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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 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,

does anybody know a good tutorial on how to setup asterisk to receive faxes (no need to send them) ? i've tried using "app_fax.so" with T38 but i keep getting "Transmission failed" -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110503/5025a050/attachment.htm>

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:

...-------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/ffa02105/attachment-0001.htm> > > ------------------------------ > > Message: 8 > Date: Fri, 8 Apr 2011 12:26:27 -0400 > From: vip killa <vipkilla at gmail.com> > Subject: Re: [asterisk-users] asterisk login to voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <BANLkTikXudxp=W=K+B6TzyCDVC10bFD17Q at mail.gmail.com> > Content-Type: text/pl...

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>

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

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>

...ified text was scrubbed... > Name: panel.txt > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110409/bfbd159f/attachment-0001.txt> > > ------------------------------ > > Message: 9 > Date: Sun, 10 Apr 2011 00:56:52 -0400 > From: vip killa <vipkilla at gmail.com> > Subject: Re: [asterisk-users] send voicemail to multiple emails > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <BANLkTikreFbYaThF_ZqWF86tPiD+j-w9vw at mail.gmail.com> > Content-Type: t...

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:

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:

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

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>

For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/9e69a5c9/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:

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

Asterisk intermittently does not send audio back to the callers in the meetme conference bridge. If the caller hangs up and calls back sometimes the audio will work and sometimes it does not. We have taken packet captures and reviewed the SIP and SDP, both are correct and you can actually hear the audio being transmitted from the callers to the conference bridge but no audio is sent back to the

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.