Hi,
I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe
3.4.4 drivers. But now I have another problem...
I want to originate calls through event socket, and I only want to receive
ANSWERED(+OK) reply when the user actually answers.
Now the situation is:
Hi guys,
I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.
Thanks,
Diego
On Thu, May 7, 2009 at 6:08 AM, Saeed Ahmedsaeedahmad1...@gmail.com wrote:
Thanks Seven
Diego Viola diego.vi...@gmail.com wrote:
I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.
This was mentioned on the list once before, and it might be what you want:
Hi,
I dont know how can I start Freeswitch using Intercom device, can you help
me on this? Is there an alternative software like X-Lite but only when I
press call on the Intercom device?
Thanks,
Edmar
--
View this message in context:
Looks nice, is anyone running that in production?
On Sun, Aug 23, 2009 at 3:08 AM, Jason White ja...@jasonjgw.net wrote:
Diego Viola diego.vi...@gmail.com wrote:
I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
Thanks a lot, moy, this is great. I'll check to see if there's somewhere I
can test it.
On Sun, Aug 23, 2009 at 2:40 AM, Diego Viola diego.vi...@gmail.com wrote:
Nice work, keep up the great work :).
On Fri, Aug 21, 2009 at 6:29 PM, Moises Silvamoises.si...@gmail.com
wrote:
So, I finally
I have a Freeswitch setup working on an Alix embedded platform in
conjunction with a USB FXO device from Sangoma. My goal is to be able
to either answer incoming calls on a softphone or on a POTS handset
elsewhere in the building, and to also be able to make outgoing calls
from either. For
Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
then went to install FreePBX v3, I've gotten all the prerequisities in the
wizard fixed except for ESL
As per
http://wiki.freeswitch.org/wiki/Event_Socket_Library
http://wiki.freeswitch.org/wiki/Event_Socket
I go into
Hello Anthony,
I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However ptime is set to 20 msec as shown in the Logs:
2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
s=call
c=IN IP4 217.xx.xx.xxx
t=0 0
m=audio 7078
I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable
endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client,
2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian
s60) and an O2 Xda Flame (windows mobile 5).
All 3 endpoints are
This is because you didn't install the zrtpagent.lua script and dial
zrtp on your keypad to enroll the FS box as a trusted man in the
middle... which btw will only work with the unreleased zfone3.
/b
On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote:
I've got 1.0.4 running with zrtp on
Brian, okay, that answers the case with FS acting as a trusted man in the
middle, but what about in the peer to peer case? Shouldn't FS just be passing
the ztrp traffic through to the endpoints? Or am I misunderstanding how it's
supposed to work?
Secondly where would I find info about
On Aug 23, 2009, at 5:39 PM, Harondel J. Sibble wrote:
Brian, okay, that answers the case with FS acting as a trusted man
in the
middle, but what about in the peer to peer case? Shouldn't FS just
be passing
the ztrp traffic through to the endpoints? Or am I misunderstanding
how it's
On 23 Aug 2009 at 17:48, Brian West wrote:
Nope. FreeSWITCH is a b2bua so the traffic is decrypted... and
Hadn't heard that term before
http://en.wikipedia.org/wiki/Back-to-back_user_agent
that clears it up. Any plans to offer straight proxy/passthru?
relayed and encrypted again to
Could you load freeswitch with a couple hundred calls then run the
test again.. and do the same to asterisk and see how the numbers stack
up then? I'm just curious to see what happens at that point.
-Ray
On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
Hi Everyone,
I'm working on a PBX
Hi,
I don't see how I can read some responses to command using esl.
I.E. esl_send_recv(handle, api show calls count\n\n);
and
printf(Header Test %s\n, esl_event_get_header(event, API-Command));
printf(Body Test %s\n, esl_event_get_body(event));
the header details are returned.
The body is
Don't forget to press tab at the asterisk console! :)
On Sun, Aug 23, 2009 at 18:23, Raymond Chandler
intralan...@freeswitch.orgwrote:
Could you load freeswitch with a couple hundred calls then run the test
again.. and do the same to asterisk and see how the numbers stack up then?
I'm just
I can confirm that it works 100% correct passing the SAS across the
bridge correctly once you trust the switch in the middle.
/b
On Aug 23, 2009, at 6:16 PM, Harondel J. Sibble wrote:
ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add
that into
the mix and cross my fingers
On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:
Hi,
I don't see how I can read some responses to command using esl.
I.E. esl_send_recv(handle, api show calls count\n\n);
and
printf(Header Test %s\n, esl_event_get_header(event, API-
Command));
printf(Body Test %s\n,
On 23 Aug 2009 at 16:16, Harondel J. Sibble wrote:
ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into
the mix and cross my fingers that zfone3 gets released soon along with it's
inclusion into the softphones I have on my smartphone devices.
Well good news for the
Wish they would send me one for my E63 for testing... only been
working with zfone 3 so far.
/b
On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:
Well good news for the Tiviphone client
___
FreeSWITCH-users mailing list
On 24/08/09 11:50 AM, Brian West wrote:
Even if I could read some comments from a usage of it would be useful.
I just find it interesting you're doing this with C.
:) The rest of the application is in C, so it makes sense to use
FreeSwitch's esl in C.
Thanks for your help man will let you
They have a trial version of the full client available from their website, I
think it's only for 10 days though. I suspect if you approached them, they'd
probably give you a full client for permanent use for interoperability
testing.
http://www.tivi.com/en/download/credit_paypal.php
Opps, my
On Aug 23, 2009, at 7:20 PM, Harondel J. Sibble wrote:
added following line to
/usr/local/freeswitch/conf/autoload_configs/lua.conf.xml
WRONG.
!--param name=startup-script value=zrtp_agent.lua/--
Don't touch this.
under this section
!--
The following options identifies
On 23 Aug 2009 at 19:53, Brian West wrote:
Just put it into the scripts folder and run it from the dialplan see
default configs.
!-- install zrtp_agent.lua into scripts (ZRTP == 9787) --
extension name=zrtp_enrollement
condition field=destination_number
Ahh, I didn't quite clue-in that I had to run the additional installers as
below when the main compile finished, I thought it was saying it had already
done that.
Makes perfect sense in hindsite ;-)
+ FreeSWITCH install Complete --+
+ FreeSWITCH has been successfully
Whoah.
2009-08-23 20:07:52.583524 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.73.45 [4ff9d452-905b-11de-8c5d-d333d780ffc7]
2009-08-23 20:07:52.740094 [INFO] mod_dialplan_xml.c:315 Processing 1001-
9787 in context default
2009-08-23 20:07:52.980164 [NOTICE]
Just a quick note, and I'm sure why, but screaming monkeys does not play
on the the default installation.
I have not looked into why, but thought I would just quickly let you
know.
Perhaps I have not done something?
regards,
sbt
___
FreeSWITCH-users
On Aug 23, 2009, at 10:26 PM, Scott Torr wrote:
Just a quick note, and I'm sure why, but screaming monkeys does not
play
on the the default installation.
It requires internet connectivity. It calls a remote system to play
which is out of our control.
I have not looked into why, but
If I understand your issue correctly, it sounds to me like FS is not
set to anchor the RTP media stream. Experience suggests that most
SBC's do not like trying to loopback RTP traffic to themselves.
Check to see what IP address's are getting used in the
c=xxx.xxx.xxx.xxx for the INVITE and 200OK
In your SIP profiles this could be set. I beleive 120 is the default setting.
param name=session-timeout value=120
On Fri, Aug 21, 2009 at 11:35 PM, bakkoasannu...@gmail.com wrote:
Do you have those lines in switch.conf file?
!--RTP port range --
param name=rtp-start-port value=1/
Hey there, FsGui uses ESL a lot and I had to go through the code to document
it so here is a few hints inline ...
Don't hesitate to keep the questions coming. I will fill in whenever I can.
jmesquita
On Sun, Aug 23, 2009 at 8:50 PM, Brian West br...@freeswitch.org wrote:
On Aug 23, 2009, at
X-Lite does support TCP, however you need to have NAPTR and SRV DNS
entries. It used to support TLS, but this seems to have been removed
:(
sip.mydomain.net. IN NAPTR 0 0 s SIPS+D2T _sips._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 1 0 s SIP+D2T _sip._tcp.sip.mydomain.net.
Thanks Andrew and Anthony,
I created a ramdisk for the db and log directories using tmpfs and now
I see better performance times:
startup:15.6 sec.
call extension: 0 sec.
shutdown: 7.5 sec
reload config: 0 sec.
I have noticed that during the startup there is a
Well if you append ;transport=tcp on the bridge lines it will use TCP
IMHO this statement needs some clarification based on the context of
this thread.
If the destination is another PBX or Freeswitch box then this is ok as
FS will be initiating the TCP connection.
For terminating calls to
freeswitch -nonat
/b
On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:
2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for
PMP [general error]
2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP
It already does exactly this.
/b
On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:
For terminating calls to registered User Agents (UA) the decision to
use TCP or UDP should be made using information collected when the UA
registered. i.e if the UA registers using TCP, FS should use TCP to
I always expected it did :)
My point was that you cannot put transport=TCP on a bridge statement
line to an internal registered client and expect it to use a protocol
that was not used at registration.
Hence the clarification based on the context of the thread :)
On Mon, Aug 24, 2009 at 2:53
Thanks Brian, now the startup time is 3 sec.
On Aug 24, 2009, at 1:52 AM, Brian West wrote:
freeswitch -nonat
/b
On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:
2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for
PMP [general error]
2009-08-24 04:39:29.910694 [DEBUG]
On 23 Aug 2009 at 20:22, Harondel J. Sibble wrote:
Whoah.
I get audio now, but it's running really slooowly. I'd say about
1/4 to 1/8 normal speech speed
Hmmm, using one of my hardphones, specifically the Integrated Networks IN-
1002
2009-08-23 20:46:07.334596 [NOTICE]
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