similar to: agi voicemail callback

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>

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

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

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.

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>

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,

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:

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:

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

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

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

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

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

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

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:

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>

I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see "Length is 186545" or something similar but a different number in Asterisk CLI everytime someone leaves a message? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110411/be7ef252/attachment.htm>

I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -------------- next part -------------- An HTML

http://pastebin.com/vxGM2n5j We are getting those errors 100x per second in console when AGI set debug is on.... It is causing extremely high CPU usage, we've tried asterisk version 1.6.1.22 and 1.6.2.18 It seems the problem is worse in 1.6.2.18 Can someone advise how to fix this? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing