similar to: Asterisk not starting up due to database problems

Hello I'm using a standard Asterisk install with default settings, and when I run "reload", I see that Asterisk fetches configuration information from a lot more sources than just my extensions.conf and sip.conf. For instance: -- Registered indication country 've' -- Registered indication country 'za' -- Setting default indication country to

Sorry, I forget to tell I tried, but not works. *Context:* context ivr_temp2 { s => { Proceeding(); str_time_01 = '06:00-12:00|*|*|*'; // Manh? ifTime (${str_time_01}) { Playback(ura/bom_dia); } } } The error is showed on "ael reload". *Console errors:* rs0000sr304*CLI> ael reload Command 'ael reload' failed.

Hello, I recently completed a fresh install of Asterisk SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the voicemail application module to load. Output from debug shows: ------------------------------- [Aug 11 22:00:01] NOTICE[20173]: loader.c:875 load_modules: 1 modules will be loaded. [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading

Hi It's possible using a variable in the iftime keyword argument? E.g: context text { s => { timerange = '06:00-12:00|*|*|*'; ifTime(${timerange} { Playback(ivr/goodbye); } } } thanks [image: Sua Foto] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS | Mobile: (51) 8174-7956 <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>

Hi, I am having a problem with (Asterisk is crashing) with a Fritz card PCI / chan_capi. Receiving Calls from PSTN works, but outbound calls make asterisk crash (Speicherzugriffsfehler/Segmentation fault). The crash occurs upon dialing with the other phone not even ringing. I hereby ask if somebody reading this list can confirm or disprove my issue. Does anbody run a recent asterisk 2.6 with

Hello, i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package: libpri-1.4.2 asterisk-1.4.9 spandsp-0.0.4 unicall-0.0.5pre1 libmfcr2-0.0.3 libsupertone-0.0.2 libunicall-0.0.3 zaptel-1.4.4 i'm using a E1 pci card with R2 but they not work, when I start the asterisk its generate this log: [May 12 08:53:24] WARNING[30814] channel.c: No channel type

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

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>

"show dialplan" keeps showing contexts created by AEL. I tried deleting /etc/asterisk/extensions.ael but kept getting these messages in the Asterisk log: Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory Feb 14 21:39:53 WARNING[6074] pbx.c: Requested contexts didn't get merged Is there any way to delete or

Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI> module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47]

How do I stop loading extensions.ael dial plan? I'm only using extension.conf. -- Joseph

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

In a (futile?) attempt to get rid of warnings, I have this: [Nov 30 10:39:49] NOTICE[68467]: loader.c:937 load_modules: 149 modules will be loaded. [Nov 30 10:39:49] WARNING[68467]: utils.c:1427 __ast_string_field_init: trying to reset empty pool (5 times more) SIP channel loading... (5 lines of AEL loading) [Nov 30 10:39:49] NOTICE[68467]: pbx_ael.c:149 pbx_load_module: AEL load process:

I have two offices sharing a phone system. They also share a common internal context because all of the employees of the second office also work for the first office. Each office has 4 outside lines and I have defined 2 channel groups in my zapata.conf. The second office needs all of their outgoing calls to go out over their lines so the people they call will have the correct callerID. I

Hi I need to authenticate all the asterisk users from the LDAP server instead of from sip.conf. If anybody already have done this then please guide. I tried to integrate authenticate asterisk users from LDAP using the open source project astirectory1.2.0. After using the astirectory1.2.0 , now when the asterisk starts then it registeres with the LDAP. Following logs shows it. Jan 18 18:36:20

Hi all, I'm new to asterisk. I have it installed and 1 x100p card. I'm trying to use kphone to connect out. But, when I start * it gives me sound card busy error. I've checked ps aux and nothing seems to have the sound card. Any ideas? Does starting asterisk automatically take the sound card? Also, can you use the phone connected into the XFO card w/ asterisk or do you need an XFS card

On starting Asterisk (1.4) I get a whole bunch of WARNING[5858]: pbx_ael.c:4040 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! I find it a bit disturbing that this message has a level of WARNING (instead of NOTICE maybe) because the extensions in question are empty on purpose. The only reason they exist are the hints. hint(SIP/3000) 3000 => {}

Hello, I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -------------- next part

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found