similar to: IAXY & Voicemailmain problem

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

Hi all, I've been told that part of best practices is to noload all modules that you don't use. Adsi being one of them (are there actually people using adsi & asterisk?) I noloaded it in /etc/asterisk/modules.conf: noload => res_adsi.so noload => app_adsiprog.so When I start * (this is cvs head of today by the way) I get: [app_voicemail.so]Aug 31 17:25:49 WARNING[4961]:

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to

Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>

On Fri, 2017-02-17 at 11:11 +1300, Gary Lockyer wrote: > Script to provide an anonymous summary from tshark > > The tshark command needs to output a PDML XML stream, which this > command > will read. The summary is intended not to expose private or customer > data while allowing a good view on the range and frequency of the > network traffic. The script Gary posted, which is

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Im having a little problem with voicemail and my cisco phones i was wondering if anyone might have seen this before and let me know whats going on. it spits this out and then my cisco ip phone reboots im using the latest cvs and a cisco 7910 phone WARNING[1234379840]: File res_adsi.c, Line 205 (__adsi_transmit_messages): Unable to send CAS -- Playing 'vm-login' (language 'en')

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason

Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

Im trying to get IAX to work between my * and FWD. I activated my iax2 account on iax.fwdnet.net and I get the output: "Registered to '65.39.205.121', who sees us as 68.14.203.254:4569" when I start asterisk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave

Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

Hey guys, Im interested in hearing about servers (and thier hardware specs) that successfully run both asterisk and samba for an office of maybe about 12 extensions (SIP) and about 12 workstations. Im hopeing to not only replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but

Hey guys, Im about to sign up for VoicePulse Connect. Of course, I plan on using my asterisk server to "register =>" with the service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hi. I can't figure this one out. Hope someone can help me. root@pbx:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName=XXXX Password=XXXX Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings=

Hi there everybody, We''re having mongrel processes slowly die on us, with a segmentation fault. I''ve seen a few other people mention similar issues. Any info or suggestions would be nice: Error: /var/www/apps/heritage/current/config/../vendor/plugins/file_column/ lib/magick_file_column.rb:7: [BUG] Segmentation fault ruby 1.8.4 (2005-12-24) [i386-linux] Server:

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney