Hi. i am running asterisk 11 in debian and i would like have a missed call notification down to extension level.
so if i get a missed call to extension 6589 then send an email to the user's email address with a subject and a text message.
is there a guide on how to create something like that?
n your extensions.conf you could and probably should set the following option prior to dialing the IAX channel, this is to enable the T.38 gateway feature of Asterisk 11:
Set(FAXOPT(gateway)=yes)
I have it working in my installation however I have incoming voice calls too hence
hi. i am using asterisk 11.24.1 in my raspberry. i do have a sip trunk with a provider with g711a. I am trying to setup a fax server by following the guide in http://the-asterisk-book.com/1.6/faxserver.html.
i do live in Greece and the number is 00302112152130
the problem is that i am getting
I did edit the features.conf. what about extensions.conf?
On 16/10/2016, 14:45 Doug Lytle <supp...@drdos.info> wrote:
On 10/16/2016 02:33 AM, tux john wrote:
The problem is that I do not know how to configure the feature access codes including tr
Thanks for the reply. I do know the security practices and I am using VoIP. The problem is that I do not know how to configure the feature access codes including transfer.
On 15/10/2016, 21:42 Steve Edwards <asterisk.org@sedwards.com> wrote:
On Sat, 15 Oct 2016, tux john
Hi. Kinda new to the area and I would like some help please. I have asterisk 11 in my system and I have 10 users and 12 DIDs. One did routed to each user and 2 DIDs for faxing. Everything works fine but I do not have call transfer between extensions and feature access codes. I have read somewhere
in asterisk 11 i have an extension 4450 that gets all the calls from a particular number (eg 1234567890).
in extensions.conf i do have the following:
exten => 1234567890,n,Dial(SIP/4450,20,mg)
exten => 1234567890,n,Hangup()
the problem that i have is that when extension 4450 is busy then
hi. i am running asterisk 11 and i am using astdb to store all my contacts and their numbers. so everytime they call me, i can see their name on the screen of the phone.
i am making use of the following to retrieve the name from the astdb
exten =>
hi.i managed to record my voicemail greeting. the only problem is that after my greeting the caller hear '...please leave your message after the tone. when done press the pound key or hangup.' is there a way to get rid of that?
Ideally i would like to have my own recording and then the beep
Hi. I would like to blacklist a few callers and I have been using the *CLI> database put blacklist 1234 "annoying callers". Instead of putting the same command for every user is there any way to have a file? Ideally a file in /opt that I would update the blacklisted numbers (add,remove). Is there
Hi. I am running asterisk 11 without any GUI and I am running a phonebook by making use of asterisk -rx "database show cidname"
Because that method is not easy to maintain I would like to ask if it is possible to sync my Google contacts to that database. Then I could set a cron job to run 5 times
being honest with i have been lost on what to do.
all i want is sent from my email a pdf file and then the server will sent it as fax.
what settings do i have to do regarding emailing to the server? what other settings do i have to do?
is there a guide on that?
Sent:Friday, June 26, 2015 at
At the moment it receives fax and sends it as pdf to my email.
i have followed this tutorial http://the-asterisk-book.com/1.6/faxserver-mit-iaxmodem-und-hylafax.html
i have setup postfix to send using my gmail account.
how do i do the reverse to send from my email?
Sent:Thursday, July 02,
hello everyone.
i am using asterisk 11.16 in my home office and i am using fax to email with it. i am quite happy with the way it works, no problems at all. when a fax arrives in a particular DID then the system sends it with mailutils to my email address as pdf. there is not any email address
a similar setup with 3 xspa3102 are working in a law office for 2 years non stop. they are very reliable, but a pain to configure them.
regarding freepbx is very helpful if you do not want to have yourself involved with pure asterisk.
Sent:Wednesday, June 17, 2015 at 4:12 PM
From:Michelle
Good day,
i am using a raspberry pi 2, where i am running asterisk 1.8. it runs really well.
to interface the 2 PSTN lines i am using 2x linksys spa3102
costs:
-raspberry with case and power supply 50
-spa 3102 25 each.
total=100
also i do have a couple of sip trunks with betamax.
the
Hello everyone. i am running an asterisk server and i would like to have the contacts from google.
so every inbound call with fetch the caller ID from google contacts and present it to my screen.
could someone help me on that please?
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2015, tux john wrote:
Hello everyone. i am running an asterisk server and i would like to have
the contacts from google. so every inbound call with fetch the caller ID
from google contacts and present it to my screen.
This is really three problems, as follows:
(1) Accessing the Google Contacts
hello everyone,
i am using asterisk 11.6 with asterisk-gui and i am stuck in setting up the
linksys spa3102 and the sip trunk for it.
may i have some help please?
friendly,
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