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
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
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]wrote:
Hello,
I'm a little confused with FreeSWITCH behavior
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
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 know if it was a temporary thing
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 jmesqu...@gmail.com:
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:
On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHVenno.egb...@googlemail.com wrote:
Hi,
sorry - i have to ask again. This is my entry in
/conf/sip_profiles/external/ :
include
gateway name=sipgate.de
/gateway
/include
and it´s still the same... from: 2395...@139.13.37.160
Do you have
On Tue, Aug 18, 2009 at 10:10 PM, NOx-WHVenno.egb...@googlemail.com 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
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjagatculj...@gmail.com 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
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshimayamatake...@gmail.com wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjagatculj...@gmail.com 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
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote:
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
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 mayamatake...@gmail.com wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.com
wrote: Hello, I'm test...
OK
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote:
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
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi mayamatake...@gmail.comwrote:
Hello,
I'm testing FS support for the header Path (FS is behind opensips).
It pretty much works: I tested calling from one user
On Sun, Sep 6, 2009 at 12:13 AM, Woody Dickson woodydick...@gmail.comwrote:
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
On Sat, Sep 5, 2009 at 5:36 PM, Juan Backson juanback...@gmail.com 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
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris m...@jerris.com 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, mayamatakeshi mayamatake
On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins m...@freeswitch.org 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
On Sun, Oct 11, 2009 at 12:18 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins m...@freeswitch.orgwrote:
FYI,
The FreeSWITCH devs have added valet parking! Check it out:
http://www.freeswitch.org/node/207
Let us know what you think
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.com 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
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.
for 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.
Can someone confirm what is the expected behavior
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 br...@freeswitch.org wrote:
have you updated to the latest SVN?
/b
On Oct 28, 2009, at 11:35 AM, mayamatakeshi
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
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 m...@freeswitch.org wrote:
On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi mayamatake...@gmail.comwrote:
About mod_fifo, it would be safe to use it in multi-tenancy scenarios
where domains are created and deleted all the time
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 fr...@impactfax.comwrote:
Yes. I am using version 5.1 I am using Fedora 12.
-Original
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
/Sofia.conf.xml#sql-in-transactions
On Thu, Dec 17, 2009 at 9:53 AM, mayamatakeshi mayamatake...@gmail.comwrote:
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
32 matches
Mail list logo
