@lists.digium.com
Date: Mon, 13 Apr 2009 23:10:13 +0200
CC: asterisk@sedwards.com
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using
Asterisk (update)
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:5060
Sent: Monday, April 13, 2009 1:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls
using Asterisk
Barry,
there is a 'send' button but pushing it before or after dialing '211' does
not really change anything...
I get no dial tone
@lists.digium.com
Subject: [asterisk-users] Asterisk-beginner : cannot make phonecalls using
Asterisk
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
just for a test, run service iptables stop as root on the asterisk
server and then reboot your phones. after that, try again and see if
the phones are making communications with asterisk.
you can turn the firewall back on with service iptables start
jonas kellens wrote:
Hi there,
this is
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
kellens
Sent: den 14 april 2009 20:04
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls
using Asterisk
There is something wrong with my IPtables !!!
When i do :
service iptables stop
I see my phones register
On Tuesday 14 April 2009 13:04:02 jonas kellens wrote:
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited
-A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
For one, these two rules are reversed. For two, you've failed to create holes
in your firewall for
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
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 first
internal communication with SIP.
Thought it would go
bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this may be
your issue.
Registration is only for receiving calls, if you are not seeing information on
the dial, then the phone is not talking to the server. I would make sure of
the settings in the
Mike,
thank you for your reply.
However I do not have the option of a DHCP-server. On the network where
Asterisk needs to be implemented all is configured statically, so also
the IP-phones need to be statically assigned an IP-address. Surely this
can not be thé problem...
Greetingz,
Jonas.
On
On Mon, 13 Apr 2009, Anthony Plack wrote:
bindaddr = 0.0.0.0
I would set this to the ethernet interface IP address, I believe this
may be your issue.
Binding to 0.0.0.0 means listen to all IP addresses on the box. It is
not the issue.
Thanks in advance,
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
-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using
Asterisk
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
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,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: April-13-09 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk-beginner : cannot make
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
-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 to Asterisk 1.4.24 currently running on asterisk (pid = 3895)/
/Verbosity is at least 5/
/asterisk*CLI /
Nothing is displayed... it
Barry,
there is a 'send' button but pushing it before or after dialing '211'
does not really change anything...
I get no dial tone, no ring tone on the other phone and no output on the
Asterisk CLI...
I thought this would go easier... Don't know what is going on here.
I followed the book
jonas kellens wrote:
Hi there,
I notice (on the Asterisk CLI) that my SIP-phones do not register.
They have a fixed IP and there account information is
If your phones don't register, then your not going to be able to make a
call.
The Grandstream phones have a web interface (At least if
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
Hi
On Mon, Apr 13, 2009 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
On 13 Apr 2009, at 20:52, jonas kellens wrote:
Hey there again !
If you are new to all this wouldn't going with some pre-made dialplan
be useful? Go for something like FreePBX
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Mon, 13 Apr 2009, 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
jonas kellens escribió:
Hey there again !
Hey, just my two cents:
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/
Hi Tzafrir,
yet with the first test, things get wrong :
asterisk*CLI logger show channels
Channel Type StatusConfiguration
--- ---
/var/log/asterisk/messages File Enabled- Warning
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 to the Asterisk server.
You
On Mon, Apr 13, 2009 at 10:39:49PM +0200, jonas kellens wrote:
Hi Tzafrir,
yet with the first test, things get wrong :
asterisk*CLI logger show channels
Channel Type StatusConfiguration
--- --
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full protocol
decode
listening on eth1, link-type EN10MB
On Mon, 13 Apr 2009, jonas kellens wrote:
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full protocol
decode
listening on eth1, link-type EN10MB (Ethernet), capture size 96 bytes
23:04:59.522498 IP 192.168.4.114.sip
2009/4/14 jonas kellens jonas.kell...@telenet.be
[r...@asterisk asterisk]# netstat -a -n -p | grep 5060
udp0 0 0.0.0.0:50600.0.0.0:*
3047/asterisk
[r...@asterisk asterisk]# /usr/sbin/tcpdump port 5060
tcpdump: verbose output suppressed, use -v or -vv for full
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