On 09/04/11 23:40, Jeremy Kister wrote:
On 9/4/2011 10:48 PM, Joseph wrote:
[globals]
DYNAMIC_FEATURES=automon
= not =
exten = 11,1,GotoIfTime(*,*,1,jan?holiday,s,1) ; new years day
hmm, the syntax seems ok. is func_logic.so loaded?
asterisk -rx 'module show like logic'
Thanks Jeremy
Hi Sammy,
Ans of 1st question:-
As per my experiance Asterisk realtime(DB) based data will lost when your
server is creash and you may not take backup of your server's DB.
If any one know then plese guide me so that I will start working on it.
Ans of 2nd question:-
Your question is correct if
1- Per my experience I've used DB with configuration files and I was amazed
that Asterisk was taking a union of DB + conf file configurations and
accepting both.So if you just make a simple script or DB function to do file
operation on some event/cronjob you'll be saved.
Moreover, if that still
Afternoon All,
Is anyone aware of a way to generate ringing as opposed to starting music on
hold for the party originating a call with followme?
I'm assuming its doable as it looks like FreePBX users get the option (Not to
say that FreePBX haven't got their own followme implementation though).
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote:
On 09/01/2011 04:39 PM, Hans Witvliet wrote:
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.
You are
Hi Sammy,
Thanks for share your experance and provide a new way of Asterisk
communication with DB.
Actually I am using this feature only for MOH feature of asterisk right now.
But I will used it to all the configuration files too as per the needs.
I am not too much aware abut the Asterisk DB
Hello,
I'm trying to page the Cisco SPA 941 by adding the SIP-header Call-Info:
answer-after=0
dialplan :
exten = _*XX*,n,SIPAddHeader(Call-Info: answer-after=0)
SIP debug :
INVITE sip:testcorp6@192.168.1.106:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5060;branch=z9hG4bK6090dca4;rport
Are you talking about AstDB or MySQL as DB backend for asterisk?
On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati virbh...@gmail.com wrote:
Hi Sammy,
Thanks for share your experance and provide a new way of Asterisk
communication with DB.
Actually I am using this feature only for MOH feature
Hi Sammy,
Yes I am asking about AstDB only.
On Mon, Sep 5, 2011 at 2:00 PM, Sam Govind govoi...@gmail.com wrote:
Are you talking about AstDB or MySQL as DB backend for asterisk?
On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati virbh...@gmail.com wrote:
Hi Sammy,
Thanks for share your
On Mon, Sep 05, 2011 at 12:15:16PM +0530, virendra bhati wrote:
Hi Sammy,
Ans of 1st question:-
As per my experiance Asterisk realtime(DB) based data will lost when your
server is creash and you may not take backup of your server's DB.
If any one know then plese guide me so that I will
On 5/09/2011 4:27 PM, Jonas Kellens wrote:
Hello,
I'm trying to page the Cisco SPA 941 by adding the SIP-header
Call-Info: answer-after=0
dialplan :
exten = _*XX*,n,SIPAddHeader(Call-Info: answer-after=0)
Try
exten = _*XX*,n,SIPAddHeader(Call-Info:\;Answer-After=0)
Larry.
--
I mean the directory of phone numbers is stored within asterisk. So the SIP
phone just fetch that list when it starts.
--
Message: 3
Date: Sun, 4 Sep 2011 19:47:00 -0400
From: Robert-iPhone rhuddles...@gmail.com
Subject: Re: [asterisk-users] Phone numbers and
Hello,
I think this not posible.
You can use remote phonebook the phones can share.
For example for yealink phone it's posible create a XML file with the phonebook
and from each phone access to this list.
Regards
--
_
--
Hello list,
I don't really understand how AMI works.
I read some information and examples on the net, but they all show how
you login to the AMI, give an action and receive a response. The end.
I guess you just re-run the script every time you want the action to be
executed.
How then does
On Sunday 04 September 2011, neo haux wrote:
Hi,
It is possible to save all the phones numbers on asterisk servers instead
of doing so manually in each VoIP device ?
Does SIP take care of such configuration ?
If you want your phones to be self-configuring (which is a good idea even if
Hi,
I've just upgraded to 1.8.6 on one server and I've been getting a lot of
codec warning, like this:
WARNING[21211]: chan_sip.c:6341 sip_write: Asked to transmit frame type
ulaw, while native formats is 0x100 (g729) read/write = 0x100 (g729)/0x100
(g729)
I do have a Digium transcoder
someone can help me to solve this problem?
thanks
--
From: Alessio ales...@asistar.it
Sent: Friday, September 02, 2011 5:10 PM
To: Lee Howard fax...@howardsilvan.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio,
remote_lostpackets)})
2011/9/5 Catalin S. jonsonpla...@gmail.com
Hello,
I have a problem with some variables in 1.8.6.0. I set on extension the
following lines:
exten = h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio,
local_lostpackets)}) ; lost packets by local end **
exten = h, n, Set (CDR (PCR) =
On 5/09/2011 10:05 PM, Alessio wrote:
someone can help me to solve this problem?
thanks
--
From: Alessio ales...@asistar.it
Sent: Friday, September 02, 2011 5:10 PM
To: Lee Howard fax...@howardsilvan.com
Cc: Asterisk Users Mailing List -
I have DID, it registers OK with the provider, but when I try to call this
number (it suppose to ring my Asterisk) asterisk 1.8 does not respond.
sip show peers
Name/username Host Dyn Forcerport ACL Port Status
actio-out/48746612254 81.15.150.20 N 5060
It seems to me nat=yes is not working correctly in asterisk 1.8.5
rtp set debug on
shows:
Got RTP packet from10.0.0.110:6000 (type 00, seq 029667, ts 2129095321,
len 000160)
Sent RTP packet to 10.0.0.110:6010 (type 00, seq 065112, ts 2129095320,
len 000160)
I've tried 'nat=yes'
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