similar to: Pass variable to voicemail script

2015-05-06 17:51 GMT+02:00 Tech Support <asterisk at voipbusiness.us>: > Hey; > > It seems to me that for what you want to do, it would be easier just > to email the user the voicemail audio file as an attachment. > Yes that's true but still, for users with an "intensive voicemail usage", unified messaging is attractive. In the long run, is unified

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 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>

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: > ------------ > Requirements > ------------ > ....... > Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Where and how do I install it into my Asterisk box?

Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

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

-----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

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

I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street

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 need to grab the voicemail WAV file once the voicemail command is done. Is there a hook to be notified that voicemail is done, and get the name of the recorded file? Thanks MD

I'm trying to use Asterisk for its voicemail capabilities while interfacing with a legacy Toshiba PBX. Is there a way to have Asterisk send a DTMF code to an extension to turn on the message waiting indicator light? When a user leaves a voicemail, I want Asterisk to pick up one of the lines attached to it, and then dial #63<ext>, which is what sets the message waiting indicator light

All; I'm currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the 'Directory' app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there

Dear list, i have a issue Asterisk crash (Module res_odbc exactly) after the same log who is "/ERROR[23805] astobj2.c: bad magic number.../" you will see on the log : Today [2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958* [2016-02-24 16:00:44] Asterisk 11.2-cert1 built by root @ Voice_server on a x86_64 running Linux on 2013-04-09

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.

Hi, I would like to see if during a call a new voicemail was recorded. I want to send a SMS to mobile phones if someone recorded a message on our voicemail system. I can use VMCOUNT to see if there are new messages in the Inbox but this will result in new SMS being sent even if the caller hangs up during the Voicemailpromt, at least if there are still unread/unheard messages in the inbox. Is

We have a couple of Asterisk boxes with one being the main system with everyone's voicemail and the other a slave used merely to link a couple of remote phones to the main system using IAX. How can one propagate message waiting indication from the main system to the remote phones? g.