similar to: voicemail odbc "Length is ....."

Displaying 20 results from an estimated 10000 matches similar to: "voicemail odbc "Length is .....""

2011 Apr 08
9
send voicemail to multiple emails
Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>
2011 Jun 21
4
call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>
2011 Apr 05
4
agi voicemail callback
I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Mar 28
8
asterisk and fail2ban
Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,
2011 Jun 15
2
change destination on digit
Is there an easy way to setup diaplan so when someone pushes a digit such as * during a call, they will be transferred to another destination. For example, a caller is hearing ringing while calling a UA, but instead of waiting for the UA to pick up, they can push * and go directly to that UA's voicemail. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Apr 27
2
asterisk practices
I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a
2011 May 27
2
disable sip registration
Is there a way to disable all SIP registration and block any requests? The reason I'm asking is this particular Asterisk server will just be originating calls. I've noticed sip attacks where the attacker attempts to register a user 100x per second causing CPU to rise significantly. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 Apr 06
2
voicemail call back loop
I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the
2011 May 18
3
asterisk's zombie processes
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part
2011 Apr 07
1
asterisk login to voicemail
Is there a way to login to a voicemail box when someone pushes '#' during greeting? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110407/176ba67b/attachment.htm>
2011 May 16
2
AMI perl daemon
Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.
2011 Feb 23
2
REFER and dialplan broken (as documented in chan_sip.c on line 11951)
There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c "this is currently broken as we have no way of telling the
2011 Mar 23
7
asking for some help
hi evrey one, i'm in some kind interesting in developping some asterisk programme like doing a small programme including some of these services that do a telephone operator. but abviously i need to know about programming in asterisk in thos to files i think :) (extensions.conf and in sip.conf files) so i'm asking if someone can give me a puch,i will be very glad thanks in advance
2012 Apr 03
5
process_sdp: Multiple audio streams are not supported
Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not
2006 Nov 27
3
Voicemail, SQL & ODBC
Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? And is the use of mySQL and ODBC at the same time still a bad idea? If so, why? I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a
2006 Nov 15
2
ODBC Voicemail Storage
I current have a working Asterisk 1.2.12 server with ODBC voicemail storage, realtime static maps for voicemail, sip and iax configuration files. Realtime extensions, etc. All works great. I have verified that this configuration works on my test server as well. Now I am trying to test the 1.4B3 version on the same test server, and all works well except for ODBC voicemail. I am using the same
2005 Jul 20
2
Test CVS HEAD Voicemail ODBC Storage
As we are getting closer to release of CVS head as version 1.2, we're in need of your help. One of the cool new features in CVS head is the ability to store the actual voicemail messages in a database. This is not using ARA, the Asterisk Realtime Architecture, but directly interfaces with ODBC from app_voicemail. It stores both meta data and audio in the database, which will give you real
2004 May 14
4
app_dbmysql and ODBC Voicemail
I have done a little work on asterisk and database integration. Below is a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure MySQL. I also ported the mysql-vm-routines.h to ODBC in case anyone is interested. You can get both of these from: http://www.cheapnet.net/~mike/asterisk They were working as of yesterday CVS, but today CVS will not compile and I have not looked
2010 Apr 07
1
celt codec for red5phone
I'm wondering if anyone is familiar with red5 (flash media server) and the phone application (http://code.google.com/p/red5phone/) It's written in java. Red5phone already uses the codecs PCMU, PCMA, iLBC and G.729. I'd like to see CELT added since it is open source and the best quality codec out there. Would anyone know how to port the celt codec to this application or to java? If
2011 Apr 01
1
call parking issues in asterisk 1.6.2.16.2
We have a problem of no MoH when parking calls running asterisk 1.6.2.16.2. Also, the parked call never goes back to the parker. We have "comebacktoorigin = yes" and "parkingtime => 180" in features.conf Anybody know why this isn't working? -------------- next part -------------- An HTML attachment was scrubbed... URL: