Displaying 20 results from an estimated 5000 matches similar to: "SIP registry fails during night"
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2009 Aug 30
2
MySQL syntax error : I really don't see where...
Hi list,
I'm stuck for the moment @ the following :
My Query (in a macro) :
exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\
vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\
SIPACCOUNT="${ARG1}")
Asterisk CLI :
[Aug 30 14:07:42] -- Executing [s at macro-vakantie:2]
MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set
2009 Jun 25
1
SIP registration fails
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
opened and 5060 forwarded to Asterisk (192.168.2.2)
Can someone see why SIP-registration fails ??
register => 092779077:XXXX at 85.119.188.3
[3starsnet]
type=peer
host=85.119.188.3
username=092779077
secret=XXXX
fromuser=092779077
fromdomain=sip.3starsnet.com
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
qualify=yes
2009 Jul 01
2
Registrations problems to SIP-provider.
Hello List,
I'm having problems with registrating my Asterisk-server to the
SIP-provider. Yesterday all worked fine, this evening I cannot call out.
What can be wrong ?
This is my registration in sip.conf :
register => 092779077:XXXX at 85.119.188.3
This the output of SIP show peers :
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port
Status
2009 Sep 29
1
Asterisk on DD-WRT : modules.conf not found
Through the optware-package I have installed Asterisk on an external
USB. Further I have a Linksys WRT610N with DD-WRT v24 mega.
I start asterisk with the following command : /opt/sbin/asterisk -c
I get the following WARNING :
root at DD-WRT:/opt/etc/asterisk# /opt/sbin/asterisk -c
Asterisk 1.4.22.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at
2009 Sep 17
1
I'm not getting the ability to leave a voicemail-message
I'm having a little problem with voicemail. Actually I'm not getting the
ability to leave a voicemail-message.
This is part of the dialplan :
> exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1})
> exten => s,n,NoOp(${ARG1}@boxes)
> exten => s,n,Voicemail(${ARG1}@boxes)
> exten => s,n,Hangup()
> exten => s,n,MacroExit
This is the
2009 Oct 03
1
Asterisk on NSLU2 : Grandstream not registering
Hello there !
I have successfully installed Asterisk on a normal server and on a
Linksys WRT610N with DD-WRT running from connected USB-stick. This is
not my first Asterisk-installation. I'm always finding a smaller and
smaller platform to install Asterisk on.
But now I encounter a problem.
I'm using the same Grandstream GXP2010 with a of my installations. Every
time registration goes
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2010 Jun 08
6
reloading realtime sip peers
Hello,
I noticed that changes to realtime sip peers are not applied until a
'reload'. A 'sip reload' does not make any changes to realtime sip peers.
When changing for instance the mailbox-parameter in the realtime
sip_buddies table, the change is not applied with a 'sip reload'.
For every change there is a complete 'reload' necessary.
Why does a 'sip
2009 Jun 27
1
2 problems I can't solve without any help
Problem 1 :
Incoming conversations from the SIP-provider come into the
[default]-context and to the 's'-extension.
I am unable to change this, even if I have :
sip.conf
[general]
;context=default ; Default context for incoming calls
register => 092779077:XXXX at 85.119.188.3
; incoming
[092779077]
type=user
host=85.119.188.3
context=from3starsnet
So I define no
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2010 Mar 01
2
Is answer() necessary ?
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where
the caller is given the
2009 Aug 29
1
GoToIfTime : how to define sep 25th till oct 10th ?
Hi list,
quick question :
With GoToIfTime, how to define a period of holiday that starts at the
end of the month and ends at the beginning of the next month ??
Like September 25th till October 10th when incoming calls need to go to
the voicemail...
Greetingz,
Jonas.
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2009 Aug 18
2
You do not appear to have the sources for the 2.6.20-prep kernel installed
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
"You do not appear to have the sources for the 2.6.20-prep kernel
installed."
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
- kernel-devel-2.6.18-128.4.1.el5.x86_64
- kernel-xen-devel-2.6.18-128.4.1.el5.x86_64
bash-3.2# uname -r
2.6.20-prep
bash-3.2# ls -l
2009 Oct 21
3
Searching on how to keep local calls... local
Hi list.
Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??
What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?
Config :
Asterisk --internet-- > router/firewall --> connected local IP-phones
Internal call :
2009 May 12
2
Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the following :
- Asterisk tells me that the conversation was hung up (the log files
tell me the command was executed)
- On my cell phone I hear
2009 Oct 14
2
FXS to SIP gateway
Hello list !
I don't have the money to test out all the products and reading the
manuals is not always that enlightening...
Does someone here know a good gateway-product that lets analogue
telephones communicate with an Asterisk-server.
I have found the Grandstream GXW-400x to be able to add SIP-accounts to
analogue telephone devices that are connected to the FXS-ports. Moreover
this
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 May 08
2
Not receiving voicemail message in mailbox
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
;
2009 Jun 23
1
SIP 482 Loop detected
-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8",
"conversation to GSM") in new stack
-- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 "Loop Detected" back from 85.119.188.3
-- Now forwarding