[Freeswitch-users] Displaying caller ID on LED?
Hello I was wondering if someone had succesfully configured FS to display caller ID on a LED like this? http://usb.brando.com/prod_detail.php?prod_id=00575 That would be a nice alternative to displaying CID information on the user's PC screen when users need to see who's calling where they're not in front of their computer (doctors, auto mechanics, etc.) Thank you. -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280730.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Displaying caller ID on LED?
... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Flushing the Event buffer in Perl Event Socket
Hi anthony,
I was in a need of flushing the events buffer without reading it.I've done
the following ESL(Async) program to flush the events.
First I register for events.
I answered the call and playback some message. Now the events would have
been queued.
I, then send noevents.
After sending that, I again register for events, and when I receive the
events, I've not got the old events. I got only new events.
But I don't know whether it is exactly a way to flush the events or not. I
just need your suggestions or your thoughts on this.
Here is the script:
use lib /usr/local/freeswitch/scripts/esl;
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip = 192.168..0.0;
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;
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);
print Host name is $host\n;
$con = new ESL::ESLconnection($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);
$con-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav);
$con-send(noevents);
sleep(5);
$con-events(plain, all);
while(my $e = $con-recvEvent())
{
print $e-serialize();
}
}
Anthony Minessale-2 wrote:
read them in a timed loop of some small number of MS until you get a
timeout
meaning you have flushed them all.
On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu
velu.techni...@gmail.comwrote:
Dear All,
I receiving the events in while loop by using recvEventTimed method
in ESL.pm. I have to flush that Event buffer after some particular time.
How
can I do it?
Thanks,
Velusamy
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http://old.nabble.com/Flushing-the-Event-buffer-in-Perl-Event-Socket-tp26125824p26281493.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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[Freeswitch-users] Patch to fix italian pronunce in mod_say_it
Hi,
yesterday i started to fix pronuce in mod_say_it for numbers, dates and
times. I needed to add some sound files because these was necessary for
a correct italian pronunce.
I've patched these three functions:
- play_group
- it_say_time
- it_say_general_count
I've diff it against revision 15396 (i've updated freeswitch tree
yesterday morning)
Can you take a look to the patch?
# Modification to play_group function
In italian we pronunce 123 as cento venti tre and not uno cento venti
tre so, if a is 1 just doesn't play the digit
# Modification to it_say_time
Our long date format is something like
WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR
so i converted the date pronunce to this.
I've dropped am/pm logic, because we have 24h standard, and minutes
related logic because, we don't have it.
# Modification to it_say_general_count
I rewrote number to string conversion to make it more readable (using
just two math operations, a module and a division) and to drop the 999
milions limit (1*)(however more code should be changed to fully drop
this limit).
Changes are mainly related to millions and thousands pronunce: in
italian, if you need to say 1 milion you doesn't say uno milione but
un milione but to say 3 millions you say tre milioni, while for
thousand you doesn't pronunce un at all.
1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it get
up to 12 digits but in xx_say_general_count manage up to 9 digits, so
the first three digits wouldn't get never pronunced
Thank you
Index: src/mod/say/mod_say_it/mod_say_it.c
===
--- src/mod/say/mod_say_it/mod_say_it.c (revisione 15396)
+++ src/mod/say/mod_say_it/mod_say_it.c (copia locale)
@@ -95,7 +95,9 @@
{
if (a) {
- say_file(digits/%d.wav, a);
+if (a != 1) {
+say_file(digits/%d.wav, a);
+}
say_file(digits/hundred.wav);
}
@@ -170,7 +172,7 @@
char *tosay, switch_say_type_t type, switch_say_method_t method,
switch_input_args_t *args)
{
int in;
- int x = 0;
+ int places_count = 0;
int places[9] = { 0 };
char sbuf[13] = ;
switch_status_t status;
@@ -179,26 +181,64 @@
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Parse
Error!\n);
return SWITCH_STATUS_GENERR;
}
-
+
+// Get in
in = atoi(tosay);
+
+// Check if number too big
+if (in 9) {
+// Fail
+return SWITCH_STATUS_FALSE;
+}
+// Check if number isin't zero
if (in != 0) {
- for (x = 8; x = 0; x--) {
- int num = (int) pow(10, x);
- if ((places[(uint32_t) x] = in / num)) {
- in -= places[(uint32_t) x] * num;
- }
- }
-
+
+// Init x to 0
+places_count = 0;
+
+// Loop until in is greater than zero
+do {
+// Get last digit
+places[places_count] = in % 10;
+
+// Drop last digit
+in = in / 10;
+}
+while(in 0 ++places_count 0 /** fake check to put in while */);
+
switch (method) {
case SSM_COUNTED:
case SSM_PRONOUNCED:
- if ((status = play_group(SSM_PRONOUNCED, places[8],
places[7], places[6], digits/million.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
- return status;
- }
- if ((status = play_group(SSM_PRONOUNCED, places[5],
places[4], places[3], digits/thousand.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
- return status;
- }
+
+// Check for milions
+if (places_count 5) {
+// Check if the millions digit is one (digit 6 = 1, digit 7
and 8 = 0)
+if (places[6] == 1 places[7] == 0 places[8] == 0) {
+say_file(digits/un.wav);
+say_file(digits/million.wav);
+} else {
+// Play millions group (digits/million.wav should be
digits/millions.wav)
+if ((status = play_group(SSM_PRONOUNCED, places[8],
places[7], places[6], digits/million.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
+return status;
+}
+}
+
+}
+
+// Check for thousands
+if (places_count 2) {
+if (places[3] == 1 places[4] == 0 places[5] == 0) {
+say_file(digits/thousand.wav);
+} else {
+// Play thousand
Re: [Freeswitch-users] Flushing the Event buffer in Perl Event Socket
$| = 1; I think that is what you're lookin for. /b On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: I was in a need of flushing the events buffer without reading it.I've done the following ESL(Async) program to flush the events. ___ 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] cd-sounds vs. sounds?
Are non-English sound files available in the SVN version of the code? I just tried installing the French sound files, but got an error: Unknown target cd-sounds-fr-install Unknown target cd-moh-fr-install make[1]: *** [cd-sounds-fr-install] Error 1 make: *** [cd-sounds-fr-install] Error 2 make[1]: *** [cd-moh-fr-install] Error 1 make: *** [cd-moh-fr-install] Error 2 [1]+ Exit 2 make cd-sounds-fr-install -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html Sent from the Freeswitch-users mailing list archive at Nabble.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] Patch to fix italian pronunce in mod_say_it
Can you post your patch to jira.freeswitch.org please.
/b
On Nov 10, 2009, at 7:00 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
yesterday i started to fix pronuce in mod_say_it for numbers, dates
and times. I needed to add some sound files because these was
necessary for a correct italian pronunce.
I've patched these three functions:
- play_group
- it_say_time
- it_say_general_count
I've diff it against revision 15396 (i've updated freeswitch tree
yesterday morning)
Can you take a look to the patch?
# Modification to play_group function
In italian we pronunce 123 as cento venti tre and not uno cento
venti tre so, if a is 1 just doesn't play the digit
# Modification to it_say_time
Our long date format is something like
WDAY_NAME, WDAY_NUMBER MONTH_NAME YEAR
so i converted the date pronunce to this.
I've dropped am/pm logic, because we have 24h standard, and minutes
related logic because, we don't have it.
# Modification to it_say_general_count
I rewrote number to string conversion to make it more readable
(using just two math operations, a module and a division) and to
drop the 999 milions limit (1*)(however more code should be changed
to fully drop this limit).
Changes are mainly related to millions and thousands pronunce: in
italian, if you need to say 1 milion you doesn't say uno milione
but un milione but to say 3 millions you say tre milioni, while
for thousand you doesn't pronunce un at all.
1*: i've noticed a little bug in xx_say_money in mod_say_xx ... it
get up to 12 digits but in xx_say_general_count manage up to 9
digits, so the first three digits wouldn't get never pronunced
Thank you
Index: src/mod/say/mod_say_it/mod_say_it.c
===
--- src/mod/say/mod_say_it/mod_say_it.c (revisione 15396)
+++ src/mod/say/mod_say_it/mod_say_it.c (copia locale)
@@ -95,7 +95,9 @@
{
if (a) {
- say_file(digits/%d.wav, a);
+if (a != 1) {
+say_file(digits/%d.wav, a);
+}
say_file(digits/hundred.wav);
}
@@ -170,7 +172,7 @@
char *tosay, switch_say_type_t type, switch_say_method_t
method, switch_input_args_t *args)
{
int in;
- int x = 0;
+ int places_count = 0;
int places[9] = { 0 };
char sbuf[13] = ;
switch_status_t status;
@@ -179,26 +181,64 @@
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, Parse
Error!\n);
return SWITCH_STATUS_GENERR;
}
-
+
+// Get in
in = atoi(tosay);
+
+// Check if number too big
+if (in 9) {
+// Fail
+return SWITCH_STATUS_FALSE;
+}
+// Check if number isin't zero
if (in != 0) {
- for (x = 8; x = 0; x--) {
- int num = (int) pow(10, x);
- if ((places[(uint32_t) x] = in / num)) {
- in -= places[(uint32_t) x] * num;
- }
- }
-
+
+// Init x to 0
+places_count = 0;
+
+// Loop until in is greater than zero
+do {
+// Get last digit
+places[places_count] = in % 10;
+
+// Drop last digit
+in = in / 10;
+}
+while(in 0 ++places_count 0 /** fake check to put in
while */);
+
switch (method) {
case SSM_COUNTED:
case SSM_PRONOUNCED:
- if ((status = play_group(SSM_PRONOUNCED, places[8],
places[7],
places[6], digits/million.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
- return status;
- }
- if ((status = play_group(SSM_PRONOUNCED, places[5],
places[4],
places[3], digits/thousand.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
- return status;
- }
+
+// Check for milions
+if (places_count 5) {
+// Check if the millions digit is one (digit 6 = 1,
digit 7 and 8 = 0)
+if (places[6] == 1 places[7] == 0 places[8]
== 0) {
+say_file(digits/un.wav);
+say_file(digits/million.wav);
+} else {
+// Play millions group (digits/million.wav
should be digits/millions.wav)
+if ((status = play_group(SSM_PRONOUNCED, places
[8], places[7], places[6], digits/million.wav, session, args)) !=
SWITCH_STATUS_SUCCESS) {
+return status;
+}
+}
+
+}
+
+// Check for thousands
+if (places_count 2) {
+if
Re: [Freeswitch-users] Patch to fix italian pronunce in mod_say_it
Hi, patch posted to http://jira.freeswitch.org/browse/MODAPP-362 Best Regards, Daniele Brian West ha scritto: Can you post your patch to jira.freeswitch.org please. /b 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
Re: [Freeswitch-users] Cordless VOIP Phones
2009/11/9 João Mesquita jmesqu...@freeswitch.org: Beat me with a dead cat all you want but I rather the snom m3 than the Siemens A580IP Siemens has very low volume which makes its call quality suck despite of being ergonomic and all... Did you flip hte option in the base station that tells it to make the audio louder? That gigaset application sucks and the base station is slow as hell... Maybe I have a bad unit? I didn't play with any of the gigaset specific stuff, I've disabled any screen savers. Maybe if I had the more fancy handsets the apps would be more useful, but when using the base handset they are not. Oh, and the weather is in C rather than F -- good for the rest of the world but not for us in the US. The base station is very slow with firefox but when I use chrome isn't so bad. Dunno if it was a combination of extensions or what. Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new. The snom m3 has its downsides, but all and all, I am happy with the phone if you consider its price tag here in South America where a Polycom can easily cost over 200USD the cheapest unit. Regards, JM On Mon, Nov 9, 2009 at 9:01 PM, Anthony Minessale anthony.miness...@gmail.com wrote: asstra has one issue where if you look at them wrong they start telling the server that the media ip is 0.0.0.0 which we have never identified but they indeed seem to work better than snom m3 On Mon, Nov 9, 2009 at 4:46 PM, Joseph L. Casale jcas...@activenetwerx.com wrote: The Snom M3 is one of the ones that I was looking at - I would be interested in the Pro's Cons ? Worst POS I have ever used, from a sound quality to ergonomics pov, tech support was as bad... I have Aastra 480i CT's which work well. jlc ___ 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 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 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 -- -Rupa ___ 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] Patch to fix italian pronunce in mod_say_it
Patch applied. // comments aren't allowed in .c files in our tree we try hard to weed them out... anyway its committed now. And thanks for your contribution. /b On Nov 10, 2009, at 8:38 AM, Daniele Salvatore Albano wrote: Hi, patch posted to http://jira.freeswitch.org/browse/MODAPP-362 Best Regards, Daniele Brian West ha scritto: Can you post your patch to jira.freeswitch.org please. /b 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 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] Cordless VOIP Phones
On Tue, Nov 10, 2009 at 6:56 AM, Rupa Schomaker r...@rupa.com wrote: Oh, and it looks like a new firmware came out for the Siemens today. Wonder what it fixes (and breaks). Hmm.. wonder where I can find a list of whats new. Well, it seems to totally break g722. I haven't had a chance to narrow it down further, but beware if g722 is important don't update... -- -Rupa ___ 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] How to pick up someone's phone remotely.
Please see the global-intercept example in the default config. /b On Nov 10, 2009, at 9:01 AM, Piotr Żurek wrote: Hello. Thank You developers for Freeswitch. I have installed it lately and it's working quite nicely, but I have one problem: I need to mimic behavior of my current analogue PBX installation using Freeswitch. This is the scenario: In the office with a few desks (extensions 1000-1010) and only one person behind one of desks (whatever extension - in example 1000). 1. There's incoming call on _one_ of extensions 1001-1010 2. The person on extension 1000 wants to answer this call on his phone so dials #37 and this call is redirected to his phone. That's how it works on my office on analogue PBX system. Anyone can answer a call from any other phone as long as it hasn't been answered already. I tried to use the intercept action (with global example in default config) but it's not what I need because it intercepts the call even if it's already answered. I need to intercept all but only unanswered calls. I tried to use Redirect but it does not work on other's extensions call's (or does it?). Please help. Peter Żurek piotr_zurek.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 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] playback from hadoop
Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? thx, mark On Tue, Nov 10, 2009 at 3:29 AM, Andrew Thompson and...@hijacked.us wrote: On Mon, Nov 09, 2009 at 08:59:54PM +0800, mark morreny wrote: Hi, Does anyone know how to playback based on files from hadoop storage. There is a libhdcp, and java api. Is there anyway to put together a sample middle piece to move files from hadoop to freeswitch using memory space, so there is no disk I/O? Any feedback or suggestion will be greatly appreciated. mod_shell_stream might work, if you can just spit out the raw audio to the shell. Or write another stream module that works with libhdcp. 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
Re: [Freeswitch-users] playback from hadoop
On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? mod_shell_stream is undocumented, but from reading the code I gather it works like this: Module calls fork() and in the child process it runs an arbitrary shell command (specified in its config file?). The parent process then reads raw audio data from the child process and uses it as an audio source. So basicially you could write the shell command in anything, so long as it outputs raw audio to FS. Or maybe I read the code wrong when I skimmed over it. If you do get it working, please contribute some documentation to the wiki. 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] How to pick up someone's phone remotely.
Add the following:
action application=set data=execute_on_answer=db
delete/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/.
after
action application=db
data=insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/
in local extensions default example, or change it globally previously than
this extension. You can join us on IRC if you can any more questions
(sekil).
Regards,
Ognjen
On Tue, Nov 10, 2009 at 4:01 PM, Piotr Żurek piotr_zu...@biprotech.comwrote:
Hello.
Thank You developers for Freeswitch.
I have installed it lately and it's working quite nicely, but I have one
problem:
I need to mimic behavior of my current analogue PBX installation using
Freeswitch.
This is the scenario:
In the office with a few desks (extensions 1000-1010) and only one person
behind one of desks (whatever extension - in example 1000).
1. There's incoming call on _one_ of extensions 1001-1010
2. The person on extension 1000 wants to answer this call on his phone so
dials #37 and this call is redirected to his phone.
That's how it works on my office on analogue PBX system. Anyone can answer
a call from any other phone as long as it hasn't been answered already.
I tried to use the intercept action (with global example in default config)
but it's not what I need because it intercepts the call even if it's already
answered. I need to intercept all but only unanswered calls. I tried to use
Redirect but it does not work on other's extensions call's (or does it?).
Please help.
Peter Żurek
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[Freeswitch-users] SPA3102 Won't drop the PSTN line (UK)
I'm close to getting my SPA3102 working - he says hopefully .. . . . Making and receiving calls seems to be OK, but the SAP3102 doesn't seem to want to let go of the phone line once it's got it. Example I can receive a call, nobody answers and it goes to voicemail - working so far. FreeSwitch processes the normal VoiceMail system playing the prompts and recording the call. At the end, it says Goodbye and hear a click (from the remote end) and I see the console message that 2009-11-10 15:52:52.625000 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/1...@192.168.1.181 [CS_EXECUTE] [NORMAL_CLEARING] 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1086 Session 362 (sofia/internal/1...@192.168.1.181) Ended 2009-11-10 15:52:52.625000 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1...@192.168.1.181 [CS_DESTROY] The above messages would suggest to me that FreeSwitch is doing its stuff right, but I have posted a dump in the pastebin just in case. The SPA3102 does not want to relinquish the line until the remote caller hangs up. Has anyone had similar problems with the SPA3102 or has any ideas where I can look to get to the bottom of the problem. (I have just upgraded the SPA3102 to the latest 5.1.0 firmware) regards Dave___ 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] Dialplans and XML_CURL
Good morning everyone. I have a question regarding using MOD XML_CURL and returning a dial plan. I have my system setup to respond with the following dialplan. ?xml version=1.0? document type=freeswitch/xml section name=dialplan description=Regex/XML Dialplan context name=default extension name=one condition field=destination_number expression=^(.*)$ !-- Do Condition stuff here ... -- /condition /extension extension name=two condition field=destination_number expression=^(.*)$ !-- Do Condition stuff here ... -- /condition /extension extension name=three condition field=destination_number expression=^(.*)$ !-- Do Condition stuff here ... -- /condition /extension /context /section /document My question is this. Can extension one, use extension two and three without XML_CURL making another dialplan request? Thanks, Jonathan Pitcher ___ 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] cd-sounds vs. sounds?
I believe that French and Spanish sounds are in the works by the community. The only other sounds I'm aware of are the Russian ones. -MC On Tue, Nov 10, 2009 at 6:13 AM, Fred-145 codecompl...@free.fr wrote: Are non-English sound files available in the SVN version of the code? I just tried installing the French sound files, but got an error: Unknown target cd-sounds-fr-install Unknown target cd-moh-fr-install make[1]: *** [cd-sounds-fr-install] Error 1 make: *** [cd-sounds-fr-install] Error 2 make[1]: *** [cd-moh-fr-install] Error 1 make: *** [cd-moh-fr-install] Error 2 [1]+ Exit 2 make cd-sounds-fr-install -- View this message in context: http://old.nabble.com/cd-sounds-vs.-sounds--tp26269842p26284109.html Sent from the Freeswitch-users mailing list archive at Nabble.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 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] Dialplans and XML_CURL
On Tue, Nov 10, 2009 at 8:55 AM, Mathieu Rene mrene_li...@avgs.ca wrote: You'll get a single xml curl request, unless you use the transfer application, which will trigger another one. Just curious: what about execute_extension? Does that cause a new XML CURL request also? I didn't see anything on the wiki about that. I'll update the wiki mod_xml_curl page accordingly. -MC ___ 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] Patch to fix italian pronunce in mod_say_it
Hi, thank you and for your work! Where i can find coding style rules? Best Regards, Daniele Brian West ha scritto: Patch applied. // comments aren't allowed in .c files in our tree we try hard to weed them out... anyway its committed now. And thanks for your contribution. /b 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
Re: [Freeswitch-users] Cordless VOIP Phones
Hey Hadley, jump up on irc sometimes. Regards, Ognjen On Mon, Nov 9, 2009 at 9:26 PM, Hadley Rich h...@nice.net.nz wrote: On Mon, 2009-11-09 at 14:05 -0600, Brian West wrote: Get an ATA with a Dect handset it works much better... the Snom M3 and the Aastra are one in the same and they both do not live up to the quality or usability requirements. That said, they are better than what else is around. I'd call them average. Nothing to write home about but you don't need to run away from them. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier ___ 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] SIP trunk without authentication
Hello. I'm FS newbie and want connect it to SIP provider which does not require authentication - it make authentication using my IP. I've searched through FS documentation and didn't find clear answer. Could you help me or maybe give a link to a doc which can help? Thanks. -- Sergey ___ 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] SIP trunk without authentication
As easy as:
action application=bridge data=sofia/external/$
{destination_numb...@ip_address_here /
in your dialplan. If you want to make a gateway out of it, you can
enter whatever you want in username and password since they won't be
used. (SIP works using challenge authentication which means the remote
UA has to send you a packet requesting the credentials).
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 10-Nov-09, at 1:27 PM, Sergey Kobzar wrote:
Hello.
I'm FS newbie and want connect it to SIP provider which does not
require authentication - it make authentication using my IP.
I've searched through FS documentation and didn't find clear answer.
Could you help me or maybe give a link to a doc which can help?
Thanks.
--
Sergey
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Re: [Freeswitch-users] How to pick up someone's phone remotely.
just change the dialplan/default.xml as mentioned by brian but i think you
can't use # as the first key 'cuz it normally used as a Send key. you may
change # to * (star key).
On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija osesl...@gmail.com wrote:
Add the following:
action application=set data=execute_on_answer=db
delete/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/.
after
action application=db
data=insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}/
in local extensions default example, or change it globally previously than
this extension. You can join us on IRC if you can any more questions
(sekil).
Regards,
Ognjen
On Tue, Nov 10, 2009 at 4:01 PM, Piotr Żurek piotr_zu...@biprotech.comwrote:
Hello.
Thank You developers for Freeswitch.
I have installed it lately and it's working quite nicely, but I have one
problem:
I need to mimic behavior of my current analogue PBX installation using
Freeswitch.
This is the scenario:
In the office with a few desks (extensions 1000-1010) and only one person
behind one of desks (whatever extension - in example 1000).
1. There's incoming call on _one_ of extensions 1001-1010
2. The person on extension 1000 wants to answer this call on his phone so
dials #37 and this call is redirected to his phone.
That's how it works on my office on analogue PBX system. Anyone can answer
a call from any other phone as long as it hasn't been answered already.
I tried to use the intercept action (with global example in default
config) but it's not what I need because it intercepts the call even if it's
already answered. I need to intercept all but only unanswered calls. I tried
to use Redirect but it does not work on other's extensions call's (or does
it?).
Please help.
Peter Żurek
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Re: [Freeswitch-users] How to pick up someone's phone remotely.
That depends on the phone... some let you do it.. some don't... WELCOME TO VOIP!!! /b On Nov 10, 2009, at 3:48 PM, Nandy Dagondon wrote: just change the dialplan/default.xml as mentioned by brian but i think you can't use # as the first key 'cuz it normally used as a Send key. you may change # to * (star key). ___ 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] Simple Conference Setup issue
I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: extension name=test !-- your provider or any name you'd like to call it -- condition field=destination_number expression=xx !-- your DID for this gateway-- action application=conference data=conference.conf+12345/ /condition /extension Help appreciated ___ 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] Simple Conference Setup issue
What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: extension name=test !-- your provider or any name you'd like to call it -- condition field=destination_number expression=xx !-- your DID for this gateway-- action application=conference data=conference.conf +12345/ /condition /extension ___ 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] Help with dynamic IVR
Hello. I am very new to FreeSwitch, Telephony and IVR. My goal is to prepare a student assessment IVR system as a college project. But this IVR is going to be dynamic. So for each student assessment may be different (number of questions, possible responses, flow of prompts, etc). Is it possible to achieve something like this with FreeSwitch? Most IVR we see are static (like a bank IVR system that flows always in same way). That is why I am confused. Please share your views. Malay Thakershi ___ 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] Simple Conference Setup issue
My mistake , it picked the default profile and was waiting for moderator in the conference.cof.xml file that is provided with the install. From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, November 10, 2009 3:50 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Simple Conference Setup issue What does your config look like? /b On Nov 10, 2009, at 4:39 PM, Ujjval Karihaloo wrote: I am trying to call into a DID that is pointed to a Conf Bridge on Freeswitch and when I have 2 people dial in, looks like the Music on Hold never stops. Here is what my public.xml looks like: extension name=test !-- your provider or any name you'd like to call it -- condition field=destination_number expression=xx !-- your DID for this gateway-- action application=conference data=conference.conf+12345/ /condition /extension ___ 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] Help with dynamic IVR
As I opinion, it's not necessary write ivr script for each student. A static ivr script load question and response dynamic is what you need. 2009/11/11 Malay Thakershi malay.thaker...@continuityhealth.com Hello. I am very new to FreeSwitch, Telephony and IVR. My goal is to prepare a student assessment IVR system as a college project. But this IVR is going to be dynamic. So for each student assessment may be different (number of questions, possible responses, flow of prompts, etc). Is it possible to achieve something like this with FreeSwitch? Most IVR we see are static (like a bank IVR system that flows always in same way). That is why I am confused. Please share your views. Malay Thakershi ___ 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 -- Lei.Tang lei.tl...@gmail.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] playback from hadoop
Hi Sorry to ask again. I know the command to copy file from hadoop file system to somewhere else. But how do I make a shell command to output raw audio? What command is it like? Is it like play()? I am confused. Thx, mark On Tue, Nov 10, 2009 at 11:56 PM, Andrew Thompson and...@hijacked.uswrote: On Tue, Nov 10, 2009 at 11:29:37PM +0800, mark morreny wrote: Hi Thanks for the tips. May I ask how to split the file from hadoop to the shell? Is it like copying the file to certain dir? I can't find any mod_shell_stream related info from the wiki. Does anyone know how to use it? mod_shell_stream is undocumented, but from reading the code I gather it works like this: Module calls fork() and in the child process it runs an arbitrary shell command (specified in its config file?). The parent process then reads raw audio data from the child process and uses it as an audio source. So basicially you could write the shell command in anything, so long as it outputs raw audio to FS. Or maybe I read the code wrong when I skimmed over it. If you do get it working, please contribute some documentation to the wiki. 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
Re: [Freeswitch-users] Help with dynamic IVR
What you will probably want, if you are looking to go 'thicker' with this would be one of the IVR scripting languages and a database connection. For instance lua, and the database connection(either mysql or postgresql or sqlite). ' From there you have users, questions, and answers mapped in the database. Feel free to e-mail me about this off list for more assistance. -William King Lei Tang wrote: As I opinion, it's not necessary write ivr script for each student. A static ivr script load question and response dynamic is what you need. 2009/11/11 Malay Thakershi malay.thaker...@continuityhealth.com mailto:malay.thaker...@continuityhealth.com Hello. I am very new to FreeSwitch, Telephony and IVR. My goal is to prepare a student assessment IVR system as a college project. But this IVR is going to be dynamic. So for each student assessment may be different (number of questions, possible responses, flow of prompts, etc). Is it possible to achieve something like this with FreeSwitch? Most IVR we see are static (like a bank IVR system that flows always in same way). That is why I am confused. Please share your views. Malay Thakershi ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto: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 -- Lei.Tang lei.tl...@gmail.com mailto:lei.tl...@gmail.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 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] Displaying caller ID on LED?
I did something like this recently.From the dial plan it is easy to execute an external application on an incoming call with the caller's info. At that point if you can just push it down to the LCD panel all the better, but if your FS server is remote, and has no direct access to the client to render the caller ID, you will have to setup a fake push to get instant responses. You can do this through apache, or a simple tcp server but the idea being the client connects up to the server, and the server blocks until an incoming call comes in, it then responds to the client, and you have the caller id fairly instantly showing up. You could also use the event socket, heck even maybe use the event socket remotely if you wanted to, and then avoid some of the server side complexity too. ~Mitch On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 codecompl...@free.fr wrote: ... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.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 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] Displaying caller ID on LED?
If you donate one to the FsGui project, I can make it happen for you. Contact me off list if you are interested. Regards, JM On Wed, Nov 11, 2009 at 1:01 AM, Mitch Capper mitch.cap...@gmail.comwrote: I did something like this recently.From the dial plan it is easy to execute an external application on an incoming call with the caller's info. At that point if you can just push it down to the LCD panel all the better, but if your FS server is remote, and has no direct access to the client to render the caller ID, you will have to setup a fake push to get instant responses. You can do this through apache, or a simple tcp server but the idea being the client connects up to the server, and the server blocks until an incoming call comes in, it then responds to the client, and you have the caller id fairly instantly showing up. You could also use the event socket, heck even maybe use the event socket remotely if you wanted to, and then avoid some of the server side complexity too. ~Mitch On Tue, Nov 10, 2009 at 5:06 AM, Fred-145 codecompl...@free.fr wrote: ... or alternatively, on one of those USB digital picture frames? www.amazon.com/Digital-Spectrum-USB-Photo-Frame/dp/B87BHC -- View this message in context: http://old.nabble.com/Displaying-caller-ID-on-LED--tp26280730p26280912.html Sent from the Freeswitch-users mailing list archive at Nabble.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 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] Freeswitch core dumped, when setting callback to events
Here is the required detail.
http://pastebin.freeswitch.org/11049
On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
1) install gdb
2) run support_d/fscore_db in the tree from the working directory of the
core.
3) if you are not on svn trunk, make current and start over.
On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
Dear all,
I did the below code, to callback a function when CHANNEL_EXECUTE_COMPLETE
event comes.
I executed the script for the 1st time and I got nothing.
When I executed the script for the 2nd time, it ended with Sedmentation
fault with core dumped.
I was unable to attach the core dump file with this mail.
Please specify how to send files to freeswitch user mailing list if need
be.
The freeswitch log is here:
http://pastebin.freeswitch.org/11038
#!/usr/bin/perl
use strict;
use Data::Dumper;
our $session;
$session-answer();
my $events=new freeswitch::EventConsumer(CHANNEL_EXECUTE_COMPLETE);
$events-pop(1);
$events-swig_e_callback_set(playvoice);
sub playvoice()
{
freeswitch::consoleLog(INFO,Call back function called\n);
}
return 1;
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Re: [Freeswitch-users] Freeswitch core dumped, when setting callback to events
You need to install the debug packages so you the symbols because that backtrace is useless. /b On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: Here is the required detail. http://pastebin.freeswitch.org/11049 ___ 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] Freeswitch core dumped, when setting callback to events
It doesn't look like its trying to look for symbols inside freeswitch
gdb /path/to/freeswitch/here /path/to/core/here
bt
thread apply all bt
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 10-Nov-09, at 8:10 PM, lakshmanan ganapathy wrote:
Here is the required detail.
http://pastebin.freeswitch.org/11049
On Mon, Nov 9, 2009 at 10:04 PM, Anthony Minessale anthony.miness...@gmail.com
wrote:
1) install gdb
2) run support_d/fscore_db in the tree from the working directory of
the core.
3) if you are not on svn trunk, make current and start over.
On Mon, Nov 9, 2009 at 5:53 AM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
Dear all,
I did the below code, to callback a function when
CHANNEL_EXECUTE_COMPLETE event comes.
I executed the script for the 1st time and I got nothing.
When I executed the script for the 2nd time, it ended with
Sedmentation fault with core dumped.
I was unable to attach the core dump file with this mail.
Please specify how to send files to freeswitch user mailing list if
need be.
The freeswitch log is here:
http://pastebin.freeswitch.org/11038
#!/usr/bin/perl
use strict;
use Data::Dumper;
our $session;
$session-answer();
my $events=new freeswitch::EventConsumer(CHANNEL_EXECUTE_COMPLETE);
$events-pop(1);
$events-swig_e_callback_set(playvoice);
sub playvoice()
{
freeswitch::consoleLog(INFO,Call back function called\n);
}
return 1;
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
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sip:8...@conference.freeswitch.org
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pstn:213-799-1400
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Re: [Freeswitch-users] Freeswitch core dumped, when setting callback to events
What is meant by debug packages. Kindly specify where it is available. On Wed, Nov 11, 2009 at 10:09 AM, Brian West br...@freeswitch.org wrote: You need to install the debug packages so you the symbols because that backtrace is useless. /b On Nov 10, 2009, at 10:10 PM, lakshmanan ganapathy wrote: Here is the required detail. http://pastebin.freeswitch.org/11049 ___ 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] Flushing the Event buffer in Perl Event Socket
That doesn't seems to work for me. Here is my need. I'm using Async in the Event socket outbound. I'll register for events plain all I'll answer the call. I'll playback a message. I'll sleep for 5 seconds. After that, I'll receive the events. I don't need the events that are for answer and playback. That action is completed and don't want to receive events for those application. I set $|=1 in my ESL script. But it doesn't seems to solve the above issue. Any helppls!!! On Tue, Nov 10, 2009 at 7:41 PM, Brian West br...@freeswitch.org wrote: $| = 1; I think that is what you're lookin for. /b On Nov 10, 2009, at 4:51 AM, lakshmanan wrote: I was in a need of flushing the events buffer without reading it.I've done the following ESL(Async) program to flush the events. ___ 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] playback from hadoop
On Wed, Nov 11, 2009 at 11:02:10AM +0800, mark morreny wrote: Hi Sorry to ask again. I know the command to copy file from hadoop file system to somewhere else. But how do I make a shell command to output raw audio? What command is it like? Is it like play()? I am confused. I was very nice and wrote up some documentation (and 2 examples) on the wiki page at http://wiki.freeswitch.org/wiki/Mod_shell_stream Now you know everything I know about using this module (which is a very cool module, by the way - thanks Tony). 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
