Hello list,
using asterisk 1.4.30
I have the strangest problem that some SIP accounts can register to my
Asterisk and others not. I see no connection between all those that can
register or all those that can't.
It's not a firewall problem as all register to port 5060 and the range
5060 --
On Wed, 23 Jun 2010, Steve Edwards wrote:
On Wed, 23 Jun 2010, Gordon Henderson wrote:
Ouch. 82.0.0.0/8 is on my block list, available at:
http://www.sedwards.com/class-a-block-list
If you don't need to receive packets from far away places, it's a great
start.
I'd like to have a
Hello Bruce,
This module is not reliable on FreePBX?
You know if there is a open source web-voicemail for Asterisk?
Best regards,
Mickael.
2010/6/23 bruce bruce bruceb...@gmail.com
It's one of the bad modules that goes with FreePBX anyhow. The moment you
go over 3000 recordings you are
I'm just about to start experimenting with realtime queues, so can't
offer anything from my own experience, but what happens if instead of
updating the existing row, you delete it and insert a new one for the new
user ?
ie. DELETE FROM queue_member_table WHERE id=1;
INSERT INTO
On Wed, Jun 23, 2010 at 04:53:27PM -0400, Zeeshan Zakaria wrote:
Zaptel and dahdi is the same thing, except the later one is weirdly named to
make it harder to pronounce. Don't worry to upgrade to dahdi. But it is not
plug and play and you'll need to configure /etc/zaptel.conf and
On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote:
On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote:
Hi all,
Anyone know why this happens?
Mem:524288k total, 508120k used,16168k free,0k buffers
Swap:0k total,0k used,0k
Jonas,
have you checked the log files?
Giorgio
Jonas Kellens wrote:
Hello list,
using asterisk 1.4.30
I have the strangest problem that some SIP accounts can register to my
Asterisk and others not. I see no connection between all those that
can register or all those that can't.
It's
On Tuesday 22 Jun 2010, Mike wrote:
Hi,
I have the following happen to me after the restart of one of my servers:
out of my 3 PRIs (all configured with the same technical settings), the
last one isn't coming back. It's underutilized (chances it didn't get a
call since my reboot), if it
Hi all,
I'm building a karaoke service. Asterisk will play a music file, people can
detect the point when they want to sing and record by press * key during the
music is playing, and press # key to stop recording.
I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and
True, what I meant was they serve the same purpose, i.e. drivers for the
non-SIP hardware.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-24 4:52 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Jun 23, 2010 at 04:53:27PM -0400, Zeeshan Zakaria wrote:
Zaptel and dahdi is the
Without submitting some logs or your sip settings, how could somebody on
this list help you?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-24 6:28 AM, Giorgio Incantalupo gincantal...@fgasoftware.com
wrote:
Jonas,
have you checked the log files?
Giorgio
Jonas Kellens wrote:
Hello
Giorgio,
there is just no registration coming in. SIP debug shows nothing on the
SIP peers that do not register. TCPdump shows nothing on incoming
registrations. Firewall is down.
I'm using SIP realtime and I'm starting to think it is a problem with
the MySQL-DB of the MySQL-driver or
What type of phone you are using? It is possible that # is used by this
phone as one of its internal functions.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-24 6:41 AM, huu giang huugiang...@yahoo.com wrote:
Hi all,
I'm building a karaoke service. Asterisk will play a music file,
If tcpdump not showing any incoming packets from the phones, then the phones
are not communicating with the server. There could be no other reason other
than iptables blocking that communication.
Did you check if iptables is blocking anything?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On
Hi,
Got some great news a few days ago from Sandro Gauci (@SandroGauci)
and we'll be talking about this with him this Friday at 1PM.
SIPVicious, the free security tools for SIP scanning, now include a
new tool: svcrash. It is aimed at helping system administrators stop
bandwidth consuming scans
Awesome!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
As I said earlier, the firewall was down. I'm using LFDCSF.
If it had anything to do with the firewall, how can the problem be
resolved by recompiling asterisk and asterisk-addons ?
Yesterday everything went well, this night something happened I guess,
and this morning on some locations some
On 24 Jun 2010, at 12:49, Jonas Kellens wrote:
It seems as if some SIPaccounts could register and others could not. I don't
think a firewall distinguishes between phone brands or SIP accounts.
Alas 'stabbing in the dark' is all we can do until you actually provide some
information for us.
I checked, it made sense. But it isnt it :-) The fourth span is commented
out, but the third one isnt (and shouldn't be since it's active).
Thanks for the tip though, could have easily been this.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Maybe we can easily extend the tool to crash Asterisk too (using some
exploits non-up2date Asterisk installations) ;-)
Am 24.06.2010 13:36, schrieb Randy R:
Hi,
Got some great news a few days ago from Sandro Gauci (@SandroGauci)
and we'll be talking about this with him this Friday at 1PM.
El 24/06/10 05:05, Tzafrir Cohen escribió:
On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote:
On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote:
Hi all,
Anyone know why this happens?
Mem:524288k total, 508120k used,16168k free,0k buffers
Steve Edwards wrote:
On Sat, 19 Jun 2010, bruce bruce wrote:
Is it possible to harvest the output of system into a SetVar(variable)?
exten = s,n,SetVar(var=system(asterisk -rx sip show channels | grep
-c (ulaw))
??? any problem with the syntax?
) Your parentheses don't match.
) You
Michael wrote:
I am attempting to setup Asterisk to work with Gtalk.
I am using the following versions:
Slackware Linux 12.0
Asterisk 1.6.2.9
GNU TLS 2.8.6
Iksemel (svn v25)
OpenSSL 0.9.8o
It all compiles however about 10 seconds after starting Asterisk it crashes.
If there is any
When I execute a dialplan reload I am getting color commands (that I
never got before)
with the echoed Dialplan reloaded.
The sip reload works as it always has with no color codes echoed.
Its not detrimental just messes up my screen some.
jerry
--
Jerry Geis wrote:
When I execute a dialplan reload I am getting color commands (that I
never got before)
with the echoed Dialplan reloaded.
The sip reload works as it always has with no color codes echoed.
Its not detrimental just messes up my screen some.
jerry
Sorry I meant to add -
I have a system running Asterisk 1.6.2.6 that generates about 80k
calls/day. Calls are fired from Asterisk Manager (async originate -- 60
second timeout). I am capturing 100% of the originate responses
(recorded in DB). About 5% of the calls result in a reason code of '3'
(Remote End is Ringing).
Hello,
I wanted to add the functionality of 3-way conference to my asterisk pbx
using meetme or confbridge. During a call the user should be able to put the
other party on hold and dial another number, then on dialing some key
sequence all three of them enters into a conference. Is it possible to
Hi Guys,
Asterisk 1.6.2.7 install from Yum Repository shows a lot of : doing
dnsmgr_lookup for sip.provider.com
Google searches show it was fixed in some version.
Is this to be ignored?
Thanks
--
_
-- Bandwidth and
On Thu, Jun 24, 2010 at 2:42 AM, Rob Coward r...@jive-videos.net wrote:
I'm just about to start experimenting with realtime queues, so can't offer
anything from my own experience, but what happens if instead of updating the
existing row, you delete it and insert a new one for the new user ?
Hi,
I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to
both a local Asterisk server and also with a trunk directly to
a VOIP provider. Everything works great, except I'm having a problem
setting the outbound caller ID to a value different from the
SIP username/authname.
The SPA8000
Its possible but not easy. Search for n-way conferencing on voip-info.org,
it has all the details on how to do it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-24 2:01 PM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
I wanted to add the functionality of 3-way conference to my asterisk
Hello folks,
I've been trying to get T.38 over SIP working with calls terminated by a
MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually
working perfectly; however, I can't get the TNT to properly terminate a
FAX call. Does anyone have a working configuration for SIP and
Hi,
I know some of you are very experienced as to the working of
networks. I wondered whether there is some accepted way of determining
bandwidth needs based on the network traffic over time. For example,
looking at the figures for the network traffic through the server
interface, we have
ISP 10% rule is what you are asking about
expected that average usage is 10% of total subscribers with bursts higher
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux
http://www.asteriskguru.com/tools/bandwidth_calculator.php
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On
On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Its possible but not easy. Search for n-way conferencing on voip-info.org,
it has all the details on how to do it.
Or you could post the direct link:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
--
Paul
I could, but it depends upon from where I am replying.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-24 4:28 PM, Paul Belanger paul.belan...@polybeacon.com wrote:
On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Its possible but not ...
Or you could post the
I know on my polycom phones, I just press the conf button, dial, and then
hit join, and all done, no special programming required on dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Dear, just a short question:
If I use G.711a and G.711b codecs between the Portech GSM Gateway and
Asterisk 1.4.23, what DTMF mode is better to use in both sides if a
mobile phone call the GSM Gateway in order to contact an internal IP
extension (Mobile to LAN scenario):
RFC2238
Inband
SIP INFO
Hi
Almost phones I used meet this problem, they are
- Nokia 1200
- Nokia 6210
- Nokia E72.
When I used Softphone to test on IP (SIP), some softphones meet similar problem
(twinkle), some don't meet (Xlite, Kapanga).
Very Thanks
Giang
From:
Hi,
When I require user enter a code and end wich # key, for example 1234#,
Asterisk can detect # key and detect the code people just enter.
Thanks
From: Zeeshan Zakaria zisha...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Nokia cell phones is a totally different case altogether because it used
inband dtmf. I can't make any guess on this.
As for the sip soft phones, for twinkle, there must be something in its
settings which is using # for something else. You mentioned Xlite works
fine. This shows that nothing is
We are an asterisk user... small time system 50-100 users or so.
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can be
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