If you are using linux firewall, try this, it was very usefull to me:
iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to
ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to
iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FOR
On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson wrote:
> The server is not behind NAT only the client above is
>
Sounds like a phone (not asterisk) issue then, make sure you have
setup your NAT and port forwarding properly on the client side.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabbe
I already have that covered
[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson"
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip
The server is not behind NAT only the client above is
On Thu, Sep 16, 2010 at 4:59 PM, Paul Belang
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson wrote:
> Also, if I disable the firewall in my router I lose incoming audio and
> outgoing audio works.
>
http://www.aocomputing.net/?p=3
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freeno
I have tried doing that with just ulaw and alaw, respectively, and nothing
changed
Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.
On Thu, Sep 16, 2010 at 2:50 PM, Sebastian wrote:
>
>
> On 09/16/2010 07:59 PM, Thomas Johnson wrote:
> > the client
On Thu, Sep 16, 2010 at 07:44:23PM +0100, Jon Farmer wrote:
> Hi
>
> I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
> I do on INVITES is to re-authenticate the user from OpenSER. Then when
> the INVITE gets passed to Asterisk I capture the AUTH to a variable in
> the dialpl
So why can't you send the Auth line into the variable and then have your script
do the parsing to break out the segments you want.
Or if need be two scripts. The first can accept the authline as a full string
from a variable and break it down to its parts and save those as channel
variables. T
On 09/16/2010 07:59 PM, Thomas Johnson wrote:
> the client that is behind nat is
> [tomfmason]
> type=friend
> secret=secret
> callerid="Thomas Johnson"
> host=dynamic
> nat=yes
> canreinvite=no
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> qualify=yes
> context=sip
>
> do I have to ena
> On 16 September 2010 19:50, Danny Nicholas wrote:
> If you make the string into a dialplan Variable, you can do pretty much
> anything with it. Let's say your dialplan is like this
>
> - exten => 1234,1,blah
> - exten => 1234,n,AGI(myagi.xx,"1234")
>
> Change line 2 to
> - exten => 1234,n,AGI(
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AGI Delimiter in 1.6
On 16 September 2010 19:50, Danny Nicholas wrote:
> Two suggestions;
> #1. "escape" the , as \,
> #2. quote the string so 1,2,3 is "1,2,3"
I have thought about both of those ideas.
Is it possible to escape the string in the dialplan?
Applying quotes didn't seem to work, however I was pretty
Hi,
On 09/16/2010 05:28 PM, Gareth Blades wrote:
> One of the main benefits of qualify=yes is to detect network problems
> with peers.
> We send a lot of calls via a service provider using SIP but we have
> qualify-yes set so that if it becomes unreachable the dial fails
> immediatly without havin
the client that is behind nat is
[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson"
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip
do I have to enable nat on all of them?
On Thu, Sep 16, 2010 at 1:36 PM, Sebastian wrote:
>
>
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer
Sent: Thursday, September 16, 2010 1:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI Delimiter in 1.6
Hi
I am currently using 1
Hi
I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things
I do on INVITES is to re-authenticate the user from OpenSER. Then when
the INVITE gets passed to Asterisk I capture the AUTH to a variable in
the dialplan and pass to an AGI script. I am now trying to set the
same thing up i
On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> I am having a one way audio issue with xlite clients behind NAT. They
> can connect to the server and make calls but no audio is heard on the
> other end.
>
> my sip conf
>
> [general]
> context=default
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup
On 10-09-16 09:43 AM, Dan Journo wrote:
>> That's not a bug. Only when the phone registers or performs some sort of
>> action
>> (such as placing a call, etc...) does Asterisk query the database. If your
>> phones have a short re-registration time this becomes less of a problem.
>
> How do you exp
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="T
On Thu, Sep 16, 2010 at 12:46 PM, Paul Belanger
wrote:
> On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
> wrote:
>> Is it normal that backtrace.txt is only 30K ??
>>
> Normal or not, simply post the results of backtrace.txt
>
Please do not send me direct email, post them to the list for others
t
On Thu, Sep 16, 2010 at 12:03 PM, Chris Owen wrote:
> On Sep 16, 2010, at 10:44 AM, Steve Totaro wrote:
>
>> The other purpose is for DCHP and the IP address of a particular phone
>> may change. If you hard code the phone and the corresponding entry in
>> sip.conf, you don't need to register or u
On Thu, Sep 16, 2010 at 1:03 PM, carem gyssell nieto
wrote:
> It's an asterisk Bug? I have asterisk 1.4.22.
>
Please direct your attention to the following:
- http://www.catb.org/esr/faqs/smart-questions.html
-
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
-
On Thursday 16 September 2010 11:23:37 Dan Journo wrote:
> Is there any development work being done on the realtime addon? Theres been
> no updates since April.
Realtime is integrated into the core; it is not an addon. Perhaps you're
referring to the mysql realtime driver? The driver modules ten
On Thursday 16 September 2010 07:50:33 Steve Howes wrote:
> On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
> > Does anyone know how to send * a semi-colon from a realtime database. I
> > know that * uses the semi-colon as a 'seperator' - but I need to be able
> > to use one in a command. I know I
- "Paul Belanger" wrote:
> On Thu, Sep 16, 2010 at 11:42 AM, Tim Nelson
> wrote:
> > First, my apologies for the OT post. Yes, I understand this is not
> the FreePBX-users mailing list. But, there are a large number of
> people that use FreePBX and I'm hoping they can be of assistance.
> >
>
I have an incomming call but when I receive a call by a 2nd line in my
softphone, lost the first call. Sometimes the first call is dropped, and
sometimes the call is active, but I can't hear the caller.
It's an asterisk Bug? I have asterisk 1.4.22.
Please help!!!
Thanks
--
Carem
On Thu, Sep 16, 2010 at 11:42 AM, Tim Nelson wrote:
> First, my apologies for the OT post. Yes, I understand this is not the
> FreePBX-users mailing list. But, there are a large number of people that use
> FreePBX and I'm hoping they can be of assistance.
>
If you know this is off-topic, and not
On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
wrote:
> Is it normal that backtrace.txt is only 30K ??
>
Normal or not, simply post the results of backtrace.txt
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
On Thursday 16 Sep 2010, Tim Nelson wrote:
> I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX
> 2.6.0. There are a large number of inbound routes configured for the
> various DID's coming in via PRI, SIP, etc. If a user calls outbound to one
> of these numbers, it goes out
One of the main benefits of qualify=yes is to detect network problems
with peers.
We send a lot of calls via a service provider using SIP but we have
qualify-yes set so that if it becomes unreachable the dial fails
immediatly without having to wait for a timeout which enables us to
seamlessly f
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
Is there any development work being done on the realtime addon? Theres been no
updates since April.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webi
qualify=2000 does not mean it sends a qualify every 2000ms, 2 seconds. It
means that the qualify timeout is 2000ms, so if it receives a response at
2600ms, it counts that phone as down. I believe the timing of qualifies is
still every 60 seconds, unless explicitly changed by the system admin:
ht
> Have you checked the Issue Tracker
Not yet. I wanted to see if it's just me before searching through/raising a bug
report.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us f
> But it doesnt explain why the phones that are hard coded in the sip.conf
file don't lose registration.
On a reload, it re-reads the sip.conf config file and sees the users in
there, so it doesn't flush them. It doesn't pull down the whole SIP table
on a reload, it only loads a realtime peer con
On 09/16/2010 05:45 PM, Paul Belanger wrote:
> On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens
> wrote:
>
>> I get so little output :
>>
>>
> You are still doing it incorrectly. As said, doc/backtrace.txt has all
> the required information.
>
bash-3.2# gdb -se "/usr/sbin/asterisk" -
On 09/16/2010 12:01 PM, Chris Owen wrote:
well that just means you need a trunked satellite pbx where all the
phones are, and that would take load off the main connection.
half those people have got to just be talking to each other and don't
need to use the gateway at all.
> On Sep 16, 2010, a
On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote:
> I prefer to keep qualify=on for all the extensions, as it gives you an idea
> which extensions are going to give you trouble. For extensions with qualify
> value greater than 300 ms you should definitely worry. For extensions at
> 2000ms de
On Sep 16, 2010, at 10:44 AM, Steve Totaro wrote:
> The other purpose is for DCHP and the IP address of a particular phone
> may change. If you hard code the phone and the corresponding entry in
> sip.conf, you don't need to register or use qualify.
>
> If the phone is reachable then it will rep
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, September 16, 2010 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] [OT-FreePBX] Outbound c
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
Greetings-
First, my apologies for the OT post. Yes, I understand this is not the
FreePBX-users mailing list. But, there are a large number of people that use
FreePBX and I'm hoping they can be of assistance.
I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX
2.6.0. Ther
I prefer to keep qualify=on for all the extensions, as it gives you an idea
which extensions are going to give you trouble. For extensions with qualify
value greater than 300 ms you should definitely worry. For extensions at
2000ms delay or more, turning qualify off simply means to ignore the obvio
On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens wrote:
> I get so little output :
>
You are still doing it incorrectly. As said, doc/backtrace.txt has all
the required information.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
bl
On Thu, Sep 16, 2010 at 11:32 AM, Benny Amorsen wrote:
> Chris Owen writes:
>
>> So I guess my question is what is the real purpose of the qualify
>> setting in a non-NAT situation and can one safely set the
>> qualification as something higher. I'd think something like 15 seconds
>> would be mor
> Noted - but if OP does a "reload" once a day, 120 seconds (2 minutes) out of
> 1 day (14400 seconds) is 99.17% uptime; "close enough" to 99.999 percent in
> most folks books. What percentage of businesses use their phones 24/7?
Even if its once a month, it's still too much in my book. No wonder
When making an outbound call, if sip peer is not registered, first it
registers itself, and then makes the call. This is why you don't see any
problem dialing out. For receiving, asterisk has to wait until the sip peer
registers, otherwise asterisk has nowhere to send the call.
I know the pain, as
Chris Owen writes:
> So I guess my question is what is the real purpose of the qualify
> setting in a non-NAT situation and can one safely set the
> qualification as something higher. I'd think something like 15 seconds
> would be more than enough for BLFs and the like.
The purpose is simply to
On Wed, Sep 15, 2010 at 6:10 PM, Al lists wrote:
> I'm a long time user of Digium carts and stupid me i wanted to give Sangoma
> a try.
> We got Sangoma A400 with 6 FXO ports.
>
> Asterisk version: 1.4.35
> Zaptel version: 1.4.11
> Wanpipe version: 3.5.11
>
> we tried to use fxtune but looks like
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Thursday, September 16, 2010 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
Hello All!
I want to add a silence to the beginning of a ring tonelist for a country
inside the indications.conf file. I want that silence to be played just once,
reason why am using an exclamation mark in front of the tone but is not
working. Am getting the ring tone right away. I tried these
Danny Nicholas wrote:
If your clients can't take 2 minutes of "downtime" on a phone, they
don't need to be on VOIP.
If VOIP ( and Asterisk ) ever really expect to be "the future of
Telephony " this ( attitude ) has to change
90 percent availability is unacceptable, even 95 percent,
> As someone else said, the answer is
"don't do a 'reload'", do an "extensions reload" or whatever it is specific
to your changes.
You are correct. I'm just being lazy. But I'm just worried that some time in
the future, I'll have to reload the sip config, and therefore flush out all the
realtime
> A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.
Finally an answer that seemed more realistic. But it doesnt explain why the
phones that are hard coded in the sip.conf file don't lose registration.
Any ideas?
Thanks
Dan
--
__
Jerry Geis wrote:
>>
>> Somewhere on your system you have a modprobe install command that's
>> running when the module is loaded. Most likely it was installed on your
>> system by
>> http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup
>>
>>
>> when you installed z
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?
A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone. If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine. Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime. I
agr
>
> Somewhere on your system you have a modprobe install command that's
> running when the module is loaded. Most likely it was installed on your
> system by
> http://svn.asterisk.org/view/zaptel/branches/1.4/build_tools/genmodconf?view=markup
> when you installed zaptel.
>
> Do you have an /etc/c
Apart from that, any other tricks that I can manipulate within asterisk.
??sip.conf parameter or other??
On Thu, Sep 16, 2010 at 12:07 AM, Luki wrote:
> > I am not sure about the problem but note that it may be related to
> incorrect
> > IP being used. Sometimes, WAN 1 and sometimes WAN 2
>
> Mo
> That's not a bug. Only when the phone registers or performs some sort of
> action
> (such as placing a call, etc...) does Asterisk query the database. If your
> phones have a short re-registration time this becomes less of a problem.
How do you explain that as soon as I issue a "reload" comma
I encountered something strange. A local business has an ACD that, when I call
it using a Polycom 550 connected through an Asterisk system, will respond to
button presses only if they are short.
Calling this business with our old (non-Asterisk) phone system or with my cell
phone works because
On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
> Does anyone know how to send * a semi-colon from a realtime database. I
> know that * uses the semi-colon as a 'seperator' - but I need to be able
> to use one in a command. I know I can use \; in the non-realtime
> configs, but this doesn't work i
Its working now when I installed openssl using yum.
yum install -y openssl-devel.
Thanks
Nikhil
On 09/16/2010 05:26 PM, Nikhil wrote:
>On 09/16/2010 04:11 PM, A J Stiles wrote:
>> On Thursday 16 Sep 2010, Nikhil wrote:
>>> Hi
>>> I got the bellow error when I try to configure ast
On 09/16/2010 12:41 PM, Philipp von Klitzing wrote:
> Hi!
>
>> Does this shine new light to the problem ?!
>>
> No. Once more: Go and read doc/backtrace.txt.
>
> And check if you have any meaningful information in /var/log/messages for
> the timestamp when asterisk crashed.
>
> Philipp
>
Hello my friends,
I would like to understand the output from "pri intense debug span X", the
Telco always says that their side is OK, but i always receive alarms,
loosing connection, take a look:
[Sep 16 13:18:19] WARNING[30364] chan_zap.c: Detected alarm on channel 1:
Recovering
[Sep 16 13:18:19
On 09/16/2010 04:11 PM, A J Stiles wrote:
> On Thursday 16 Sep 2010, Nikhil wrote:
>>Hi
>> I got the bellow error when I try to configure asterisk code.
>>
>> $./configure --with-ssl=/usr/local/ssl
>> ...
>> ...
>> ...
>> checking for mandatory modules: OPENSSL... fail
>>
>> configure:
Hi list,
Does anyone know how to send * a semi-colon from a realtime database. I
know that * uses the semi-colon as a 'seperator' - but I need to be able
to use one in a command. I know I can use \; in the non-realtime
configs, but this doesn't work in realtime.
Cheers,
Andrew Thomas
Technical
Hello
You has installed a2b 1.7 version, and also had not do some permissions
on folder and files.
/usr/local/src/a2billing/admin/templates_c'. Be sure $compile_dir is
writable by the web server user. in
/usr/local/src/a2billing/common/lib/smarty/Smarty.class.php on line 1093
I think the best
Hi!
> Does this shine new light to the problem ?!
No. Once more: Go and read doc/backtrace.txt.
And check if you have any meaningful information in /var/log/messages for
the timestamp when asterisk crashed.
Philipp
--
_
--
On Thursday 16 Sep 2010, Nikhil wrote:
> Hi
>I got the bellow error when I try to configure asterisk code.
>
> $./configure --with-ssl=/usr/local/ssl
> ...
> ...
> ...
> checking for mandatory modules: OPENSSL... fail
>
> configure: ***
> configure: *** The OPENSSL installation appears to be
Hi
I got the bellow error when I try to configure asterisk code.
$./configure --with-ssl=/usr/local/ssl
...
...
...
checking for mandatory modules: OPENSSL... fail
configure: ***
configure: *** The OPENSSL installation appears to be missing or broken.
configure: *** Either correct the instal
Hello,
I have new information from a newly created test environment :
[Sep 16 13:41:01] -- Executing [...@macro-vakantie:1]
MYSQL("SIP/test1-0008", "Connect connid localhost username passwd
AsteriskHosted") in new stack
[Sep 16 13:41:01] -- Executing [...@macro-vakantie:2]
MYSQL(
Hello,
You sould go to the admin page (a2billing/admin/). There are two possibles web
pages for a2b: the admin page and the customer page. You should point to the
one you like in each moment :)
César Pinto
Alhambra-Eidos
De: asterisk-users-boun...@lists.digi
В Чтв, 16/09/2010 в 12:44 +0530, DHAVAL INDRODIYA пишет:
> Thanks for update if a file is converted to text then where can i find
> a text file like after running
> pocketsphinx_continuous command where text saved.
Text is in the last line:
0: we've entered the property the identificatio
Thanks for update if a file is converted to text then where can i find a
text file like after running
pocketsphinx_continuous command where text saved.
regards
dhaval
On Thu, Sep 16, 2010 at 12:29 PM, Nickolay V. Shmyrev wrote:
> В Чтв, 16/09/2010 в 10:35 +0530, DHAVAL INDRODIYA пишет:
> >
> >
Hello list,
are there special things that needs to be done when converting BLF from
asterisk 1.4 tot 1.6.2 ?!
I have replaced call-limit with call-counter, but it seems that the
lights on the phone no longer give the status of the extension they monitor.
On Snom phones, when the lights shou
В Чтв, 16/09/2010 в 10:35 +0530, DHAVAL INDRODIYA пишет:
>
> Hi Nickolay,
>
> here i attached my file. please have a look into it.
Hello DHAVAL
As I wrote your file has wrong format.
$ file ask-propertyid.WAV
ask-propertyid.WAV: RIFF (little-endian) data, WAVE audio,
GSM 6.10, mono 8
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