similar to: Error in app_voicemail2.so after CVS update

Hi, Yesterday evening I have done a full update of Asterisk on a test system. The version is CVS-08/25/03-15:55:51 After this operation I get some big problems: - the Voicemail2 application does not work anymore. I must disable it in modules.conf file in order to be able to start * without crashing. The following settings: noload => app_voicemail2.so noload => app_sayunixtime.so If

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

Hi, compiled 1.0 on macosx latest (10.3.5). compiled fine. when running, complains about voicemail2 module. Any hints? Marc. == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium. Written by Mark Spencer <markster@digium.com>

Does anybody have any reason why I should *not* permenantly replace app_voicemail with app_voicemail2? If so, speak now or forever cvs update -D "8/18/2003". Mark

Hi All, I am a newbie and having some trouble getting Asterisk to work. I have checked out zapata zaptel libpri and asterisk (both v1-0_stable and regular--in separate directories). All built according to the documentation I have found and installed correctly. I have modified/created zaptel.conf zapata.conf as attached. I get the following messages/error during asterisk startup. ==

Hi, after cvs upgrading my * installation yesterday, the prompts in both VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial "blip" followed by the Voice taking breath and being cut off before she has a chance to say "Comedian Mail". All other prompts (ie the Playback application) seem to work fine. I can still login to VoiceMailMain2, however, each

Hi all - I'm making gradual progress implementing asterisk on my box! Now, when I type asterisk it dies at this point. Does anyone have any idea why this is happening! It have checked everything but running out of options! [app_voicemail2.so] => (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2'

Whenever I issue the reload command, asterisk crashes. Below is the output I get from (gdb) bt Any help is appreciated. *************************************************************** *CLI> reload == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or

I've commited changes to Voicemail2: * Handle properly when being left a message while checking VM -- this should fix the "saving to your inbox" issue too, at least in principle. And to DISA: * Properly handle extensions with multiple matches and "dots" Please let me know on or off list about any feedback you have regarding these changes. Mark

Hi all, i'm trying to install Asterisk with Junghanns QuadBri ISDN card: I have followed all the instructions, and when I run "|ztcfg -tvv" I get | | Zaptel Configuration ======================| SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02)

Hi list, I am using Asterisk1.6.2 form a long time and upgarding to Asterisk-1.8.23.1. I am using mysql backend for cdr. in asterisk-1.6.2 i have usegmtime=yes and it works as expected insert cdr date in GMT0. now i tested Asterisk-1.8.23.1 and asterisk-11.5 with same results no matter what i configure in cdr_mysql.conf "timezone=UTC usegmtime=yes" cdr always inserted in local time. I

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

I submiited to the list last night, but it never showed up. Here we go again. I've tried building Asterisk 1.4.15 on Solaris based on instuctions here, http://forums.digium.com/viewtopic.php?t=5888. However, this is the message I get. This is Solaris on X86. Any ideas? [CC] stdtime/localtime.c -> stdtime/localtime.o stdtime/localtime.c: In function `localsub':

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

I am still getting a segmentation fault when I try to reload asterisk Here is the output: Asterisk Ready. *CLI> reload == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/rtp.conf': Not found (No such file or directory) == RTP Allocating from port range 5000 -> 31000 -- Reloading module

I am having trouble with FC3. After doing a yum update (of 1264 packages) I still cannont compile 1.2.1 from source: make[1]: `libedit.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline' make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: `libdb1.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/db1-ast'

I've just made (on FreeBSD 4.10) cvs co asterisk gmake gmake install asterisk -vvvvvvc and got a new error message (was working ok before the cvs) [codec_gsm.so]Oct 6 17:23:16 WARNING[135192576]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/codec_gsm.so: Undefined symbol "Short_term_analysis_filteringx" Oct 6 17:23:16 WARNING[135192576]: loader.c:429 load_modules:

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten => 400,1,Answer() exten => 400,n,Wait,1 exten => 400,n,SayUnixTime(,EST5EDT,) exten => 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious!