I use following diaplan for extension 2005:
-
extension name=2005
condition field=destination_number expression=^2005$
action application=set data=”call_timeout=10/
action application=set
Hello all !
FreeSWITCH works good with Sangoma A 10X cards in Linux ( it was checked by
us ).
For this cards FreeSWITCH uses TDM API interface, which was written specially
for
FreeSWITCH into few last releases of WANPIPE.
In this case for timeslots such names are created:
Try upgrading to trunk... There are several known issues with 1.0.1 that
will not be fixed in that release... 1.0.2 is on the burner and will be out
shortly...
From: [EMAIL PROTECTED] [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 25 Nov 2008 12:01:58 +0200
I have a perl script (for dialplan generation) that works fine.
When I try to use the DBI module I get a segmentation fault. My OS is
Linux CentOS 5.2
and I am using freeswitch-1.0.1.
If I can recall correctly, some other guy had the same problem a few
months ago but I cannot
find the mailing
I faced a strange behavior with my freeswitch...
Freeswitch setup is as follows:
Any call comes to freeswitch are sent to javascript, for routing based on
some logic.
After call completes, the same script insert the CDR values into mysql
database.
All is working fine for quite long time,
But
Faisal Maqsoodi wrote:
I ve just installed freeswitch on my system. What should i try on it now
to take a start? I ve gone through the theory of pbx and all that. faisal
http://wiki.freeswitch.org/wiki/Getting_Started_Guide
http://wiki.freeswitch.org/wiki/SOHO_PBX_Example
--
View this
I ve just installed freeswitch on my system. What should i try on it now to
take a start? I ve gone through the theory of pbx and all that. faisal
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try using export instead of set for that var.
Mike
On Nov 25, 2008, at 4:05 AM, henkoegema wrote:
I use following diaplan for extension 2005:
-
extension name=2005
condition field=destination_number expression=^2005$
action
You have just shown exactly why cdr direct to database is a bad idea.
Most likely there was a blip in connectivity to the db and it didn't
reconnect correctly. This behavior would be caused by some blocking
in your script.
Mike
On Nov 25, 2008, at 4:55 AM, shehzad p wrote:
I faced a
it should work both ways but you are better off using it in native mode.
When you use it as a zaptel device just follow the instructions for a zaptel
device and forget
it's a wanpipe device and it would work
here is the zaptel and wanpipe version of the same span.
[span zt myspan]
fxs-channel =
also try
action application=bridge
data={originate_tiimeout=10}sofia/internal/2005%$${domain}/
On Tue, Nov 25, 2008 at 6:46 AM, Michael Jerris [EMAIL PROTECTED] wrote:
try using export instead of set for that var.
Mike
On Nov 25, 2008, at 4:05 AM, henkoegema wrote:
I use following
Hi Michael,
Sorry to nag you again.
I'm getting close to my deadline and I would like to give our customer
some feedback on this issue.
Would you be so kind to explain what the issue is with recording rings
when doing 'new Session(originate_url)'?
As far as I've been able to work out, this
hi,
i have big problems with disconnects, when bridging and unbridging calls.
because i had random diconnects when testing fs from one softphone to
the other, i set up a little dialplan (socket outbound), to do some
hardcore testing.
after the inbound is answered, i do an originate to an
The originate method does not return until either early media or answer has
been received.
in other words the very instant it returns is the soonest there is even any
media to record.
you can execute record_session app on the A leg before you call and that's
the best you can do.
Doens't make any difference. :working:
Michael Jerris wrote:
try using export instead of set for that var.
Mike
On Nov 25, 2008, at 4:05 AM, henkoegema wrote:
I use following diaplan for extension 2005:
-
extension
Hi Anthony,
Thanks for your feedback.
I still cannot get it to work, maybe the snippet and the output will
give you an idea?
*** I've got the following javascript:
var endpoint_url = 'sofia/gateway/westhawk/0663';
var recordfile = /usr/local/freeswitch/recordings/dispatcher.wav;
Hi Peter,
Thanks (and to Mike) - an update to current's got things working just fine.
Cheers --
Dave
I do and I got it finally working under Ubuntu 8.041:
Here is how it worked under Ubuntu 8.04:
First, disable mod_flite as they are incompatibel
Then set environment var export
Sent from my iPhone
On Nov 25, 2008, at 6:03 AM, Birgit Arkesteijn [EMAIL PROTECTED]
wrote:
Hi Michael,
Sorry to nag you again.
I'm getting close to my deadline and I would like to give our customer
some feedback on this issue.
Would you be so kind to explain what the issue is with
Anthony Minessale-2 wrote:
also try
action application=bridge
data={originate_tiimeout=10}sofia/internal/2005%$${domain}/
YES, that works !!
But... why doesn't action application=set data=call_timeout=10/
work ?
--
View this message in context:
iirc call_timeout has been changed to originate_timeout
On Tue, Nov 25, 2008 at 1:04 PM, henkoegema [EMAIL PROTECTED]wrote:
Anthony Minessale-2 wrote:
also try
action application=bridge
data={originate_tiimeout=10}sofia/internal/2005%$${domain}/
YES, that works !!
On Tue, Nov 25, 2008 at 11:44 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
iirc call_timeout has been changed to originate_timeout
That seems to jive with what's in switch_ivr_originate.c as there is no
mention of call_timeout anywhere in the function switch_ivr_originate(),
but
Michael Collins-11 wrote:
On Tue, Nov 25, 2008 at 11:44 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
iirc call_timeout has been changed to originate_timeout
That seems to jive with what's in switch_ivr_originate.c as there is no
mention of call_timeout anywhere in the function
I used call-timeout because that's how I understood it from the Wiki.
(?)
Yep, that's all that there is on the wiki. Unfortunately the channel
variables page is one of many in need of some attention. I will add
originate_timeout right away. The only question remaining is what, if
anything,
FYI, it is on the channel variables page but it's in a crazy place under
unknown functionality which is silly.
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
next month when I have some time...
-MC
On
I can call between VoIP phone ext 1001 and 1003 fine. I can call
from VoIP phone ext 1003 over a winkstart line into the PBX fine.
Before updating to the current SVN I could also call from the PBX
over a winkstart line to VoIP ext 1003 fine. Now what I see is:
[NOTICE] switch_channel.c:551
One way to help narrow it down is to revert back to an earlier working
version. In fact, if you can find the tipping point where it went from
working to not-working then that would really help narrow things down.
Do you know the last rev that actually worked? Looking at the change
logs I see a
He's using the domain aware configs without setting the domain_name
variable thats all :P (might wanna set the variable so it knows what
domain the call should land in!)
/b
On Nov 25, 2008, at 7:23 PM, Michael Collins wrote:
BTW, this is why I try to upgrade frequently - if it worked
Sent from my iPhone
On Nov 25, 2008, at 5:28 PM, Brian West [EMAIL PROTECTED] wrote:
He's using the domain aware configs without setting the domain_name
variable thats all :P (might wanna set the variable so it knows what
domain the call should land in!)
Yeah I suppose, but I think the
make sure you have the domain_name variable set so the call can go to
the right domain name.
Set where? I'm pretty much using the stock sample files. vars.xml contains:
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
It was my understanding that domain defaults to the IP address of
No domain != domain_name they are different.
/b
On Nov 25, 2008, at 6:54 PM, John Wehle wrote:
Set where? I'm pretty much using the stock sample files. vars.xml
contains:
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
It was my understanding that domain defaults to the IP address
No domain != domain_name they are different.
Well ... yes and no. default.xml has:
extension name=set_domain continue=true
condition field=${domain_name} expression=^$/
condition field=source expression=mod_sofia/
condition field=${sip_auth_realm} expression=^$
This is the key to why its not working source is mod_openzap so might
Yes ... I understand that. I'm pretty much running the supplied sample
configuration which apparently doesn't handle openzap. Is there a
specific JIRA category I should use to log this issue? I took a quick
glance and
Well for OpenZAP I'll have to work out how I should work it in to the
default configs. Its not a bug its just something that isn't
accounted for in the configs as those configs are sip centric.
/b
On Nov 25, 2008, at 7:59 PM, John Wehle wrote:
Yes ... I understand that. I'm pretty much
Brian,
So all you did was create a dp entry named set_openzap_domain and
have it match on two separate conditions: ${domain_name} is empty and
source is mod_openzap. The only thing it does is to set the domain
name to the default domain. And, since you don't want this entry to be
the only
Were are the PocketSphinx's recognizer and recordings logs kept in Windows
version of FS?
Has anyone tried to getting the proper cmn value from looking at these logs and
added -cmninit option to the recognizer?
Mark.
?
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