I believe, I'm looking for a radius equivalent of the userfield
(http://wiki.freeswitch.org/wiki/Mod_cdr_csv#userfield)
br
On Wed, Aug 19, 2009 at 9:11 PM, Hristo Trendevdist.li...@gmail.com wrote:
Hi,
Does anyone know of a way to add custom information in AVP in radius
accounting messages?
I apologize in advance if this message duplicates a previous post or the topic
has been covered elsewhere. If that's the case, a pointer would be appreciated!
I have multiple offices each running Freeswitch and a number of Polycom phones
and softphones.
I would like to share the presence
Hi Kenneth,
I'm not going to answer your question! Instead I would like to emphasize on
the thing you are going to achieve because some times ago I've post this
question in some other title but unfortunately did not get any answer. As
the SIP protocol's point of view you should be able to
Dear All,
my telco can provider SS7 (Now i use ISDN PRI). I have
question about SS7.
1. Can i setup SS7 with out ss7box ?
2. I found some tutorial about libss7 in Asterisk. Is FS
support all feature like that.
Best Regards.
Dome C.
On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:
1. Can i setup SS7 with out ss7box ?
Sangoma's ss7box is the only solution right now thats turn key.
2. I found some tutorial about libss7 in Asterisk. Is FS
support all feature like that.
You can't really use libss7 in
Hi,
We have an application written in ANSI C which currently talks to the
Asterisk Manager to make phone calls.
We're possibly looking at converting this to FreeSwitch.
At the moment it has an abstraction layer from Asterisk and speaks to
between 1 and 80 Asterisk machines using round robin
Hi,
I would recommand mod_event_socket, also look at libs/esl (event
socket library), a C library that takes care of all the parsing to
talk to mod_event_socket.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On
Matt,
Event Socket would be perfect for your needs. If you run into any
troubles don't hesitate to join #freeswitch or ask on the list to help
you get on the right track.
Thanks,
/b
On Aug 20, 2009, at 4:35 AM, Matt Riddell wrote:
Would you recommend that I use the mod_event
On 21/08/09 1:26 AM, Mathieu Rene wrote:
Hi,
I would recommand mod_event_socket, also look at libs/esl (event
socket library), a C library that takes care of all the parsing to
talk to mod_event_socket.
Awesome thanks man :)
--
Cheers,
Matt Riddell
Director
On 21/08/09 1:26 AM, Brian West wrote:
Matt,
Event Socket would be perfect for your needs. If you run into any
troubles don't hesitate to join #freeswitch or ask on the list to help
you get on the right track.
Cheers, will do!
--
Cheers,
Matt Riddell
Director
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number (conference, mailbox) i hear a
very choppy voice coming from the fritzbox side. I think it may have to
do with the ptime 20msec/30msec.
Example: When calling from the fritzbox to a
This is a bug in the fritzbox... you have to set your codec neg. to
greedy on the sofia profile and that should fix it.
/b
On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number
On Thu, 2009-08-20 at 21:35 +1200, Matt Riddell wrote:
We have an application written in ANSI C which currently talks to the
Asterisk Manager to make phone calls.
We're possibly looking at converting this to FreeSwitch.
At the moment it has an abstraction layer from Asterisk and speaks to
Mathieu,
have to confess- you are right! uuid_bridge works as expected. Usual saying
- is didn't work last time I tried!
Anyway, thank you much!
Artem
On Wed, Aug 19, 2009 at 5:02 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
Hi,
Eavesdrop kind of works yeah, you can use the intercept
Besides taking a hammer to it? Have you tried to make sure you have
the latest firmware? Try setting the ptime on the fritz to 20ms?
I really can't trust a product that has fritz in its name... it might
go on the fritz :P pun intended.
/b
On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
2009/8/18 Christian Löschenkohl christian.loeschenk...@xpirio.com
hi
does anybody know how to send a sip options message to a registered user,
using the event socket
or something else build in freeswitch
i think the ping parameter does something like this for gateways.
what i want/need is
For my gateway, I had to set ignore_early_media=true.
From: Mathieu Rene
Sent: Thursday, August 20, 2009 7:00 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls withsame
Gateway
Hi
How are you bridging the calls in FS (which
I have been experiencing this as well. It happens randomly and I
haven't been able to find out what the issue is. I think there is some
delay when the RTP ports are being negotiated/allocated. Or something.
What helped me a bit: I start with playing a file containing 1 second
of silence and only
2009/8/20 Brian West br...@freeswitch.org:
On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:
1. Can i setup SS7 with out ss7box ?
Sangoma's ss7box is the only solution right now thats turn key.
If i seting up asterisk with libss7. should be same ss7box right ?
2. I found
originate
{ignore_early_media=true}sofia/gateway/epik.com/914159927717bridge(sofia/gateway/
epik.com/914154650027)
is the string I was using from the console.
On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene mrene_li...@avgs.ca wrote:
Hi
How are you bridging the calls in FS (which api call or C
Hello!
I am using Callcentric for my tests and have observed what appears to me
a possible bug in the way Freeswitch handles DNS SRV records.
Callcentric uses DNS SRV records as a way to direct traffic to their SIP
server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:
On Aug 20, 2009, at 1:28 PM, Carlos S. Antunes wrote:
Hello!
I am using Callcentric for my tests and have observed what appears
to me
a possible bug in the way Freeswitch handles DNS SRV records.
Callcentric uses DNS SRV records as a way to direct traffic to their
SIP
server. A 'srv'
On 08/20/2009 05:22 AM, Michael Collins wrote:
There is no noise on those 3 beeps. In fact, for something that's been
through ulaw/alaw compression those beeps are very clean. They are
quite
short, though.
Heck yeah they're short! Steve, in your experience is there a
Check out mod_event_multicast. I think you should be able to share
using that to share events between boxes (although I have never tried
it).
Mike
On Aug 20, 2009, at 4:29 AM, afshin afzali wrote:
Hi Kenneth,
I'm not going to answer your question! Instead I would like to
emphasize on
Oh, I missed that one.
param name=nat-options-ping value=true / (in the sip profile)
It does it only when NAT is detected though.
Tschuess,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 20-Aug-09, at 2:27 PM, Michael
Or as I have argued today they should fix their SRV records to be zero
weighted.
/b
On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:
You can bypass the srv records if you like by passing a :port with the
hostname where you use it in freeswitch.
Raymond Chandler wrote:
very true, but i've been reading over the RFCs on this, and it seems
that FreeSWITCH isn't doing anything incorrectly.
in RFC3263 (section 4), when talking about client usage of SRV:
The procedures here MUST be done exactly once per transaction, where transaction is as
Agreed. That being said, having a way to force Freeswitch to stick
to the same IP address in the middle of authorization/authentication
wouldn't violate any specs but would certainly make things easier
when dealing with not so well implemented round robin scenarios. Do
you think a new
Hello Brian,
yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
are widely spread here in Germany. I know of a SIP provider who has 5
Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
Germany, and they are covering a big stake of in the market. So they
Actually, disregard my previous mail... this patch probably wouldn't
be gladly accepted... I seem to have forgotten about the part where
the DNS SRV lookup MUST be done once per transaction
So if we don't do that, then we would be breaking spec. Just disabling
SRV on the profile to
Raymond Chandler wrote:
Actually, disregard my previous mail... this patch probably wouldn't
be gladly accepted... I seem to have forgotten about the part where
the DNS SRV lookup MUST be done once per transaction
Hmm, where does it say that, after the lookup, one cannot use the
On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
Hmm, where does it say that, after the lookup, one cannot use the same
IP address as before? :)
Section 4 of RFC3263 as quoted in my first email
The procedures here MUST be done exactly once per transaction, where
transaction is as
Raymond Chandler wrote:
On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
Hmm, where does it say that, after the lookup, one cannot use the same
IP address as before? :)
Section 4 of RFC3263 as quoted in my first email
The procedures here MUST be done exactly once per transaction,
Raymond Chandler wrote:
On Aug 20, 2009, at 5:25 PM, Carlos S. Antunes wrote:
Raymond Chandler wrote:
On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
Hmm, where does it say that, after the lookup, one cannot use the same
IP address as before? :)
Section 4 of RFC3263 as quoted in my
Matt,
For your information the tones you gave me are exactly 738Hz. If you
want to try that tone detection thing.
Cheers.
Eric des Courtis
On Thu, Aug 20, 2009 at 2:20 PM, Michael Collinsm...@freeswitch.org wrote:
On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood ste...@coppice.org
wrote:
Matt,
As is mod_vmd will not detect tones shorter then 138ms. However I
could get that value down to ~30ms at best by making a few
modifications to the algorithm.
Cheers.
Eric des Courtis
On Thu, Aug 20, 2009 at 7:51 PM, Eric des
Courtiseric.des.cour...@gmail.com wrote:
Matt,
For your
I changed
/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a real-world environment. at 4000 it fired the event like 5
times in a session, but 6500 only once. Do you think I should expect a lot
of false
Matt,
I think the only way to know for sure is to try it. I would try to get
the value as high as possible while still detecting that 738Hz sine
(with a small margin of error). Lowering the value increases false
positives rapidly.
Eric des Courtis
On Thu, Aug 20, 2009 at 8:36 PM, Matthew
On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fongmattdf...@gmail.com wrote:
/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a real-world environment. at 4000 it fired the event like 5
times in a session,
On Thu, Aug 20, 2009 at 9:16 PM, Gabriel Gundersong...@gundy.org wrote:
On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fongmattdf...@gmail.com wrote:
/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a
Hi,
I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
I need list of application variables so I can easily translate my asterisk
configuration in FreeSwitch. Like I don't know how I translate my dial plan
in FreeSwitch as I like to forward it to another context as sample
For dialplan apps take a look at.
http://wiki.freeswitch.org/wiki/Mod_dptools
For channel variables:
http://wiki.freeswitch.org/wiki/Channel_Variables
For FS commands:
http://wiki.freeswitch.org/wiki/Mod_commands
You could use transfer in order to send a call to another extension/context.
Let us know if you have more questions or need more help :)
On Fri, Aug 21, 2009 at 1:35 AM, Diego Viola diego.vi...@gmail.com wrote:
For dialplan apps take a look at.
http://wiki.freeswitch.org/wiki/Mod_dptools
For channel variables:
http://wiki.freeswitch.org/wiki/Channel_Variables
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