Displaying 20 results from an estimated 2000 matches similar to: "asterisk-users Digest, Vol 59, Issue 28"
2011 Sep 14
1
Sip re-register / delay problem.
Hello,
For the moment I have the following settings in my sip.conf. I want to
optimize them to archive the following things:
- for the moment all my users will re-register too often. I want that only
lagged users to re-register quickly.
- check from time to time all users but no too often to see if is logged and
can be called.
Overall i want only lagged users to reregister and users with good
2010 Jul 26
1
Optimize peers registration under jitter/delay.
Hello,
I want to optimize my registrations and calls of peers to my asterisk
with the following options in sip.conf:
---///---
qualify = yes
qualify = 500
qualifyfreq=5
registerattempts = 0
registertimeout = 10
maxexpiry = 60
minexpiry = 20
defaultexpiry = 600
---///---
Can someone more experienced with these settings to help me to
optimize connections from peers with mobile phone that using
2010 Nov 03
1
inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
<--- SIP read from UDP:147.135.32.221:5060 --->
INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0
Call-ID: 31007e-31 at 147.135.32.221
CSeq: 1 INVITE
From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
To: "Gregory Malsack"<sip:s at
2015 Aug 05
2
Asterisk uses "Anonymous", but why?
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank"
2015 Aug 06
4
Asterisk uses "Anonymous", but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com>
wrote:
> Tested with X-Lite and it worked fiine. Is there some way to replace
> "Anonymous" with a config parameter?
>
> Thanks for your kind help
>
> ----------------------------------------
> > From: murthy64 at hotmail.com
> > To: asterisk-users at lists.digium.com
>
2007 Oct 03
1
Asterisk Keep Loosing Registration
Hello All,
For some odd reasons my Asterisk is keep on loosing registration of my
SIP devices. On the SIP device it shows I am RESISTED but when I do "sip
show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on
flapping "Peer '9099993456' is now UNREACHABLE!" and "Peer '9099993456'
is now REACHABLE!"...
I changed my
2015 Jul 29
3
Windows Asterisk Help
Hi All,
Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7.
Here is my sip.conf
[general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV
2008 Oct 12
5
One Way Audio Problem
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS
(channel 1). My SIP extension phone located inside the LAN is a SNOM
300 IP phone.
This one way audio
2015 Jul 29
2
Windows Asterisk Help
To: asterisk-users at lists.digium.com
From: webaccounts173 at jgoettgens.de
Date: Wed, 29 Jul 2015 16:11:31 +0200
Subject: Re: [asterisk-users] Windows Asterisk Help
Downloaded latest version of Asterisk from
www.asteriskwin32.com and installed on Windows 7.
Here is my sip.conf
2009 Jan 27
2
Module res_odbc is not loading
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
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2007 Apr 18
2
incoming SIP call
Hello all,
I'm having a quite simple configuration like:
SIP provider <=> asterisk SIP <=> lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net
2019 Oct 08
2
defaultexpiry & maxexpiry on peer level
Hello
is it possible to determine the SIP.conf parameters 'defaultexpirty' and
'maxexpiry' on a peer basis ?
My default value is 300 seconds, but some specific SIP-clients can only
send a SIP REGISTER every 3600 seconds. In current configuration these
SIP peers now become "Unreachable" after 300 seconds.
Or is there another way to differentiate ?
Kind regards.
2009 Jan 07
5
recommendation for German sound files
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
2009 Jan 26
2
custom cdr userfiled
Dear,
I added new field to cdr table , named "service" and type varchar(20),
but in extensions.conf with the following command, nothing to be saved.
exten => _X.,1,Set(CDR(service)=OUT)
does asterisk support this ability ?
is any setting must be changed, before that ?
best
Mani
2009 Nov 10
2
Hangup
Hi, is it possible to hangup a channel from another channel?
I want to finish a call from another channel, but if I put
exten => h,n,HangUp(channelname)
it doesn't hangup... Is that correct?
Thanks,
_________________________________________________________________
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2009 Feb 09
2
InUse&Ringing
Hello,
I'm just wondering if anyone has fixed the 'InUse&Ringing' problem.
* v1.4.23.1
Ta
2009 May 24
7
Asterisk, SQL Database Update
Is there any method in Asterisk to enable the updating process
into another SQL database through entering IVR options during the call.
Thanks a lot.
_________________________________________________________________
Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!
2009 Mar 04
5
AEL2: If-then-else not permitted in Switch-Case
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
switch(${DIALSTATUS})
{
case NOANSWER:
{
// if-then-else not permitted
If (${ael-var} = 1)
{
Playback(beep);
2009 May 17
4
Can YOU find a trailing parenthesis?
On 1.6.1, I must be losing my eyesight:
[internal]
include => outbound-pstn
.............
include => meetme ; 2663
include => setup-meetme-conf-room ; 6000xxxYYYY
[setup-meetme-conf-room]
exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" )
........
CLI:
-- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49]
2010 May 16
1
Aastra SIP phone regisration problems
I have 8 aastra phones that are loosing registration. On the phone gui it
says 408 as the registation error after a minute or say they register. In
the cli it eill say the phone is now unreachable then it will show it
registering then available. At first they did it every hour all the phones.
After messing with the experation it does it every 15 nin.
Any ideas on how to troubleshoot this? I tried