_key': pbx_dundi.c:1315: warning: implicit declaration of function `crc32' pbx_dundi.c: In function

Anyone seen this...? Is it a known issue? I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't against the latest code I get given crap for it. Given that most of the time you don't know HOW to reproduce a problem on the latest code anyway, not accepting bugs from older versions does the community no service, because potential bugs are never accepted for

hello All, what could be the problem in dundi lookup *pbx_dundi.c:4109 dundi_result_read: Result number 1 is not valid for DUNDi query results for ID 879!* though it should return some results , it failed in getting those . foloowing is my DIALPLAN exten => s,n,Set(ID=${DUNDIQUERY(${NUMBER},priv,b)}) exten => s,n,NoOp(DUNDI-QUERY-ID [ ${ID} ]) exten =>

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

Hi, >is Bind9 running ? Yes netstat -lntup | grep 53 tcp 0 0 127.0.0.1:953 0.0.0.0:* OU?A 13296/named tcp 0 0 0.0.0.0:49153 0.0.0.0:* OU?A 15105/samba: task[d tcp6 0 0 :::49153 :::* OU?A 15105/samba: task[d /etc/init.d/bind9 status ? bind9.service - BIND Domain

Hi, I have solved. I change parameter "listen-on port 53", as follows: From: listen-on port 53 { 192.168.1.20; 127.0.01; }; To: listen-on port 53 { any; }; netstat -lntup | grep 53 tcp 0 0 0.0.0.0:81 0.0.0.0:* OU?A 534/lighttpd tcp 0 0 192.168.1.20:53 0.0.0.0:* OU?A 1930/named tcp 0 0

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get

Hi, I'm having trouble compiling zaptel 1.4.4 on SUSE 10.1. I'm really only interested in getting ztdummy to work because this is a dev machine with no zaptel h/w. SUSE 10.1 is a 2.6 kernel: asterisk-dev:/home/hugh # uname -r 2.6.16.13-4-default It seems that my problem is related to autoconf.h - I cannot find that file: asterisk-dev:/home/hugh # find / -name 'autoconf.h'

I'm trying to configure Chan_H323 to register with GnuGK... without success... i've failed finding sample configurations. I'd greatly appreciate anyone who can provide sample config of H323.conf and gnugk.ini I am tyring to configure Asterisk as a neighbor in GnuGK. I'm always getting this error on Asterisk. ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration

I have an Asterisk serving 15 people using the X-Lite soft-phone. Currently they all register to the internal IP address of Asterisk (192.168.1.110). I only use VoIP internally. External calls go PSTN. I'd like to arrange it so that they register to our external WAN address (port forwarded to Asterisk) so that they can go mobile and still have Asterisk service. Is it possible to arrange it

Am 26.12.2014 um 02:20 schrieb Edwardo Garcia: > On 12/26/14, Jeff Mitchell <jeffrey.mitchell at gmail.com> wrote: >> On Dec 25, 2014 3:15 PM, "Reindl Harald" <h.reindl at thelounge.net> wrote: >>> >>> your Gentoo is nice in a small environment >>> >>> on larger setups someone is using binary packages and can setup his own >>

Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this

https://bugzilla.samba.org/show_bug.cgi?id=4520 Summary: Add the ability to specify a password on the command line Product: rsync Version: 2.6.8 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned@samba.org

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype ? much

I'm getting the following error over and over on the console: pbx_dundi.c:2975 dundi_rexmit: Max retries exceeded to host Any idea how to troubleshoot this? My network latency is roughly 40-50ms between all hosts in my dundi cloud. Jeremy Mann Director of IT Texas Health Management Group Direct Line: 817-310-4956 Main Line: 817-310-4999 Fax: 817-310-4990 Email: jmann at txhmg.com

Quick question re. NAT traversal. I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? A web brower behind a NAT sends an

Curious whether the ChanIsAvail command is thread safe. By that I mean, if I use ChanIsAvail to determine which channel to use, can I be sure that it will still be available when I go to Dial it on the next line? It occurs to me that there's a possibility the channel could get used by a competing thread AFTER my thread has determined it is available and BEFORE my thread gets a chance to

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How