Hello,
I'm using mod_event_socket to monitor FS.
I'm using "events plain ALL' and I get lots of channel events. But
curiously, when some channel puts the call on-hold/off-hold, I don't
get any notification. Is it possible to get these events? Am I missing
some setting?
regards,
takeshi
__
2009/8/6 João Mesquita :
> I only see one way out of this. If you manage presence, an event like the
> following is sent:
>
> Event-Name: PRESENCE_IN
> Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
> FreeSWITCH-Hostname: cl-t146-421cl
> FreeSWITCH-IPv4: XX
> FreeSWITCH-IPv6: %3A%3A1
> Event-D
On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHV wrote:
>
> Hi,
>
> sorry - i have to ask again. This is my entry in
> /conf/sip_profiles/external/ :
>
>
>
>
>
>
>
>
>
>
> and it´s still the same... from: 2395...@139.13.37.160
>
> Do you have another tip?
I don't see where you are setting the para
On Tue, Aug 18, 2009 at 10:10 PM, NOx-WHV wrote:
>
> Hi,
>
> this is the text without brackets:
>
> include
> gateway name="sipgate.de"
> param name="username" value="2395805"/
> param name="from-domain" value="sipgate.de"/
> param name="password" value="abcde"/
> param name="proxy" value="sipgate.
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga wrote:
>
> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
> 4000
> scenario file: uac_redirect.xml
> FS dialplan: public.xml
> SIP trace:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi wrote:
> On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga wrote:
>>
>> sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
>
Hello,
I'm testing FS support for the header Path (FS is behind opensips).
It pretty much works: I tested calling from one user to the other and calls
work perfectly.
However, I've noticed that when I register my terminal directly with FS
without going thru the proxy, I receive an unsolicited NOTIF
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote:
> Hello,
> I'm testing FS support for the header Path (FS is behind opensips).
> It pretty much works: I tested calling from one user to the other and calls
> work perfectly.
> However, I've noticed that when I regi
hanks.
I've located the option in the mod sofia source:
send-message-query-on-register
It seems it is not in the wiki. I will update the mod sofia page.
> On Sep 2, 2009 9:20 PM, "mayamatakeshi" wrote:
>
>
>
> On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi
> wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote:
> Hello,
> I'm testing FS support for the header Path (FS is behind opensips).
> It pretty much works: I tested calling from one user to the other and calls
> work perfectly.
> However, I've noticed that when I regi
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi wrote:
>
> On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote:
>
>> Hello,
>> I'm testing FS support for the header Path (FS is behind opensips).
>> It pretty much works: I tested calling from one user to the o
On Sun, Sep 6, 2009 at 12:13 AM, Woody Dickson wrote:
> Hi,
>
> I would like to set up freeswitch to automatically expire a user
> registration if either NOTIFY or REGISTER is not received within certain
> time frame.
>
> Does anyone know how to do that?
>
I don't know about NOTIFY.
But for REGIS
On Sat, Sep 5, 2009 at 5:36 PM, Juan Backson wrote:
> Hi,
>
> I am getting no dial-string available error when using xml_odbc module to
> bridge a call. How can I resolve this problem?
>
Hello,
I never tried the mod_xml_odbc.
But as the message says, you are not providing a dial-string.
I belie
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris wrote:
> Following up, did a bug get created for this issue?
>
Hello,
yes.
http://jira.freeswitch.org/browse/MODSOFIA-26
>
> On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
>
>
> On Fri, Sep 4, 2009 at 7:28 PM, mayamatake
On 9/12/09, mayamatakeshi wrote:
>
>
> On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris wrote:
>
>> Following up, did a bug get created for this issue?
>>
>
> Hello,
> yes.
> http://jira.freeswitch.org/browse/MODSOFIA-26
>
Just to simplify things in ca
On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins wrote:
> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>
Very nice.
But I think a valet_unpark app is missing.
If the intention of the person sent to the
On Sun, Oct 11, 2009 at 12:18 PM, mayamatakeshi wrote:
>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us
Hello,
I'm playing with FreeSwitch svn revision 8849.
When I try to put an established call on hold (I'm using eyebeam),
FreeSwitch takes 5 seconds to reply with 200 OK.
This would not be a big problem by itself, but it seems this OnHold
processing blocks the reply for all other incoming calls (or
Brian, thanks,
it's fine now.
On Sat, Jun 28, 2008 at 1:20 AM, Brian West <[EMAIL PROTECTED]> wrote:
> Try anything after rev 8857
> /b
>
> On Jun 26, 2008, at 8:23 PM, mayamatakeshi wrote:
>
> Hello,
> I'm playing with FreeSwitch svn revision 8849.
>
Hello,
I'm a little confused with FreeSWITCH behavior.
At http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:
"The directory section is used to add accounts for all users that should be
able to register in the pbx by using User Agents (SIP Phones)."
So I suppose I should provide
On Wed, Aug 6, 2008 at 4:42 PM, mayamatakeshi <[EMAIL PROTECTED]>wrote:
> Hello,
> I'm a little confused with FreeSWITCH behavior.
> At http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
> we can read:
> "The directory section is used to add accounts for all
gt; Could you please post a bug report of this on jira.freeswitch.org? This
> is definitely not expected behavior, so if this is happening it should be
> fixed.
>
> -Ray
>
> mayamatakeshi wrote:
>
>
>
> On Wed, Aug 6, 2008 at 4:42 PM, mayamatakeshi <[EMAIL PROTECTED]&g
Hello,
I'm trying to update my sources, but now http://svn.freeswitch.org is asking
for usr/pass. It never did this before.
Is this on purpose? Should we register for them?
Regards,
Takeshi
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswi
e 'false' and not 'true'
(that's the reason why the thing is commented in the sample file).
I believe the correction mentioned by Brian also corrected that, as rev 9232
does honor the value set for accept-blind-reg and 'false' is assumed by
default.
So, case closed
On Thu, Aug 7, 2008 at 9:49 PM, mayamatakeshi <[EMAIL PROTECTED]>wrote:
> Hello,
> I'm trying to update my sources, but now http://svn.freeswitch.org is
> asking for usr/pass. It never did this before.
> Is this on purpose? Should we register for them?
>
I don't k
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov wrote:
> Hi there!
>
> Please, suggest how to specify custom caller sip domain (logical) in
> originate command.
> I've been trying several alternatives but no one worked:
> 1) specify full sip address in
> origination_caller_id_number=1...@uat.pbx
Hello,
in this page
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:
=
Variables:
Any variables defined in the domain or user will be defined as channel
variables when there is a call to user or when there is an inbound calls
from that user.
===
that
> user.
>
> /b
>
> On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:
>
> > I can see the channel variables are set when there is an incoming
> > call from the user, but not when FS sends a call to the user.
>
I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets from
eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
it doesn't change the RTP destination to the source of those packets and
sends the packets to the
I'm really sorry, guys.
I was just some revisions behind and forgot to make current and test before
posting.
It's fine now.
Thank you.
On Thu, Oct 29, 2009 at 1:51 AM, Brian West wrote:
> have you updated to the latest SVN?
>
> /b
>
> On Oct 28, 2009, at 11:35 AM, mayam
Hello,
it is my understanding that FreeSWITCH doesn't provide a ParkServer per se.
So, to provide for this, I will have an entry in the dialplan to play MOH on
the channel continuously till someone retrieve the call.
And then, I need to publish a NOTIFY to all subscribed users informing the
status
About mod_fifo, it would be safe to use it in multi-tenancy scenarios where
domains are created and deleted all the time and in consequence, fifos are
created all the time? I mean, fifos are eventually destroyed by mod_fifo
itself. Is this correct?
br,
takeshi
_
On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote:
>
>
> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi wrote:
>
>> About mod_fifo, it would be safe to use it in multi-tenancy scenarios
>> where domains are created and deleted all the time and in consequence, fifo
I had this same problem today.
I solved it using
OPTION = 67108864
instead of
OPTIONS = 67108864
I'm using CentOS5.3 (x86_64)
br,
takeshi
On Sat, Nov 28, 2009 at 12:36 AM, Frank @ Impact wrote:
> Yes. I am using version 5.1 I am using Fedora 12.
>
>
>
> -Original Message-
> *From:* fr
It seems to me, in previous revisions of FS, we could successfully call a
registered user as soon as his terminal gets 200 OK for REGISTER.
But after testing recent revisions, it seems we must wait a little (I wait 1
second) otherwise a call to bridge would end with this:
2009-12-17 22:46:14.15516
://wiki.freeswitch.org/wiki/Sofia.conf.xml#sql-in-transactions
> On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi wrote:
>
>> It seems to me, in previous revisions of FS, we could successfully call a
>> registered user as soon as his terminal gets 200 OK for REGISTER.
>> But af
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