sean darcy a écrit :
[...]
Context names cannot be duplicated, unless you suffix them with (+) to
allow them to be added together. It does not matter whether it is the
'global' context or any other context.
Well
Dialplan reloaded.
== Parsing '/etc/asterisk/extensions.conf': ==
11 feb 2010 kl. 08.49 skrev Ron Arts:
Op 11-02-10 03:42, sean darcy schreef:
Kevin P. Fleming wrote:
sean darcy wrote:
I found out that the [globals] section in extensions.conf is ignored if
an #include 'd file has a [globals] section. Is this intended?
In this particular case, the
Hi
We use Snom phones pretty exclusively and find them to be really good.
So long as you get the firmware version up to 7.3.28 you shouldn't have
any issues.
They are really good for remote configuring and management and also have
a really nice web interface for the user to use if the phones
On the 'used' market I can't really fault the Swissvoice 1P10S (The SIP
version). Web Browser/Phone configuration, simple, works flawlessly with
Asterisk, can take a headset and around £10-£20 used.
--
_
-- Bandwidth and
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
[general]
rtpstart=3
rtpend=30100
so 100 ports available. I know that up to 4 ports per
Hi!
On the 'used' market I can't really fault the Swissvoice 1P10S (The
SIP version). Web Browser/Phone configuration, simple, works flawlessly
with Asterisk, can take a headset and around £10-£20 used.
Agreed - but do note that this is a single line phone (correct me if I am
wrong). And
Hi!
My rtp.conf has
[general]
rtpstart=3
rtpend=30100
My understanding is that rtpstart should be an odd number, and rtpend an
even one.
so 100 ports available. I know that up to 4 ports per channel can be used
and so up to 25 channels are possible. But even earlier I often get
On Thu, Feb 11, 2010 at 06:20:54PM +1100, Lee, John (Sydney) wrote:
Just to share some experience with everyone about what happened today to
our Asterisk 1.4 box with Digium TE412P card.
We had an unscheduled power outage which shut down the Asterisk box.
When the power went up, Asterisk
On Wed, 10 Feb 2010, Jeff LaCoursiere wrote:
I haven't used any standard Grandstream IP phones, but I am *trying* to
stabalize the new video phones they have come up with. I have several
GXV3000 and GXV3140s. I got through central provisioning using their java
based tool and for the most
On Wed, Feb 10, 2010 at 09:53:46PM -0600, Lyle Giese wrote:
Warren Selby wrote:
On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Here's a start for you, just run from cron once a day:
Lyle
So basically, nothing built into
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Feb 10, 2010 at 09:53:46PM -0600, Lyle Giese wrote:
Warren Selby wrote:
On Tue, Feb 9, 2010 at 5:54 PM, Lyle Giese l...@lcrcomputer.net
mailto:l...@lcrcomputer.net wrote:
Here's a start for you, just run from cron once
Hi all,
I have a lot of call in wich I found that my Asterisk doesn't answer the BYE
message, then the BYEs are retransmitted, but the call ends, when the
Asterisk sends a BYE.
Time AS.TE.RI.SK
CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From:
I have always found that the marginal extra cost of a Polycom, Snom or
Cisco phone is very, very well worth it in the long run versus the
giant catastrophe that is Grandstream and other low-budget hardware.
The customers seem to agree.
Budget-conscious doesn't have to mean masochistic.
--
das sandesh wrote:
Hi,
Asterisk got stopped this morning after 20 minutes and phones went to
'No Service' and then got started automatically after 20 min, as I could
see in the full log that asterisk got started at so and so time:
[Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
[general]
rtpstart=3
rtpend=30100
Jason Parker wrote:
Brian wrote:
Each time the server is rebooted Asterisk duly
deletes the manually created /var/run/asterisk directory - quite why it
does this I just don't know - perhaps it is a bug?
Your assumption is incorrect. Some Linux distributions will empty /var/run/
on
On Thu, 11 Feb 2010, Alex Balashov wrote:
I have always found that the marginal extra cost of a Polycom, Snom or
Cisco phone is very, very well worth it in the long run versus the
giant catastrophe that is Grandstream and other low-budget hardware.
The customers seem to agree.
SPA504G seems to be a good choice since I don't want to have any issues and
this still have the same provisioning scheme I'm working right now.
Also I like the HD feature.. what is the bandwith consumption for that
feature? It will work only between 2 spa504 right? or any G722 device can
Hello,
I have the following situation: A firewall is blocking all SIP and RTP
traffic in the side of some of my clients. My clients cannot change settings
of the firewall.
I need to solve this problem and I need some help from you.
I have this idea: implement a SIP user agent which does
Hello,
I have the following situation: A firewall is blocking all SIP and RTP
traffic in the side of some of my clients. My clients cannot change settings
of the firewall.
I need to solve this problem and I need some help from you.
I have this idea: implement a SIP user agent which does
On Tue, 2010-02-09 at 11:26 +0100, Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I have searched a bit for information regarding the status on the
dialplan in lua (pbx_lua.so). I know that 'hint' won't work and has to
be put in the regular
Sebastian Milioto wrote:
Hi all,
I have to install 25 IP Phone in some building. I want just basic IP
Phones like:
Cisco-Linksys SPA922 u$s 146
Grandstream GXP-2000 u$s 105
Snom 300 u$s 119
The most valuables parameters for me are (in importance
Have you considered using IAX instead of SIP? IAX2 is a VoIP protocol that
carries both signaling and media on the same port:
http://en.wikipedia.org/wiki/Inter-Asterisk_eXchange
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
running freepbx 2.6.1 and asterisk 1.6.0.21
i did a clean install with the versions listed above. I was on freepbx 2.5.0
and asterisk 1.6.0 ( i think)
I have aastra 57i phones. with the old versions i could hit the dnd(*76) button
and i would hear dnd activated and the the light at the top
You didn't state what kind of computer the TE412P is in, but IME, the first
thing to do if you have a hardware problem after a power bounce is to
shutdown everything, power it off, wait 30 seconds, then turn it back on
normally. Sorry you lost the day of usage.
--
Danny Nicholas
--
-Original
try Yealink, this is damn good phone and we distribute in India.
Rgards
MT Kondela
www.kevesystems.com
On 2/10/10, Sebastian Milioto smili...@gmail.com wrote:
Hi all,
I have to install 25 IP Phone in some building. I want just basic IP Phones
like:
Cisco-Linksys SPA922 u$s 146
I've scoured the web for hints, and find a lot of chatter about one-way
audio with IP Kall, but no definitive explanation. I have the default range
(5000-31000) of UDP RTP ports forwarded right to my asterisk box, and have
no other difficulties with one-way audio on any other peers. Does anyone
See at:
1) openvpn / ipsec tunnels
2) IAX protocol
Firewall defines the report not on ports, and traffic contents. Change
of ports will not help
hope it helps..
On Thu, 2010-02-11 at 14:37 +0100, mosbah.abdelkader wrote:
Hello,
I have the following situation: A firewall is blocking
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
On Thu, Feb 11, 2010 at 6:37 AM, mosbah.abdelkader
mosbah.abdelka...@gmail.com wrote:
Hello,
I have the following situation: A firewall is blocking all SIP and RTP
traffic in the side of some of my clients. My clients cannot change settings
of the firewall.
I need to solve this problem and
Thank you Jamie for your good reply.
It is a very good idea to hava the media and control transported over the
same port with IAX protocol.
The difficulty is in that the port is not well known by the network admins.
It is usually blocked.
My idea is to use a well know port like port 80 (that
The Asterisk Development Team has announced the release of Asterisk 1.2.39.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Asterisk 1.2.38 was created, but not released, to resolve two regression fixes
caused by security updates. Prior to
Problem is that the port 80 you are talking about is a TCP port. Voip
(iax and rtp) use UDP
On 2/11/10, mosbah.abdelkader mosbah.abdelka...@gmail.com wrote:
Thank you Jamie for your good reply.
It is a very good idea to hava the media and control transported over the
same port with IAX
hi
anyone experience with that and maybe asterisk / switchvox?
thx
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On 9/02/10 12:59 PM, Tilghman Lesher wrote:
add to the top of /etc/resolv.conf
nameserver 127.0.0.1
If you're using DHCP on any of your interfaces, you'll need to configure
dhclient (or whatever dhcp client you're using) to prepend in the
configuration with (e.g. /etc/dhcp3/dhclient.conf):
Olle E. Johansson wrote:
11 feb 2010 kl. 08.49 skrev Ron Arts:
Op 11-02-10 03:42, sean darcy schreef:
Kevin P. Fleming wrote:
sean darcy wrote:
I found out that the [globals] section in extensions.conf is ignored if
an #include 'd file has a [globals] section. Is this intended?
In this
At 6:48 PM on 11 Feb 2010, sean darcy wrote:
OK, now clear on suffix v. prefix ( Doh! ) and having RTFM,
I have extensions.conf:
[general]
#include exts/gvoice.exten.conf
[...]
[globals]
pstnline = DAHDI/4
...
and exts/gvoice.exten.conf:
[globals](+)
sean darcy wrote:
OK, now clear on suffix v. prefix ( Doh! ) and having RTFM,
I have extensions.conf:
[general]
#include exts/gvoice.exten.conf
static=yes
writeprotect=no
autofallthrough=yes
[globals]
pstnline = DAHDI/4
...
and
You didn't state what kind of computer the TE412P is in
It was a DELL PE2950.
the first
thing to do if you have a hardware problem after a power bounce is to
shutdown everything, power it off, wait 30 seconds, then turn it back
on
normally.
You could be right.
I think this is what someone
What is the output of 'cat /proc/dahdi/1' ?
I did not record it but it just shows every channel as 'red alarm'.
What do you have in /etc/zaptel.conf ?
loadzone=au
defaultzone=au
#
# For OnRamp 10
#
span=1,1,0,ccs,hdb3,crc4
bchan=1-10
unused=11-15,17-31
dchan=16
#
# Rhino 24-port Channel Bank
#
Hi,
I am breaking my fingers in configuring an asterisk (1.6) to
successfully transmit audio with the following setup:
asterisk, resides in local network, ip is 10.26.208.252
versatel business router (directly connected to a dsl, configured by
sip-provider), WAN ip 89.244.13.25
versatel
Here's part of the output of running an AGI file:
-- Playing 'degrees' (escape_digits=) (sample_offset 0)
-- Playing 'fahrenheit' (escape_digits=) (sample_offset 0)
-- Playing 'wx/humidity' (escape_digits=) (sample_offset 0)
-- DAHDI/1-1 Playing 'digits/40.ulaw' (language 'en')
Kevin P. Fleming wrote:
sean darcy wrote:
OK, now clear on suffix v. prefix ( Doh! ) and having RTFM,
I have extensions.conf:
[general]
#include exts/gvoice.exten.conf
static=yes
writeprotect=no
autofallthrough=yes
[globals]
pstnline = DAHDI/4
...
and
From a technical point UDP and TCP ports are separate, a server
listening for TCP requests on port 80 wont see any UDP traffic on that
port unless it explicitly opens a UDP socket. Tunneling in on UDP port
80 might be possible if the routing rules that are in place dont
specify to allow only TCP
On Thu, 11 Feb 2010, Richard Kenner wrote:
Here's part of the output of running an AGI file:
[snip]
-- DAHDI/1-1 Playing 'digits/9.ulaw' (language 'en')
-- Playing 'percent' (escape_digits=) (sample_offset 0)
-- Playing 'wx/winds' (escape_digits=) (sample_offset 0)
-- DAHDI/1-1
Hi, I was looking for a value buy g722 IP Phone available in India for
experimental use.
It will be great if someone can provide me with inputs of any supplier in
the region.
Thanks,
Vineet
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