Hi, all.
Is there any chance to set individual CALLERID(num) for channels SIP/peer1,
SIP/peer2 in a call Dial(SIP/peer1SIP/peer2). There is an option to use
Dial(SIP/peer1SIP/peer2,,M(set_callerid)), but the macro will be launched
after the channel answered. Not really want to use local
trying
https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification
06.08.2014, 10:13, r...@yandex.ru r...@yandex.ru:
Hi, all.
Is there any chance to set individual CALLERID(num) for channels SIP/peer1,
SIP/peer2 in a call Dial(SIP/peer1SIP/peer2). There is an option to use
Hello,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.
Can you guys tell me what might be causing this? I
if you use a papt2 or so spa2101 then you could have alert info set to
different lengths or styles of ringers
i use that in a dorm with phones and have the phones ring short rings at
night so it wont wake up the students
On Tue, Aug 5, 2014 at 10:24 PM, Kevin Larsen
Olli Heiskanen wrote:
Hello,
Kia ora,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the
INVITE leaves Asterisk.
Can you guys tell
Hi,
There we go, that was it. Thank you Joshua!
cheers,
Olli
2014-08-06 15:26 GMT+03:00 Joshua Colp jc...@digium.com:
Olli Heiskanen wrote:
Hello,
Kia ora,
I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the
if you use a papt2 or so spa2101 then you could have alert info set
to different lengths or styles of ringers
i use that in a dorm with phones and have the phones ring short
rings at night so it wont wake up the students
I do not use either of those devices, but after posting this
Hello
We use originate that places a call in a queue (channel parameter is a
Local/dialplan)
When the call is answered in queue, it is bridged with the operator, and
then starts the second channel leg: Dial out to wherever trough local
channel
we set a sip header with dialstatus, so if the
The basic concept is that the original call will run a script that creates a
call file to call the paging system and play a specific audio file. It also
passes into the paging call its channel name. In the call to the paging
system, I use the SHARED function to write back to the original calls'
I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood,
and I was hoping someone on this list might have managed to do this.
I have configured some numbers to route to a SIP endpoint
%e164@customer's server
and convinced the customer to open up UDP ports 5060 and 1 -
Will your approach handle ringing more than one of the three
extensions simultaneously?
--Don
Not if they are in the same paging zone, but neither would using the night
ringer function on the pa system, so I consider that acceptable. Not even
sure what would be considered correct in
All,
I am running the following setup:
Linux 3.10.0-123.4.2.el7.x86_64 #1 SMP Mon Jun 30 16:09:14 UTC 2014 x86_64
x86_64 x86_64 GNU/Linux
Asterisk 12.4.0
Cisco IP Phone 7960G
I have an issue with MWI. For some reason after I delete my voicemail messages,
the MWI of the phone is ON for
Tonight the Asterisk issue tracker will have intermittent availability
due to maintenance. This maintenance will begin at approximately 9:00
PM CST[1] and should last no longer than one hour.
The affected services are:
* issues.asterisk.org
Thank you for your support!
[1]:
We have been running around than 40 asterisk servers running on Debian
Squeeze for last three years, handling traffic of more than few
hundred thousand calls per day.
Our setup's PRI-banks were using Redfone's Fonebridge. We had PRIs
from multiple telephony providers. And Redfone's Fonebridge
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