similar to: res_odbc and configuration files

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

So I have a route with [1-9][0-9][0-9][1-9][0-9]* as a base route that should match NXXNX. Right? I built another route 01144[0-9]* that I thought would match 01144X. and send the call to the UK but the script is matching 01144207108???? With the first route. Can someone smarter than me help with some samples? Please? If I can get one for 1NXXN. and 01144. I should be able to figure the rest

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

-----Original Message----- > -this is very true, however, the current version of the Axxess software > (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess > upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider.

Sorry everyone, I know this has been hashed over a bunch of times but I can't find anything that pertains to specific cracking and popping on the FXO modules of a TDM04. This happens on inbound or outbound calls. This is the first install I have done with a TDM card for FXO modules so please, be kind if I am missing something really simple. Damn I wish everyone wanted t-1's or

I have been playing with res_odbc in these last days, but it doesn't want to work. This is the output when starting Asterisk, so everything seems OK: [res_odbc.so] => (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jul 7 20:11:32 NOTICE[-1084915040]: res_odbc.c:132 load_odbc_config: registered database handle 'mysql' dsn->[MySQL-asterisk] Jul 7

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

I have 4 7960's that I am trying to get working but 2 of them will not update to the SIP image on my tftp server like the first ones did. i keep getting the error on the phone 'Defaulting CM to TFTP server' like it isn't seeing the *.bin on the server. are you supposed to have on of those for each phone? would be like cisco et al to do something like that. TIA Jason Kawakami

Hi I'm using asterisk 1.0.3_1 from the freebsd ports. Everything works ok, except I can's get res_config to populate sip.conf (or any other conf file for that matter). my unixodbc works and I can access the database with the isql tool. There are no warnings or anything else in the asterisk log file indicating a problem. I have used the guide from

Hi there, admittedly this is a slightly aged RH system. ;-> Yet what I'd like to know is if this compile problem is clearly due to the gcc version, or if there is a way to solve the issue...? Cheers, Philipp [root@box src]# gcc -v Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs gcc version 2.96 20000731 (Red Hat Linux 7.2 2.96-112.7.2) [root@box src]# rpm -q unixODBC

Good morning, I'm having suddenly cut-offs and I don`t know why. It's been hapenning since I enabled cdr_odbc/func_odbc in my system. We use func_odbc to register some queue member's events (login, pause, etc.) at an external DB ('remoto' connector) and to uptade local tables at a local DB ('local' connector). Currently we are usind cdr_odbc to Postgresql and cdr_addon

I want to let Asterisk read its configuration from a mysql database. I configured everything according to the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+res_config. However it doesn't work. I am using 1.0.8 asterisk version and here are my config files: Extconfig.conf: [settings] ;uncomment to load queues.conf via the db engine. ;queues.conf =>

I have a mixed PBX system with both Asterisk 1.4.21 and 1.2.27 (moving to 1.2.28). For now I need to keep a few boxes in 1.2 and not migrate them all to 1.4. However, I would like to have func_odbc and res_odbc on all servers. On 1.4.21, native func_odbc seems to work fine. On 1.2.27, the func_odbc backport is giving me an error (I know that this backport is not "officially supported"

Dear list, i have a issue Asterisk crash (Module res_odbc exactly) after the same log who is "/ERROR[23805] astobj2.c: bad magic number.../" you will see on the log : Today [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958* [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09

I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to

Nope, I got the stable 1.08 release from cvs. Frank From: "Wei Kun" <wei-kun@mediaring.com> Subject: RE: [Asterisk-Users] Asterisk configuration from database withres_config To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <00a301c5a3c5$93384370$6f01010a@corp.mediaring.com> Content-Type:

----- Original Message ----- > From: Greg Hill <gregh-asterisk@hillnet.us> > Subject: Re: [Asterisk-Users] New to * > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <Pine.LNX.4.44.0409031231070.1975-100000@hillnet.us> > Content-Type: TEXT/PLAIN; charset=US-ASCII > > On Fri, 3 Sep 2004, Bill

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a

Back in the office post-astricon. 1.0.0 running in the lab. YIIIIIIIHAAAAAAAAAA! THIS GUY!!!! rocks. Thanks to Mark for *, Steve and Olle for the conference and to ALL community members. Everyone using * is contributing in one way or another. See y'all next year Jason Kawakami www.optellabs.com