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
.
On Mon, 2009-04-13 at 12:28 -0400, Michael van der Stoop wrote:
jonas kellens wrote:
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
Tony Plack,
this is the result form Asterisk CLI :
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for
James,
when I run Asterisk -vr and I enter 210 on one phone to call the
other, nothing is displayed on the CommandLine...
I know this is not right, just don't know what is wrong. I really need
someone to guide me a bit...
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24,
Danny,
this is from the Asterisk CLI :
asterisk*CLI dialplan reload
Dialplan reloaded.
== Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'default'
-- Including context 'intern' in context 'default'
-- Registered extension context 'intern'
--
I pick up the phone, and dial 211 on the BT201. This is the Asterisk
CLI :
Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)
Verbosity is at least 5
asterisk*CLI
Nothing is displayed... it stays that way...
Jonas.
On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley
These are the settings on my BT201 (GXP1200 is the same interface) :
Account Name:(e.g., MyCompany)
SIP Server:(e.g., sip.mycompany.com, or IP address)
Outbound Proxy:(e.g., proxy.myprovider.com, or IP address)
SIP User ID:(the user part of an SIP address)
-- I put here the
Asterisk, the future of telephony...
Thanks for your reply !
Greetingz,
Jonas.
On Mon, 2009-04-13 at 14:04 -0400, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
jonas kellens wrote:
I pick up the phone, and dial 211 on the BT201. This is the Asterisk CLI :
/Connected
Hey there again !
I've changed some things now :
1) IP-phones get there IP from a DHCP
2) sip-accounts simplified :
[r...@asterisk asterisk]# cat sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
[210]
type=friend
context=intern
host=dynamic
at 06:18:58PM +0200, jonas kellens wrote:
I pick up the phone of the BT201 and dial 211... nothing happens.
I pick up the phone of the GXP1200 and dial 210... nothing happens.
I would love to have your feedback on this. Where could this problem be
situated ?
Your basic mistake
, jonas kellens wrote:
1) IP-phones get there IP from a DHCP
The source of the address is not the issue.
I still see no register-message on the CLI. This really should happen
now, as they are defined host=dynamic !
I suspect you have not [correctly] configured the phones to register
I will summarize everything again and try to answer all the questions
asked while I was away.
First I stop Asterisk :
[r...@asterisk asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes
There is something wrong with my IPtables !!!
When i do :
service iptables stop
I see my phones register on the CLI !!
I can place a call and the phone rings !! I see a whole lot of
SIP-requests on the CLI with SDP-message in body !! That's good news...
What is wrong with my IPtables-rule
For an Asterisk-environment with no more then 10 SIP-phones, I would
open 10 x 4 = 40 UDP ports for RTP/RTCP-traffic ( 4/call). Can you
confirm ?!
rtp.conf :
rtpstart=30500
rtpend=30550
Ok, there's 50 here... a round number right ?!
All SIP-communication stays on the LAN. There's a NIC
I have 2 SIP-clients defined in my sip.conf :
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=yes
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=yes
When I make a call from one to another this is
asterisk]# cat iax.conf
[general]
autokill=yes
bindport=4569
bindaddr=0.0.0.0
[Jonas]
type=friend
host=dynamic
;auth=md5
username=jonaskellens
password=zoiper
callerid=Jonas Kellens 100
context=intern
disallow=all
allow=gsm
allow=speex
allow=alaw
On the CLI :
Verbosity is at least 20
asterisk*CLI
14:38:01.229941 IP 192.168.4.240.sip 192.168.4.248.sip: SIP, length:
889
14:38:01.230127 IP 192.168.4.248.sip 192.168.4.240.sip: SIP, length:
515
14:38:01.251558 IP 192.168.4.240.sip 192.168.4.248.sip: SIP, length:
497
14:38:01.271714 IP 192.168.4.240.sip 192.168.4.248.sip: SIP, length:
1060
[r...@asterisk asterisk]# cat iax.conf
[general]
autokill=yes
bindport=4569
bindaddr=0.0.0.0
[jonaskellens]
type=friend
host=dynamic
;auth=md5
username=jonaskellens
password=zoiper
callerid=Jonas Kellens 100
context=intern
disallow=all
allow=gsm
allow=speex
allow=alaw
asterisk*CLI iax2 reload
How come the mask is 255.255.255.255 ??
asterisk*CLI iax2 show peers
Name/UsernameHost Mask Port
Status
jonaskellens/jo 192.168.4.169 (D) 255.255.255.255 4569
Unmonitored
1 iax2 peers [0 online, 0 offline, 1 unmonitored]
Greetingz,
Jonas.
How do I know that de hardware echo canceller module on my Digium
TDM403E is recognized by Asterisk ?
After having configured /etc/zaptel.conf :
[r...@asterisk etc]# /sbin/ztcfg -vv
Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01:
VoIP-wiki.org states :
Digium resources http://www.asterisk.org/zaptel-to-dahdi
/etc/zaptel.conf Becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.conf
Now, what do I have installed on my system :
/etc/zaptel.conf and /etc/asterisk/chan_dahdi.conf
Will
.
Forwarded Message
From: jonas kellens jonas.kell...@telenet.be
To: asterisk-users@lists.digium.com
Subject: Zaptel to Dahdi
Date: Sun, 19 Apr 2009 17:17:39 +0200
VoIP-wiki.org states :
/etc/zaptel.conf Becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf Becomes /etc
one of these files to make chan_dahdi.conf interact with
zaptel.conf (zaptel kernel module) in stead of the dahdi-linux kernel
modules ??
Greetingz,
Jonas.
On Mon, 2009-04-20 at 15:57 +0300, Tzafrir Cohen wrote:
On Sun, Apr 19, 2009 at 05:17:38PM +0200, jonas kellens wrote:
VoIP-wiki.org
Even if Zaptel is compiled, you can also compile Dahdi because Asterisk
will choose the DAHDI-module... it seems.
So I left Zaptel... and compiled Dahdi (everything went well, I followed
the steps) en then Asterisk again (with dahdi support!!).
Yet another episode in this nightmare :
: 03)
3 channels to configure.
[r...@asterisk asterisk]# /usr/sbin/dahdi_hardware
pci::04:05.0 wctdm24xxp+ d161:8005 Wildcard TDM410P
[r...@asterisk asterisk]# /usr/sbin/dahdi_tool
shows me an 'OK' under Alarm for my Wildcard TDM410
Jonas Kellens
part of extensions.conf:
exten = 11,1,Answer()
exten = 11,n,NoOp(CallerID : ${CALLERID(all)})
exten = 11,n,Playback(/tmp/welkom-tcs.alaw)
exten = 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten = 11,n,NoOp(Oproep tijdens
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
allow=ulaw
language=be
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=no
callerid=Jonas Kellens 52
qualify=yes
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO
modules.
When I plug one PSTN-line into a FXO-port I am able to receive calls on
this line and I can also make calls from an internal SIP-phone to the
external PSTN-network.
Still I am bothered about something that appears on the
According to my IAX-provider, an account has been created for me on
their Asterisk-server...
But the Asterisk CLI tells me this :
asterisk*CLI iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
[Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring
bindport on reload
[Apr
I have connected my Asterisk-box directly to my internetconnection. I
have disabled my firewall.
Still I am unable to register with my IAX-provider. Can someone please
point me out why I am unable to register my Asterisk to another
Asterisk-box ?
A RegReq is send to the other Asterisk-box but no
Thanks for the feedback !
I know the IP-address of my Asterisk-server.
The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1).
I have port 4569 forwarded on my NAT/firewall.
Strangely I have the same 'notice' when being attached directly to the internet
(so no firewall in between).
Gavin,
My Asterisk-server has 2 interfaces :
- eth0 = LAN-interface (for SIP-phones to register)
- eth1 = WAN-interface (for IAX-trunking to IAX-provider)
Asterisk is behind NAT (has internal IP-address 192.168.3.248 for WAN_if)
SETUP :
m0n0wall 192.168.3.250 -- 192.168.3.248
-mail,pager,options
50 = 4569,Jonas Kellens,jonas.kell...@thecomputerstore.be,,tz=belgie|
attach=yes
But I do not receive an e-mail after having left a voicemail message on
the voicemailbox 50.
What mail-server does Asterisk uses to send his mail ???
Sendmail is not active on my CentOS-box. I have
Dave,
can you help me with my configuration of mutt (MUA) + msmtp (MTA) ?
I have included the following in my voicemail.conf :
mailcmd=/usr/sbin/mutt
But how will Asterisk know how to use Mutt to attach its
voicemail-message (.wav-file) ???
I use Mutt together with msmtp to send me weekly the
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
I have the same problem with Asterisk 1.4.24 and a Grandstream GXP2020
SIP-phone.
I want to park a call by pressing the 'TRANSFER' and then 90. My parking
lots are from 91 till 95.
The call is parked at extension 91, but the parking lot '91' is not
announced by Asterisk...
I have tried to park
I have changed the features.conf file, yes.
And I put this in my extensions.conf :
include = parkedcalls
Is it better to put exten = 90,1,park() into my dialplan ?
Greetingz,
Jonas.
On Thu, 2009-05-14 at 16:08 -0500, Danny Nicholas wrote:
Did you change 700 to 90 in features.conf? I’d put
I call the firm from my portable at home (zoiper softphone). I have
internal extension 60, and I call the internal SIP-client 10 at the firm
via an IAX-connection over internet.
My colleague at phone 10 answers my call. I ask him to transfer me with
my colleague at extension 50. He then presses
. wrote:
mutt will not deliver a email message, so you are using the wrong
command. The email message with attachment is created by Asterisk and
needs msmtp to deliver the message.
On Sun, May 10, 2009 at 9:10 AM, jonas kellens jonas.kell...@telenet.be
wrote:
Dave,
can you help me with my
Today I get the remark that a call got disconnected after 10 minutes.
This what my VERBOSE-logfile tells me :
[May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing
[00493516...@intern:1] NoOp(SIP/51-b76023b8, Gesprek naar GSM-nummer
via Telenet) in new stack
[May 18 15:36:30] VERBOSE[3940]
Check out the Grandstream GXP-serie also...
http://www.grandstream.com/gxp2020.html
You can program the line buttons to support BLF (red, red blinking, green)
- Oorspronkelijk bericht -
Van
: Olivier [mailto:oza-4...@myamail.com]
Verzonden
: dinsdag
, mei
19, 2009 08:21 AM
Aan
:
To feed your curiosity... I'm about to implement it.
I have several GXP2020 and GXP1200 Grandstream telephones. I'm reading
documentation to know how to start and what to expect.
I'm hoping that implementing BLF on these Grandstreams in combination with
Asterisk is easier then configuring
Gordon,
have you not defined a context [BLF_group] in your extensions.conf ??
And a subscribecontext in sip.conf ?
The Grandstream documentation does mention this.
Have you configured the speed dial buttons (to the right of your grandstream)
or the phone line buttons (to the left of the
Hey there list !
I'm receiving negative feedback when people try to pickup another
ringing phone by pressing *8 on there own Grandstream device.
These are my setting that should make pickup possible :
all my sip-clients (Grandstream) have this in their config (sip.conf) :
callgroup=1
-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
belgie=Europe/Brussels|'vm-received' Q 'digits/at' R
[Voicemail-context]
60 = 4569,Jonas Kellens,jonas.kell...@telenet.be
In my extensions.conf I has the following :
exten = 2000,1
jonas kellens wrote:
My /root/.msmtprc-file has the following :
# Set default values for all following accounts.
defaults
logfile ~/.msmtp.log
There is NO entry in the logfile of msmtp (/root/.msmtp.log). No error,
no success.
Is Asterisk running as root or as the asterisk user
David,
what is your SMTP-client then ?
Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it
still /usr/sbin/sendmail ??
I use version 1.4.24.
Thanks for your reply.
Greetingz,
Jonas.
On Fri, 2009-05-22 at 10:59 -0400, David wrote:
-BEGIN PGP SIGNED MESSAGE-
I thought that /var/log/maillog was for sendmail ?? I'm not using
sendmail...
My /var/log/maillog is empty :
[r...@asterisk ~]# cat /var/log/maillog
[r...@asterisk ~]#
How about the system()-application ?? Why is that also not working for
me ??
On Fri, 2009-05-22 at 16:25 +0100, Geraint Lee
Hey list !
I'm getting the feedback of a customer that a conversation is like half
duplex : when he talks, the other end of the call is no longer heard.
What could be the cause of these drop-outs ?
A call that is coming in from the PSTN is routed through an IVR-system
to the correct internal
I have posted a similar problem earlier on this mailing list with my
Asterisk-system + TDM410 + Grandstream telephones.
But there has not yet been a response to this.
My client is also experiencing a 'simplex' conversation. There seems
that audio can only flow 1 one way at the same time.
What I
On my TDM410P pci-card I have an hardware echo cancellation module
(Digium VPMADT032 EC Modul).
I have set 'echocancel=yes' in my chan_dahdi.conf to activate this
hardware module.
Do I now have 2 echo cancellers that are activated ? A software echo
canceller and a hardware echo canceller ??
Form
On Sat, 2009-05-30 at 23:15 -0300, Daniel Bareiro wrote:
I was testing calling from my cell phone to an analog telephone and if the
other person hangs before I do it, I see that in the my cell phone the call
even continues persisting so that if the person of the other endpoint take the
How about this :
if you add the option 'g' in your Dial()-command, then when the caller
hangs up Asterisk will continue to execute the commands hat follow.
You could then read the ${DIALSTATUS}-variable (which will be 'CANCEL')
and execute a command based on this.
On Fri, 2009-06-12 at 23:58 +1000, Alex Samad wrote:
what is the best way forward to recompile with hardware echo canceller
support.
No need to do anything special during compilation. For hardware echo
cancellation just put the option echocancel=yes in chan_dahdi.conf
There are some things that are not that clear to me :
When I want to write CDR-info to an external MySQL-DB
- do I need to install the asterisk-addons prior to installing Asterisk
or after having installed Asterisk ??
- How do I tell Asterisk not to write CDR-info to the Master.csv file
but into
On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote:
In modules.conf: noload = cdr_csv.so
Are there other modules I need to load or unload ??
asterisk*CLI module show like cdr
Module Description
Use Count
cdr_addon_mysql.so MySQL CDR Backend
0
dstchannel lastapp lastdatastart answer end duration
billsec disposition amaflags
Why does it want to write to a column calldate ?? Where is this
defined ??
Thanks for the help !
Jonas.
On Fri, 2009-06-19 at 14:13 -0500, Miguel Molina wrote:
jonas kellens
Thanks for your reply. I saw that info also on voip-info.org.
I was wondering if I could define other columns, like those used for
billing (as defined in my sip.conf).
Jonas.
On Tue, 2009-06-23 at 09:22 +0100, Ishfaq Malik wrote:
Hi
The calldate column is the date and time of the call,
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it
to in the table, most notably
accountcode and userfield. There is more info here.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
I'm not sure about defining additional columns and writing to them
through the dialplan but I don't think you can.
Ish
jonas kellens wrote
-- Executing [0473775...@intern:1] NoOp(SIP/twinkle-088e6ea8,
conversation to GSM) in new stack
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-088e6ea8,
SIP/3starsnet/0473775006) in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 Loop Detected back from
Do you understand what is happening ?
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-08de0490,
SIP/3starsnet/0473775006) in new stack
-- Called 3starsnet/0473775006
-- SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
-- Got SIP response 500 Service
-0400, Steve Totaro wrote:
On Tue, Jun 23, 2009 at 8:09 AM, jonas kellens
jonas.kell...@telenet.be wrote:
-- Executing [0473775...@intern:1]
NoOp(SIP/twinkle-088e6ea8, conversation to GSM) in new
stack
-- Executing [0473775...@intern:2
, channel 'SIP/twinkle-0a0567f8' status is
'CONGESTION'
Really destroying SIP dialog
'340811e66bc43ba36fb5d507066fc...@192.168.2.2' Method: INVITE
Really destroying SIP dialog 'xfdsxekzwoxc...@localhost' Method: ACK
Jonas.
On Wed, 2009-06-24 at 02:47 +1000, Rob Hillis wrote:
jonas kellens wrote
:13 -0400, Steve Totaro wrote:
On Tue, Jun 23, 2009 at 9:36 AM, jonas kellens
jonas.kell...@telenet.be wrote:
Do you understand what is happening ?
-- Executing [0473775...@intern:2]
Dial(SIP/twinkle-08de0490, SIP/3starsnet/0473775006
I want to use JabberSend in my dialplan, but I saw that my Asterisk does
not support Jabber.
Also I have nowhere a module res_jabber.so...
So I thought I'd rebuild my Asterisk. In menuselect I saw that
res_jabber was dependent of 'iksemel' and 'gnutls'.
In my yum repositories I can find a
I wonder if there is a GUI that does not change the underlying hand-made
configuration ?!
What I'm looking for actually is a GUI for adding a new SIP-client +
voicemail, so that a company does not have to call me when they hired a
new employee.
I don't want a GUI that over-writes my hand-made
24, 2009 at 09:20:44PM +0200, jonas kellens wrote:
I wonder if there is a GUI that does not change the underlying hand-made
configuration ?!
What I'm looking for actually is a GUI for adding a new SIP-client +
voicemail, so that a company does not have to call me when they hired a
new
I feel a great preference for sticking to manually editing
the .conf-files. But if I define in the contract that changes to the
Asterisk-PBX need to be done by me, I force a maintenance cost towards
the customer and that is not always what is requested...
On Thu, 2009-06-25 at 15:26 +0530,
know PHP + MySQL...
Will have to do some studying then...
On Thu, 2009-06-25 at 11:40 +, Jeff LaCoursiere wrote:
On Thu, 25 Jun 2009, jonas kellens wrote:
I feel a great preference for sticking to manually editing
the .conf-files.
Then why did you ask for a GUI?
But if I
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:x...@85.119.188.3
[3starsnet]
type=peer
host=85.119.188.3
username=092779077
secret=
fromuser=092779077
, jonas kellens wrote:
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 ??
___
-- Bandwidth and Colocation Provided by http://www.api
asterisk*CLI sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
[Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889
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:x...@85.119.188.3
;
I was trying to enable CDR in a mysql-database when the following
occured :
asterisk*CLI cdr status
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: yes
CDR registered backend: cdr_manager
CDR registered backend: cdr-custom
asterisk*CLI exit
[Jun 29 21:56:52] Executing last
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:x...@85.119.188.3
This the output of SIP show peers :
Actually it was my Firewall (Endian). By rebooting my firewall, all
problems were solved and till this moment every communication succeeds.
I do expect them back...
I don't want to hijack this Asterisk-mailinglist, but I think that
firewall-issues also are related to Asterisk-support.
So my
Why do I see many setups where there is an Asterisk server in
combination with a SIP-server like OpenSER or Kamailio ?
Isn't Asterisk enough as SIP-server ??
It can communicate with many databases through ODBC, with many other
software through an API (AGI), with other servers like OpenFire for
On Fri, 2009-07-03 at 11:58 +0100, Mike wrote:
tempest:~# lspci
00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
I don't think this is you TDM-card...
This is mine :
04:05.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)
Subsystem: Digium, Inc.
I am trying to connect Asterisk and Openfire together, but it's not yet
working completely...
I don't know for sure if my manager.conf-file is set correctly. I use
this manager.conf file just to let Openfire talk to Asterisk...
[general]
displaysystemname = yes
enabled = yes
webenabled = yes
Every morning I check my SIP registry to the SIP-provider. And I must
conclude that during the night somewhere registry has failed.
asterisk*CLI sip show registry
HostUsername Refresh State
Reg.Time
85.119.188.3:5060 092779077 105
Thanks for your reply, Steve.
My firewall is a 3-NIC pc with Endian Community installed.
Jonas.
On Mon, 2009-07-06 at 09:49 +0100, Steve Howes wrote:
On 6 Jul 2009, at 09:37, jonas kellens wrote:
What could be failing ?? Is this a NAT issue of some kind ? Could it
be that my firewall
Hi Steve,
I have qualify for all the peers that are defined, and so also for the
peer I have defined for my SIP-provider.
What you see below is such qualify :
[Jul 6 11:38:26] Reliably Transmitting (NAT) to 85.119.188.3:5060:
OPTIONS sip:85.119.188.3 SIP/2.0
Via: SIP/2.0/UDP
Registration timed out... This time a 'sip reload' doesn't help (why
would it?).
Registration uses no NAT, and a SIP option uses NAT...
Verbosity is at least 25
[Jul 6 11:58:09] -- Remote UNIX connection
[Jul 6 11:58:39] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:--
Registration
I have installed gnutls and gnutls-devel from RedHat repositories
[r...@asterisk asterisk]# yum install gnutls gnutls-devel
I have installed iksemel with gnutls support :
[r...@asterisk asterisk]# cd /usr/src/iksemel-1.3/
[r...@asterisk asterisk]# ./configure --with-gnutls --prefix=/usr
On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote:
I can assure you that it works, and that it works well. We use it ;)
My jabber.conf :
[general]
debug=yes ;;Turn on debugging by default.
autoprune=no;;Auto remove users
Yet another night has passed...
This morning :
Verbosity is at least 25
asterisk*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
Realtime
twinkle-candy/twinkle-can (Unspecified)D 0
UNKNOWN
twinkle-jonas/twinkle-jon
[Jul 7 11:54:38]
JABBER: asterisk INCOMING: iq type=error id=a
from=192.168.2.5 to=openfire.jocan.local/e7beae90query
xmlns=jabber:iq:authusernameasterisk/usernameresourceasterisk/resourcedigest90a141d72ee469dc30bc95c661d0c299ead11061/digest/queryerror
code=400 type=modifybad-request
This is my jabber.conf :
[general]
debug=yes ;;Turn on debugging by default.
;autoprune=no ;;Auto remove users from buddy
list.
;autoregister=yes ;;Auto register users from buddy
list.
[asterisk]
To calculate the monthly data traffic that is generated by VoIP-calls,
is it as simpel as
80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month =
585.9375 MB traffic / month
???
Jonas.
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Same question here : How about in Belgium ??
Because core show application like sms gives information about the UK.
Jonas.
On Thu, 2009-07-09 at 11:26 +0200, ESGLinux wrote:
Hi all,
I´m a beginner with asterisk and I want to know if with asterisk I can
send sms to a mobile, I´m on
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable ?
I guess it is impossible for Asterisk to directly contact the private
Thanks Alex for your explanation.
Does this NAT-mapping means that TAPI would also be possible ??
Jonas.
On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote:
Yes, this problem has a solution. The NAT gateway creates a UDP state
mapping between internal source ports and external source
at 06:33 -0400, Alex Balashov wrote:
jonas kellens wrote:
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable
Asterisk
send an sms through the IP-connection with the VoIP-provider ??
Greetingz,
Jonas Kellens.
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Try putting exten = 8001187e0,1,Dial(SIP/263) in the [default]-context.
I have the same issue. Apparently your SIP-provider send calls to your
Asterisk-box from multiple IP's so that Asterisk cannot match the
inbound call on source IP and therefore sends it to the default-context.
Jonas.
On
Would it be possible to execute some kind of script when for example
Asterisk restarts... or stops... ?
How can one read the status of Asterisk so that when the service is
stopped I could be notified by mail, by text message,... ?
I don't know how to read the status of Asterisk (or the change of
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
-
(according to the info of
my Hosting provider).
Don't really know what info I can provide you with to help me find and
install the needed sources...
Jonas.
On Tue, 2009-08-18 at 15:16 +0300, Tzafrir Cohen wrote:
On Tue, Aug 18, 2009 at 02:07:39PM +0200, jonas kellens wrote:
I want to install
: Could one use different family-names in the
Asterisk Realtime Architecture ??
Greetingz,
Jonas Kellens.
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