Re: [Freeswitch-users] Callback to the user in ESL
In the previous reply you told me to use new OUTBOUND connection.
But in this post you mention INBOUND connection.
That confusion only made me to ask the question once again. Pardon me if I
made any mistake.
Making a new inbound connection does the task. Thanks for that.
On Sat, Nov 28, 2009 at 12:49 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I told you to make a new separate inbound connection back to the server
from your script, do not use the same one thta was tethered to the call
because its too late to use that one.
Why do I have to answer you twice?
On Thu, Nov 26, 2009 at 3:27 AM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
Hi, Any help or suggestion regarding my previous post. Especially
I also noted that, if I don't receive any events, especially
SERVER_DISCONNECTED, then the connection is in established state, but once
I receive the SERVER_DISCONNECTED event, the connection is closed. Is it
correct??
Here is the program by which I confirmed the above!
require ESL;
use IO::Socket::INET;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort = '8447',
Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $con;
my $type = user/;
for(;;) {
# wait for any client to connect, a new client will get connected
when a new call comes in the dialplan.
my $new_sock = $sock-accept();
# Do fork and let the parent to wait for more clients.
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
# Extract the host of the client.
my $host = $new_sock-sockhost();
# file descriptor for the socket.
my $fd = fileno($new_sock);
print Host name is $host\n;
# Create object for the ESL connection package to access the ESL
functions.
$con = new ESL::ESLconnection($fd);
# Gets the info about this channel.
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid,
$info-getHeader(caller-caller-id-number),
$info-getHeader(caller-destination-number);
# Answer the channel.
$con-execute(answer);
# Set the event lock to tell the FS to execute the instructions in
the given order.
$con-setEventLock(true);
# Play a file Get the personal number from the user.
$con-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav);
$con-execute(hangup);
while($con-connected())
{
my $e=$con-recvEvent();
my $ename=$e-getHeader(Event-Name);
print $e-serialize();
print $ename\n;
print Connection exists\n;
sleep(1);
}
print
Bye\n--\n;
close($new_sock);
}
I've not registered for any events.
In the above program I'm receiving the SERVER_DISCONNECTED event.
Output when receiving event:
Host name is 192.168.1.222
Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
Event-Name: SERVER_DISCONNECTED
SERVER_DISCONNECTED
Connection exists
Bye
When I comment the recvEvent line, I got the following output.
Host name is 192.168.1.222
Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
Connection exists
Connection exists
Connection exists
Connection exists
Connection exists
On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
I've tried the following program as per the suggestion that you've told.
But it seems, no success. Once the connection is closed, I created a new
connection and I send originate to originate a new call. But it is not
working.
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort =
'8447', Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $make_call;
my $con;
my $type = user/;
for(;;) {
my $new_sock = $sock-accept();
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
my $host = $new_sock-sockhost();
my $fd = fileno($new_sock);
$con = new ESL::ESLconnection($fd);
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid,
$info-getHeader(caller-caller-id-number),
$info-getHeader(caller-destination-number);
$con-filter(Unique-Id, $uuid);
$con-events(plain, all);
$con-execute(answer);
$con-setEventLock(true);
my $number=$con-execute(read,2 4
[Freeswitch-users] errors installing wanpipe drivers
Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: cd wanpipe-3.5.6.5/ make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] errors installing wanpipe drivers
I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. Then you can go back to freeswitch and build the mod_openzap/libopenzap. François On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel wrote: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: cd wanpipe-3.5.6.5/ make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] errors installing wanpipe drivers
make openzap is the correct way to build when using with openzap/freeswitch. If you are having issues with this you should check with sangoma support as to why that build of the drivers is not supporting it properly and what version you should be using. Mike On Nov 30, 2009, at 5:41 AM, François Legal wrote: I did manage to build these drivers, but maybe you're not doing it the right way. Sangoma document state that the drivers should be built by using their ./Setup script that does all that is required. I did use ./Setup install which builds the kernel modules, the wanrouter utilities and install all the required stuff. Then you can go back to freeswitch and build the mod_openzap/libopenzap. François On Mon, 30 Nov 2009 15:19:43 +0530, Neil Patel ne...@cs.stanford.edu wrote: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: cd wanpipe-3.5.6.5/ make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Thanks, Neil ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] DTMF Digits Lost when Under Load
Hi All,
Thought I would share my solution to this DTMF problem: it turns out my
ISP was capping my bandwidth dropping packets to keep the connection
1Mbps, so the experienced DTMF loss was actually packets being discarded.
On my way to this discovery I tested Freeswitch DTMF quite thoroughly
never actually found any problems even at hundreds of concurrent calls.
Here is how I tested, who knows this might be useful to someone:
- I used SIPp to generate calls a Python script to log the received
DTMF digits
- SIPp command line:
- sipp -sf dtmfSenario.xml -d 1 -s 451 -l 96 -mp 5606 -i
xxx.xxx.xxx.xxx
- dtmfSenario.xml below
- Dialplan:
- extension name=test_dtmf_capture_test
!--Grab calls for dialing --
condition field=destination_number expression=(^100100$)
action application=answer/
action application=python data=writeDtmfStats/
/condition
/extension
- Python:
- import sys
from freeswitch import *
def get_number(session,invalid,num=20):
digits = session.getDigits(num, , 15000)
consoleLog(info,Got '%s' digits from user.\n % digits)
if digits == '':
# Invalid call
if invalid == 3:
consoleLog(info,Three invalid
attempts!!\n)
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav)
session.hangup()
sys.exit(0)
else:
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav)
get_number(session,invalid + 1)
else:
consoleLog(info,Got a valid number: %s,
proceeding...\n % digits)
return digits
def handler(session, args):
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
numberToDial = get_number(session,2,num=10)
consoleLog('info','Got 10 DTMF digits. Writing 1 to file...\n')
fo = open('/tmp/dtmfData.csv','a')
fo.write('1\n')
fo.close()
# Do some stuff wait for SIPP to hangup
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav)
return
- DTMF senario file:
- # cat dtmfSenario.xml
?xml version=1.0 encoding=ISO-8859-1?
!DOCTYPE scenario SYSTEM sipp.dtd
!-- This program is free software; you can redistribute it
and/or --
!-- modify it under the terms of the GNU General Public License
as --
!-- published by the Free Software Foundation; either version 2 of
the --
!-- License, or (at your option) any later
version.--
!--
--
!-- This program is distributed in the hope that it will be
useful,--
!-- but WITHOUT ANY WARRANTY; without even the implied warranty
of --
!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See
the --
!-- GNU General Public License for more
details. --
!--
--
!-- You should have received a copy of the GNU General Public
License --
!-- along with this program; if not, write to
the --
!-- Free Software Foundation,
Inc.,--
!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307
USA --
!--
--
!-- Sipp 'uac' scenario with pcap (rtp)
play --
!--
--
scenario name=UAC with media
!-- In client mode (sipp placing calls), the Call-ID MUST
be --
!-- generated by sipp. To do so, use [call_id]
keyword.--
send retrans=500
![CDATA[
INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp sip:s...@[local_ip]:[local_port];tag=[call_number]
To: sut sip:[servi...@[remote_ip]:[remote_port]
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 18 100
[Freeswitch-users] park on hook
Hi, Is there anyway to detect when a channel is park in a way that is similar to hangup-hook or answer-hook? I would like to detect that inside a custom mod, without using the event mechanism? woody ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] CLIP on FXS channels with mod_open zap
Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] errors installing wanpipe drivers
On Mon, Nov 30, 2009 at 4:49 AM, Neil Patel ne...@cs.stanford.edu wrote: Hi All, I am currently installing a Sangoma A102 card to work with FS using wanpipe drivers (OS = Ubuntu Jaunty). The problem is I can't get openzap-related modules to compile: cd wanpipe-3.5.6.5/ make openzap ... make[2]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libsangoma' make -C api/libstelephony clean make[1]: Entering directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make[1]: *** No rule to make target `clean'. Stop. make[1]: Leaving directory `/usr/src/wanpipe-3.5.6.5/api/libstelephony' make: *** [all_lib] Error 2 The libstelephony directory has no Makefile in it. Why is it missing? Is there a version of wanpipe drivers that will work? I have been unsuccessful with 3.4.4 and 3.5.6 in similar fashion. Hi Neil, Most likely the creation of the Makefile failed (since you mention you can't see a Makefile). Please be sure to have installed the pre-requisites listed at http://wiki.sangoma.com/Requirements Particularly in this case, libtool, autoconf and automake packages. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap
can you test svn trunk or latest pre release of 1.0.5 On Mon, Nov 30, 2009 at 9:36 AM, François Legal de...@thom.fr.eu.orgwrote: Hello, I'm using Freeswitch with a Sangoma A400 card, and I'm having CLIP problems on the FXS ports. When I ring on FXS ports, the connected phone does not display callerid/callerid-name. I tried turning the stuff of in openzap.conf.xml () but it did not help. As a side note, turning this on on the FXO ports drops the callerid information on incoming calls. Running freeswitch 1.0.4 on linux 2.6.27. François ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Sangoma RTP TAP
Hello, has anyone of you tried the RTP TAP function of sangoma`s wanpipe driver? It is described here: http://wiki.sangoma.com/wanpipe-voice-rtp-tap On my side wanrouter log says that RTP TAB is configured and enabled, but I can't detect any udp packets received by the remote server (which is described by RTP_TAP_IP, RTP_TAP_MAC and RTP_TAP_PORT). I've latest driver and double checked the wanpipe.conf config. I tried to send some udp packets from wanpiping server to remote server, where the packets were shown up via tcpdump. So there is no FW problem involved. Each try to do some kind of printf debugging in wanpipe-driver doesn't succeed. Any ideas? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Accessing custom SIP headers
The correct way to pass non-standard headers is X- not X_ .
action application=set data=accountcodec=${sip_h_X-ACCOUNTCODE} /
On Sat, Nov 28, 2009 at 12:47 PM, Simon Woodhead simon.woodh...@me.com wrote:
Hi folks,
I'm hoping someone can help me get at custom headers in the dial-plan. I've
read about X- headers being accessible but need to get at some X_ headers
passed through from a proxy. Reading the info app docs, the X shouldn't
actually matter but no matter which way I try I always seem to get a null
result.
An example header in an INVITE is:
X_ACCOUNTCODE: XX.
I've tried the following dial-plan structures hoping one might work but none
do:
action application=set
data=accountcodea=${variable_sip_h_X_ACCOUNTCODE} /
action application=set data=accountcodeb=${variable_sip_h_ACCOUNTCODE}
/
action application=set data=accountcodec=${sip_h_X_ACCOUNTCODE} /
action application=set data=accountcoded=${sip_h_ACCOUNTCODE} /
Any help would be much appreciated.
Thanks,
Simon
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--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com
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[Freeswitch-users] Polycom Phones and Domains
I'm attempting to configure several varieties of polycom (SoundPoint IP 550, SoundPoint IP 601) phones to connect to a freeswitch instance using a domain other than default (i.e. the ip address). Everything works wonderfully as long as the domain is named exactly the same thing as the server host provided to the phone (whether that is the server's ip address, or a hostname resolved via dns). As long as those match everything is fine. What I'm trying to sort out is, is it possible to convince the phone to use something other than the server's hostname/ip as the second part of the user name (i.e. u...@host)? Or should I just resign myself to making the domain name some host name that can resolve via DNS? I've tried including @somedomain in the authid field of the line of the phone, but freeswitch reports that the user someu...@somedomain@serverip couldn't be authenticated. It appears that the phone always appends the server name... Does anyone know of a way that I configure a polycom to connect as us...@d1 and another to connect as us...@d2 where d1 and d2 are NOT in DNS or are polycoms just going to always use the server host as the @ part of the username? Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700
I don't quite understand what you are talking about? So you have bypass_media=true and you attempt to make an attended xfer as soon as you complete the transfer according to your trace FS does re-invites to convert the call to be exchanging media with FS. The o= lines you don't like are being set by the anonymous device in your callflow and should not impact anything at all. Are you saying something that used to work suddenly has caused you problems or is this the first time you are trying this because we have tested this scenario many times. Are you getting packet captures also and checking where the media is going after those re-invites? if you are intentionally using an ALG you might try without it because 100% of ALG we have ever seen have been badly broken when working with something like FS. On Sun, Nov 29, 2009 at 6:51 AM, John Platts john_pla...@hotmail.comwrote: To clarify the problem, the invite message is incorrect because comfort noise is being negotiated in the re-invite instead of G.711 or G.729: INVITE sip:19729831...@168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKF1KrDreNFQgaj Max-Forwards: 69 From: John Platts sip:19725357...@168.75.202.212sip%3a19725357...@168.75.202.212 ;tag=c61Drt38KF72m To: sip:19729831...@ipipgw.ipdimensions.comsip%3a19729831...@ipipgw.ipdimensions.com ;tag=2B1339E0-1A2C Call-ID: 1c095553-5741-122d-33a8-00185167f91d CSeq: 123615824 INVITE Contact: sip:mod_so...@168.75.202.212:5062 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15700M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 183 X-FS-Support: update_display Remote-Party-ID: John Platts sip:19725357...@168.75.202.212sip%3a19725357...@168.75.202.212 ;party=calling;screen=yes;privacy=off v=0 o=- 123576 123577 IN IP4 192.168.1.4 s=- c=IN IP4 168.75.202.212 t=0 0 m=audio 30186 RTP/AVP 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 How do I get it to negotiate G.711, G.729, or other codec instead of comfort noise? Our IP phones, our FXS gateways, and our IP to IP gateways expect G.711, G.729, iLBC (if supported by the endpoints), G.722 (if supported by the endpoints), or G.726 (if supported by the endpoints) be negotiated. From: john_pla...@hotmail.com To: freeswitch-users@lists.freeswitch.org Date: Sat, 28 Nov 2009 23:34:24 -0600 Subject: [Freeswitch-users] Call transfer fails in proxy media and bypass media modes in FreeSWITCH revision 15700 I have updated my FreeSWITCH installation to revision 15700. I am experiencing call transfer problems whenever proxy media or bypass media is enabled. When proxy media and bypass media are both disabled, the call transfer does not fail and there are no audio issues. When proxy media mode is enabled, the call stays up after the transfer occurs, but there is no audio flowing on either end of the call. When bypass media mode is enabled, there is no audio flowing on either end of the call, and the call actually gets disconnected. I have collected detailed traces using the TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch command. I have attached a ZIP file named freeswitch-rev15700-traces-112809-2210.zip, which includes the following traces: - freeswitch-rev15700-trace-112809-2210-proxyandbypassoff.txt - A trace with both media proxying and media bypass disabled. The call is being transferred without any problems in this scenario. - freeswitch-rev15700-trace-112809-2210-proxyonandbypassoff.txt - A trace with media proxying enabled and media bypass disabled. Media proxying is enabled for the call legs in this scenario. The call stays up in this scenario, but there is no audio flowing after the transfer completed. In this scenario, FreeSWITCH does not shutdown cleanly, and there is a segmentation violation when FreeSWITCH is terminated. - freeswitch-rev15700-trace-112809-2210-proxyandbypasson.txt - A trace with both media proxying and media bypass enabled. Media bypass is enabled for the call legs in this scenario. The call actually gets dropped and there is no audio after the transfer is completed in this scenario. I have looked over the SIP traces of the failing scenarios. I have caught the following problems in the failing scenarios: - The o= line in SDP descriptors coming from the IP phone contains the private IP address, but the c= line in the SDP descriptors coming from the IP phone contains the public IP address. I have noticed a problem in re-INVITEs being sent from in proxy media and bypass media modes. The c= line in the re-invites contains the private IP address instead of the public IP address. The c= line was modified by a SIP ALG to contain a public IP address, but FreeSWITCH is actually not
Re: [Freeswitch-users] Sangoma RTP TAP
Hello Helmut, On Mon, Nov 30, 2009 at 11:52 AM, Helmut Kuper helmut.ku...@ewetel.dewrote: Each try to do some kind of printf debugging in wanpipe-driver doesn't succeed. Any ideas? The way the rtp tapping works right now is kinda hackish and pretty much Asterisk/Zaptel-based. We depend on the application (either Asterisk or FreeSWITCH) to enable/disable echo cancellation via zaptel commands. When echo cancellation is enabled we assume a call started and enable the tapping, when echo cancellation is disabled we stop the tapping. This behavior has yet to be implemented for FreeSWITCH. An easy way to do it is just to have the wanpipe card to work in zaptel mode and then add a call to zap_channel_command(tech_pvt-zchan, ZAP_COMMAND_ENABLE_ECHOCANCEL) on call start and ZAP_COMMAND_DISABLE_ECHOCANCEL on call stop in mod_openzap.c. The right way to do it is via new API in libsangoma to start tapping and stop tapping. I will add the new libsangoma API to my todo list, hopefully will be done sometime this month. If you want to test the first quick approach send me an off-line message with ssh connection information to get into your box to do these changes so you can test them. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Questions on ISDN support for Freeswitch
Hi to all, shortly i'll make a pbx for a customer that uses a couple of isdn bri lines and, looking for hardware, i've seen that not too much expensive isdn cards that works well are the ones that uses hfc-4/8s controller (specifically i'll use a OpenVox B200P that has 2 ISDN ports and use an HFC-4S controller). I've seen that FreeSwitch doesn't support mISDN but uses openzap (trough ozmod_isdn.so). I got some serious troubles using OpenZAP on analogical lines (bad dtmf recognition on fxs ports (like press 4 and get 44), annoying noise, busy/hangup tone unrecognized [tones, in italy, differs by cadency and not frequency], and more). Using tone_detect i bypassed the tone recognition problem and the noise was however acceptable: the blocking problem was the bad dtmf recognition. In the end (hope god forgive me) i put asterisk as (*1*)(*2*)sip proxy between zaptel and freeswich: i need to fix in config hangup cause detection but it seems to works fine. So my questions are: - Do freeswitch supports mISDN? - If it doesn't support mISDN, tone and dtmf recognition will be done by the isdn card, the kernel module or will be done by openzap? - There are alternative ways to use (*3*) mISDN with freeswitch, apart put asterisk as proxy? Thank for your support! Best Regards, Daniele --- (*1*) At beginning i tried using IAX but freeswitch segfaults when it try to answer the call and when i reload the module (trought reload mod_iax) shutdown routing didn't get called(the used iax library, smartly, start using another port without saying anything), however i need to do more testing: hope to open some tickets on jira in short. (*2*) I've seen that mod_iax config file support a context variable, but it isn't used so i wrote a small fix to use it if context isn't specified in the iax request (*3*) I need to use mISDN, rather other things, because i should use octasis soft echo cancellation and them supports only mISDN and Zaptel modules attachment: info.vcf___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Holiday routing examples
Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Holiday routing examples
Thanks for this goodness. I am sure to use it so it is appreciated. On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson and...@hijacked.us wrote: Tony committed my patch for doing 'week of month' conditions in the XML dialplan along with some holiday routing examples to the default dialplan. Now you can detect all the major US holidays in pure dialplan XML without having to do any nasty math or anything (I did it all for you). I've also added a page to the wiki describing how to use it for other dates (like non-US holidays): http://wiki.freeswitch.org/wiki/Holiday_Routing Hope this helps some people. Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Passing incoming remote-party-id from called to caller
Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
Hello Anthony, I think I did not explain myself correctly: The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller. Thanks! __Yehavi: 2009/12/1 Anthony Minessale anthony.miness...@gmail.com Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 1-Dec-09, at 1:42 AM, Yehavi Bourvine wrote: Hello Anthony, I think I did not explain myself correctly: The destination sends the Remote-Party-ID in the Ringing and OK replies, but they are not relayed to the original caller. Thanks! __Yehavi: 2009/12/1 Anthony Minessale anthony.miness...@gmail.com Just set the variables effective_callee_id_name and effective_callee_id_number in your dp before you answer the call On Dec 1, 2009 12:08 AM, Yehavi Bourvine yehavi.bourv...@gmail.com wrote: Hello, I would like Freeswitch to pass the Remote-Party-ID field of the called party (sent in the Ringing OK when answering the call) back to the originator's phone. How can I do that? The drive for this is: Our Freeswitch is connected via a Cisco gateway and PRI to the university's phone exchange. When we call some university's extension the Cisco gateway adds Remote-Party-ID field to the Ringing and OK which includes the called party's name. I would like Freeswitch to relay this to the caller so he/she can see the name of the one who they called. Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Passing incoming remote-party-id from called to caller
Are you on SVN trunk? As far as I recall the callee_id_number/name stuff isnt in 1.0.4. No, because the SVN has problems with Emailing the voicemail... We use 1.0.4 and set sip_callee_id_number/name which works. I would like to not set it and get it from the other side... Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
