similar to: voicemail call back loop

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'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 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 have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in

I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Right now, I am setting up asterisk to use voicemail with my Cisco Call Manager (Which I detest BTW...) and I have everything working, EXCEPT: I cannot get my externnotify script to run when any changes have been made to the VoiceMail... Scenario: Bob gets a call -> Bob

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

On Thu, Mar 23, 2023 at 10:18:56AM +0300, Viktor Prutyanov wrote: > On Thu, Mar 23, 2023 at 4:22?AM Xuan Zhuo <xuanzhuo at linux.alibaba.com> wrote: > > > > On Wed, 22 Mar 2023 17:10:31 +0300, Viktor Prutyanov <viktor at daynix.com> wrote: > > > According to VirtIO spec v1.2, VIRTIO_F_NOTIFICATION_DATA feature > > > indicates that the driver passes

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

On Wed, 22 Mar 2023 17:10:31 +0300, Viktor Prutyanov <viktor at daynix.com> wrote: > According to VirtIO spec v1.2, VIRTIO_F_NOTIFICATION_DATA feature > indicates that the driver passes extra data along with the queue > notifications. > > In a split queue case, the extra data is 16-bit available index. In a > packed queue case, the extra data is 1-bit wrap counter and

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

On Thu, Apr 19, 2018 at 07:39:21PM +0200, Paolo Bonzini wrote: > On 19/04/2018 19:35, Michael S. Tsirkin wrote: > > virtio is using barriers to order memory accesses, thus > > dma_wmb/rmb is a good match. > > > > Build-tested on x86: Before > > > > [mst at tuck linux]$ size drivers/virtio/virtio_ring.o > > text data bss dec hex

On Thu, Apr 19, 2018 at 07:39:21PM +0200, Paolo Bonzini wrote: > On 19/04/2018 19:35, Michael S. Tsirkin wrote: > > virtio is using barriers to order memory accesses, thus > > dma_wmb/rmb is a good match. > > > > Build-tested on x86: Before > > > > [mst at tuck linux]$ size drivers/virtio/virtio_ring.o > > text data bss dec hex

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this