Dear all,
I am in the process of implementing an IVR using event outbound
socket. In the dial plan I used the following statement.
action application=socket data=127.0.0.1:5000 async full/
I implemented a Perl script which is listening on the dial plan
Hello world,
I have two freeswitch servers that I want to interconnect. Each one
of them has their own sip domain and we want to enable sip
communications between them. All the PC can ping together.
Client #1 --- Freeswitch #1 --(gateway)--- Freeswitch #2 --
Client# 2.
Michael Collins a écrit :
A 404 in SIP is just like a 404 when web surfing: the target server
can't find whatever it is that you're looking for. In other words,
your FS server made contact with the server at the far end, told it
what endpoint you're looking for, and the server there said,
Gregory Charles gregory.char...@sogeti.com wrote:
2009-07-29 17:20:18 [ERR] sofia_glue.c:568
sofia_glue_ext_address_lookup() STUN Failed!
stun.freeswitch.org:3478 [Timeout]
Set external_sip_ip and external_rtp_ip to something reasonable, e.g.,
$${local_ip_v4} in vars.xml, or change it in
Hi,
I am trying to use the fifo app, but I am hitting on the following error.
Does anyone know how to resolve the No code is found error?
Dialplan: sofia/internal/1...@192.168.1.102 Regex (PASS) [internal-call]
destination_number(5501) =~ /^(.*)$/ break=on-false
Dialplan:
Are you using late neg. or proxy media? Also answer the call before
you put it into the fifo. Notice it only pre_answers the call.
/b
On Jul 30, 2009, at 4:17 AM, Juan Backson wrote:
Hi,
I am trying to use the fifo app, but I am hitting on the following
error. Does anyone know how to
Hello,
I have had great success bringing up Freeswitch in my lab with various
makes/models of SIP hardware CPE, and I am starting to delve into some more
complex scenarios. I have two questions specific to Cisco's current generation
IP handsets (7965/7945):
1. Is integration of busy lamp
I don't think it will work the format is slightly different from the
standard last I checked and I couldn't get it working.
/b
On Jul 30, 2009, at 12:28 AM, Pat Jensen wrote:
Hello,
I have had great success bringing up Freeswitch in my lab with
various makes/models of SIP hardware CPE,
Hi Mike,
I configured the following lines into intenal.xml
param name=manage-presence value=true/
param name=manage-shared-appearance value=true/
param name=dbname value=share_presence/
param name=presence-hosts value=$${domain}/
SIP presence signalling is no stille take
We have a carrier that we receive calls from over SIP. They are getting
the call from the PSTN and then sending to us as sip. We then get other
calls to our FS via sip from other carriers that are also from the PSTN.
We bridge these calls in FS. All connections to FS are sip. So.
On Jul 30, 2009, at 8:27 AM, DA wrote:
We have a carrier that we receive calls from over SIP. They are
getting
the call from the PSTN and then sending to us as sip. We then get
other
calls to our FS via sip from other carriers that are also from the
PSTN.
We bridge these calls in
David Knell wrote:
On Thu, 2009-07-30 at 09:21 +0800, Steve Underwood wrote:
High quality conferencing is a difficult task, and still a research
topic. No two conferencing systems perform alike. The interesting thing
about this and other reports is that the conferencing in Freeswitch
Hi.
I'm trying to receive a fax using Freeswitch. It's a SIP channel. I know
there's no T38 support (yet) but I think I can get decent result with
G711 (as I got with Asterisk and Callweaver on non-T38 providers).
In FS, all fails just some seconds after the channel has been answered.
Here's the
We may have seen this also once before.
But in this case, it is 'needed' (or would be helpful) because the telco
is not doing the EC correctly.
I can see how the originating telco should be the one to fix the
problem. But it always seems that is easier said than done.
Is there no way at all
Hi,
This might be Sangoma config issue, so apologies in advance for
posting it here if it is. I am waiting for Sangoma helpdesk to get
back to me!
But I have a Sangoma a101 and trying to get it working with
Freeswitch. Have E1 line coming from telco and everything set up
correctly (to my best
Its not technically feasible to do so its also resource intensive for
something that you shouldn't have to be dealing with in the first
place. Take a cluebat and beat your telco if they don't fix it.
/b
On Jul 30, 2009, at 9:26 AM, Frank @ Impact wrote:
We may have seen this also once
While it might not be feasible it may be possible (and I don't see why not).
There are many, many commercial devices that claim to do this. Then
again there are just as many devices that claim to do everything and
anything...
I agree with Brian. It shouldn't be something you have to deal with.
You need
param name=inbound-late-negotiation value=true/
In your SIP profile in order to be able to use those channel variables.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Am 30-Jul-09 um 10:16 AM schrieb Stefano
Mathieu Rene ha scritto:
You need
param name=inbound-late-negotiation value=true/
Yes, I've tried both with that and without, but no differences.
Stefano
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are you answering the call somewhere or running any other dialplan
actions that may establish media before you set proxy?
MIke
On Jul 30, 2009, at 11:05 AM, Stefano Marinelli wrote:
Mathieu Rene ha scritto:
You need
param name=inbound-late-negotiation value=true/
Yes, I've tried both
The far end is sending a bye for no clear reason I can see in this log.
Mike
On Jul 30, 2009, at 10:15 AM, Stefano Marinelli wrote:
Hi.
I'm trying to receive a fax using Freeswitch. It's a SIP channel. I
know
there's no T38 support (yet) but I think I can get decent result with
G711 (as
On Thu, Jul 30, 2009 at 3:26 AM, Brian West br...@freeswitch.org wrote:
Remove async from the options list on the socket.
See the notes and info about async mode on this page:
http://wiki.freeswitch.org/wiki/Event_socket_outbound#Configuration
It should help shed some light on the various
Michael Jerris ha scritto:
are you answering the call somewhere or running any other dialplan
actions that may establish media before you set proxy?
Yes...there was an answer...now I've removed it and the channel is
proxied in the right way.
Sending faxes from the outside is ok but when I
i think you mean
$conn-setEventLock(true);
which should only have to be set once and all the commands will then queue
up until you set it back to false
On Thu, Jul 30, 2009 at 10:25 AM, Michael Collins m...@freeswitch.orgwrote:
On Thu, Jul 30, 2009 at 3:26 AM, Brian West br...@freeswitch.org
Kristian Kielhofner wrote:
While it might not be feasible it may be possible (and I don't see why not).
There are many, many commercial devices that claim to do this. Then
again there are just as many devices that claim to do everything and
anything...
People tell me commercial offerings
TTNC - Adnan Barakat wrote:
Is there a way to terminate the C leg when using att_xfer if the C leg
ends up being a voicemail?
Any ideas anyone??
Thanks
Adnan
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i think you dial # instead of 0
please do not more emails to the same thread asking ppl to answer sooner.
On Thu, Jul 30, 2009 at 12:23 PM, TTNC - Adnan Barakat techni...@ttnc.co.uk
wrote:
TTNC - Adnan Barakat wrote:
Is there a way to terminate the C leg when using att_xfer if the C leg
FYI,
BandwidthU's interview with Anthony has been posted. Click
herehttp://digg.com/d3yzzjto visit part 3 of the interview which has
links to parts one and two. Also,
be sure to Digg it!
Thanks,
Michael
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Hello all!
Please visit the ClueCon website http://www.cluecon.com for information
about the meet and greet on Tuesday, August 4th being sponsored by iCall.
Please be sure to RSVP at the link provided so that the great folks at iCall
know how many people they'll be hosting.
Can you believe it's
Hi,
I have a small JS script that calls a phonenumber, when the call is answered
it plays a wave file, then it calls a second phonenumber and bridges the
calls. Is it possible to make wave-playing async, so that the second call is
generated as soon as the first is picked up? Right now the wave
If you are using centOS for your Freeswitch installation, you should
probable read the article on planet.centos.org and the www.centos.org
http://www.centos.org/ the open letter to Lance Davis one of the
founders of centOS.
CONFIDENTIAL NOTICE : If you have received this email in error,
if you want async use the socket app with ESL none of the scripts can do it.
On Thu, Jul 30, 2009 at 1:38 PM, Matthew Fong mattdf...@gmail.com wrote:
Hi Nicolas,
I believe you could bridge the call first, ignore early media, and then use
uuid_displace...
there might be a better way tho,
On Thu, Jul 30, 2009 at 12:36 PM, Nicolas Brenner nico...@medularis.comwrote:
Thanks, I'll try that.
How can I play a wav to an active call through the socket?
When you are on a socket you can do just about anything you could do at the
CLI. Look at all the uuid_XXX commands. Example:
Matthew, Anthony and Michael, thank you very much, seems like you gave me
exactly the info I needed!
On Thu, Jul 30, 2009 at 4:25 PM, Michael Collins m...@freeswitch.org wrote:
On Thu, Jul 30, 2009 at 12:36 PM, Nicolas Brenner
nico...@medularis.comwrote:
Thanks, I'll try that.
How can
Hello,
I built libesl and JAVA mod. (make and make javamod). But when I try to run a
JAVA script with the following code
ESLconnection connection = new ESLconnection(127.0.0.1, 9000, );
ESLevent events = connection.getInfo();
System.out.println(events.toString());
I get the following error
Thank you Anthony.
--- On Thu, 7/30/09, Anthony Minessale anthony.miness...@gmail.com wrote:
From: Anthony Minessale anthony.miness...@gmail.com
Subject: Re: [Freeswitch-users] JAVA ESL
To: freeswitch-users@lists.freeswitch.org
Received: Thursday, July 30, 2009, 9:00 PM
it might be a build
Hey,More noob questions,
Background: I have a server running FS and clients connecting to the server.
I have re-installed FS so that i could start from scratch and have LAN calls
working (using default settings and users like 1014 and 1015). I have
incoming call going to the server, and I can
On Thu, Jul 30, 2009 at 7:04 PM, Drew Hopcroft hopcroft.d...@gmail.comwrote:
Hey,More noob questions,
Background: I have a server running FS and clients connecting to the
server. I have re-installed FS so that i could start from scratch and have
LAN calls working (using default settings and
CentOS has been a trusted platfrom for me from last 3+ years. I have
developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in
production environment. I really wish and hope this great project continues.
I
If I missed out the async mode from the dial plan I am not able to get the
events from it.
Here is my Perl script.
# Create a conenction with Event socket library.
my $conn = new ESL::ESLconnection($fd);
# Getting the connection informations and values of the variables.
my $info =
On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
CentOS has been a trusted platfrom for me from last 3+ years. I have
developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
boxes, and 27 out of 49 Asterisk box are still ruining on CentOS
Please read my email as,
CentOS has been a trusted platfrom for me from last 3+ years. I have
developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
boxes, and 27 out of 49 Asterisk box are still *running* on CentOS in
production environment. I really wish and hope this
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