Hi Michael,
My dial plan is:
When call come to xxx.xxx.xxx.x system, It answer the call and only wait for
20 seconds (NO playback only wait) and hangup.
Please find debug trace for above dialpla
Raul Fragoso wrote:
> Hi João,
>
> Please say hello to Giancarlo at Khomp for me :)
>
> Khomp is the best example in Brazil of what good engineering and fair
> commercial prices can do to a country that suffers from high import
> taxes. Unfortunately some companies are still inclined to buy importe
I'm running rev 11131, and I cannot make these hangup events work in
javascript... does anyone have a working example? I tried this script
from the wiki: http://wiki.freeswitch.org/wiki/Example_Hangup_hook
--Stephen
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Fre
Hi João,
Please say hello to Giancarlo at Khomp for me :)
Khomp is the best example in Brazil of what good engineering and fair
commercial prices can do to a country that suffers from high import
taxes. Unfortunately some companies are still inclined to buy imported
products/brands instead of fav
You may need to set P-Asserted-Identity or Remote-Party-ID headers
Sent from my iPhone
On Jan 26, 2009, at 8:05 PM, Ron McCarthy wrote:
> Hi,
>
> I am having a weird issue with setting the callerID number for
> outbound calls, I have this:
>
> data="effective_caller_id_number=17025551234"/>
Hi João,
João Mesquita wrote:
> Steve,
>
> As we speak I am actually negotiating with one of those companies to
> make a mod for their cards. Khomp has a very nice product and they are
> exporting to the rest of latin america now.
>
It surprises me someone doesn't assemble Tormenta 2
Steve,
As we speak I am actually negotiating with one of those companies to
make a mod for their cards. Khomp has a very nice product and they are
exporting to the rest of latin america now.
Thanks,
Mesquita
On Jan 27, 2009, at 12:46 AM, Steve Underwood wrote:
> Hi Abdul,
>
Hi Abdul,
Abdul Hakeem wrote:
> Is Brazil a 3rd world country ? The last I hear Brazil was building
> aeroplanes, has it's own space and nuclear program and a GNP UK would be
> envious of.
> Cheers,
> AH
>
What relevance does that have to the current discussion?
Brazil is a country with lar
Hi,
I am having a weird issue with setting the callerID number for outbound
calls, I have this:
Ive set the callerID correcly on a Asterisk box and the carrier sees it and
passed it correct, but for some reason any calls from Freeswitch won't work,
it either shows up as private or another numb
Brian West wrote:
> haha so was that one! :P Anchors Away!!!
>
> /b
>
> On Jan 26, 2009, at 6:11 PM, Adam Long wrote:
>
>> Sorry that last link was mangled...
>>
>> If you are running inside ESX might want to have a look here...
>> http://kb.vmware.com/selfservice/microsites/search.do?language=en_
Double check your registration because the user isn't registered.
Check "sofia status profile internal"
/b
On Jan 26, 2009, at 6:16 PM, Jim Archer wrote:
> Sorry for all these simple questions, but I'm having a registration
> issue.
>
> I have two Polycom 501 phones and have configured both
haha so was that one! :P Anchors Away!!!
/b
On Jan 26, 2009, at 6:11 PM, Adam Long wrote:
Sorry that last link was mangled...
If you are running inside ESX might want to have a look here...
http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis
playKC&externalId=2219
Sorry for all these simple questions, but I'm having a registration issue.
I have two Polycom 501 phones and have configured both to FreeSwitch. One
is extension 1000 and the other is extension 1001. I can call 1000 from
1001, but not the reverse. If I try to call 1001, I go straight to voice
Sorry that last link was mangled...
If you are running inside ESX might want to have a look here...
http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis
playKC&externalId=2219
-Adam
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:fr
Hello Everyone,
Has anyone done a scalable deployment of calling to Skype/Gtalk/Asterisk
Say many 3-25/50 users with various endpoint
(skype/gtalk/asterisk)platforms.
http://blog.tmcnet.com/blog/tom-keating/asterisk/skype-for-asterisk-launches.asp
http://www.mhspot.com/siptheeskype_skype_trunk_ho
Haha... it is probably either a case of overloading the ESX host server
(too many other guests on same machine)
Or your ESX isn't pumping out enough interrupts.
Have a look here.. this might help.
http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=dis
playKC&externalId=2219
Oh, there is a "digit map" in there. All fixed, thanks very much!
--On Monday, January 26, 2009 3:23 PM -0800 Michael Collins
wrote:
> Sounds like the Polycoms are expecting only two digits when you start
> with a 1. Have these phones been used before? Perhaps you could reset
> one of them to
running ntpd will help but if your clock is slipping that much you
might want to re-task the machine as a boat anchor.
/b
On Jan 26, 2009, at 5:42 PM, Chav Paskov wrote:
> Thanks for the prompt response.
> will it help if i set a cron job to keep up the clock correct?
> i'm just curious be
Brian West wrote:
> It means your clock is slipping .. which is bad... The error is for
> when you do a migration say between two xen boxes or two openvz boxes
> so that things will carry on once the migration is done. (or your
> clock is slipping)
>
>
> /b
>
> On Jan 26, 2009, at 5:28 PM, Ad
It means your clock is slipping .. which is bad... The error is for
when you do a migration say between two xen boxes or two openvz boxes
so that things will carry on once the migration is done. (or your
clock is slipping)
/b
On Jan 26, 2009, at 5:28 PM, Adam Long wrote:
> I have seen tha
I have seen that randomly during core dumps back when mod_managed was
causing core dump on load.
I am running VMWare ESX 3.5 not ESXi
My curiousity was peaked as well, however I dismissed that as something to
do with slow/inacurate timing in vmware.
I do not have my ESX interupts tuned at all, jus
You'll need to fix your dialplan in the polycom itself its sending the
invite after you dial 10.
/b
On Jan 26, 2009, at 5:16 PM, Jim Archer wrote:
> enum...@default]
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http://lis
Sounds like the Polycoms are expecting only two digits when you start
with a 1. Have these phones been used before? Perhaps you could reset
one of them to the factory default and start from scratch and see what
happens.
-MC
On Mon, Jan 26, 2009 at 3:16 PM, Jim Archer wrote:
> Hi All...
>
> I inst
Hi everybody,
i got a strange message today:
[CRIT] softtimer_runtime() Virtual Migration Detected! Syncing Clock
i'm running 1.0.trunk (10803) as a VM on ESXi
i have a similar setup of 1.0.1 as VM on ESXi but i've never seen
this message.
Does anybody have an idea what is wrong here?
A
Hi All...
I installed FS trunk a few days ago on a Debian Etch AMD64 machine. I
configured two Polycom 501 phones to talk to it. From each phone, I can
dial extension to get the MOH and 5000 to get the sample IVR. But, I
can not call one phone from the other. One is extension 1000 and t
Can you try the groups that come with the default dialplan and see if
they work? Just curious. Also, what revision are you running?
-MC
On Mon, Jan 26, 2009 at 2:19 PM, Laurent Fabre
wrote:
> Hi,
>
>
>
> I'm having trouble with call groups. They are declared in the directory.xml
> as mentioned by
Hi,
I'm having trouble with call groups. They are declared in the directory.xml as
mentioned by the documentation :
And included in the dialplan :
The endpoints are registered, the dialplan match but I get a cause:
NO_ROUTE_DESTINATIO
Is Brazil a 3rd world country ? The last I hear Brazil was building
aeroplanes, has it's own space and nuclear program and a GNP UK would be
envious of.
Cheers,
AH
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
Oh it was you that Hijacked the thread... sorry Ron... It wasn't you
this time! ;)
David naughty! :)
/b
On Jul 29, 2008, at 7:20 AM, David Knell wrote:
> Hi -
>
> chat - sending a messages - only works to a SIP endpoint if the user's
> profile is the same
> as their host - the relevant code
Yes you would use the directory... you add a cidr= attr to the user in
the directory:
Say you add a user into "domain.com":
...
...
Then in acl.conf.xml you have something like this:
the above entry would create an acl from all users in "dom
Hi Michael,
I'm not sure about this, when we need to use G729 on asterisk for example, we
pay the digium (U$ 10/channel) licenses.
U$ 1.00 dolar = R$ 2.31 (Brazilian Real - local currency).
Att.,
Em 23-01-2009 18:29, Michael Collins escreveu:
> On Fri, Jan 23, 2009 at 12:16 PM, Rodrigo P. Telle
Hi,
If I allow the IPs in a ACL the context always ges changed, is their a way
you can have inbound calls be auth'ed via a gateway and then sent to a
context. Do I use the directory in that case and not treat it as a external
gateway, it would be considered internal then?
Thanks!
_
You may want to define the conf/tones.conf file with the proper tone
definitions for your country, then specify that tone group (tonegroup=XX
where XX is the 2 letters code for your country). For example, this is
the definition for my country (Brazil) and I also have tonegroup=br in
my openzap conf
you're right, its fixed int tree to explain the issue now.
On Mon, Jan 26, 2009 at 11:41 AM, Milena wrote:
> On a second thought...
>
> it would be nice if the console showed a message when there is a wrong
> password, something more descriptive than just doing nothing,
> also it is not clear t
On a second thought...
it would be nice if the console showed a message when there is a wrong
password, something more descriptive than just doing nothing,
also it is not clear to me why the console didn't require me to put the
password before i deleted that folder's content even knowing that i ch
Hello freeswitchers,
I'm experimenting with sip clients registered to fs and Instant
Messaging.
I've seen SIP Messages are properly routed by the sofia sip stack in fs.
However it seems no event is ever generated on the event socket.
Would it be possible to make fs reporting the SIP Messaging? :-)
yes some code was missing for some reason, try again
On Mon, Jan 26, 2009 at 10:11 AM, Tamas Cseke wrote:
> Hello,
>
> I tested with the attached patch.
> It is working fine in a normal case.
>
> I have only problems with the automatic calls, because in this case the
> loopback channel is in the
thanks ^^
2009/1/26 Mathieu Rene
> Loos like the wrong password to me, look in
> /usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use
> fs_cli -p [pass]
>
>
> Mathieu
>
> On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote:
>
>> Good morning
>> Ok, here is what i get from the con
Loos like the wrong password to me, look in
/usr/local/freeswitch/conf/autoload_configs/event_socket.conf.xml and use
fs_cli -p [pass]
Mathieu
On Mon, Jan 26, 2009 at 8:33 AM, Milena wrote:
> Good morning
> Ok, here is what i get from the console, do you know what can i do to fix
> it? thank y
What have you tried?
-MC
On Mon, Jan 26, 2009 at 2:25 AM, Sias Mey wrote:
> On a similar note is it possible to use api commands from the dialplan.
>
> I would like a execute_on_answer to run a script in the same fasion, but
> I cant seem to get it to execute as a api command.
>
> ___
Hello,
I tested with the attached patch.
It is working fine in a normal case.
I have only problems with the automatic calls, because in this case the
loopback channel is in the fifo, but the record_session is running on
the sofia channel.
Maybe it could be sort out with putting the bug pause/r
Good morning
Ok, here is what i get from the console, do you know what can i do to fix
it? thank you very much
-bash-3.2# /usr/local/freeswitch/bin/fs_cli -d 7
[DEBUG] libs/esl/src/esl_config.c:56 esl_config_open_file() Configuration
file is /root/.fs_cli_conf.
[DEBUG] libs/esl/src/esl_config.c:56
I think the problem with changing the CPU's is off your going from a single
to multi, or multi to single CPU and not adjusting the kernel from SMP to
normal or vise versa?
I've got a nice ESXi farm at the moment (32 cores, 48 gigs of ram) thats
running at under 5% usage. Haven't noticed any timing
Great, well I am not a programmer and do not understand the code at all :)
My Apology
Rehan
> On Saturday 24 January 2009 05:59:08 Rehan Allah Wala wrote:
> > Spacibah Balshoi
> >
> > When are you making g723 for the Russians?
> >
> I'm so sorry, but g729 is only one we need. But you can do it y
On a similar note is it possible to use api commands from the dialplan.
I would like a execute_on_answer to run a script in the same fasion, but
I cant seem to get it to execute as a api command.
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On Saturday 24 January 2009 05:59:08 Rehan Allah Wala wrote:
> Spacibah Balshoi
>
> When are you making g723 for the Russians?
>
I'm so sorry, but g729 is only one we need. But you can do it yourself from
free asterisk codec - it's not so hard, just see my code and compare it with
mod_g723 and as
Did you ever post your dialplan and a debug trace of a call to the
pastebin? If not, please do so and we will check it out.
-MC
On Sat, Jan 24, 2009 at 5:47 AM, shehzad p wrote:
>
> Hi all,
>
> On my existing Freeswitch 1.0.2, I installed and configured mod_vmd as
> below:
> make mod_vmd-install
I have a TDM400 clone and I will see if I can reproduce these
symptoms. BTW, are you in the Philippines? Is there any difference in
the dial tone there than in the US?
-MC
On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote:
> i monitored the line using another phone. there's indeed dialtone i
Hello,
Thank you your help.
I tested with r11489, but moh is still recorded in fifo.
I quess you I should test the CF_PAUSE_BUGS in r11466.
But I didn't find where you check this flag.
Is it maybe possible you forget to commit something?
Thanks,
Tamas
I didn't find where you
Anthony Minessal
MC / Anthony,
Muchas gracias ... looks like I'm going to be on my way to being * free :)
Ran a small production test today (about 20,000 dials) and going to get
after it tomorrow with a real campaign.
You guys kick ass.
Thanks again for the assistance fine sirs.
Shelby
On Fri, Jan 23, 2009 a
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