Re: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
Hello Michael, how much $$ are we talking about? I need this issue to be solved quickly and it's worth to spend some money. I've read the following post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html and have the same symptom with after hundreds of calls I start to get b channels that are stuck in states like TERMINATING or HANGUP Best regards Peter Michael Collins schrieb: I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -MC On Wed, Jan 14, 2009 at 9:44 AM, Peter P GMX prometheus...@gmx.net wrote: After a time I receive the following error when a call comes in on our OpenZap span 2: parse error [-3012] [Q931E_INVALID_CRV] Here's the log 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 23 (0x17, CTX: Originator) 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:352 zap_isdn_931_34() zchan 0 (-1:-1) source isdn_data-channels_remote_crv[0x17] 2009-01-14 13:14:11 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] 2009-01-14 13:14:15 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_23() READ 5 When freeswitch is restarted or mod_openzap is reloaded, the error is gone away. Any idea what this can be? Best regards Peter ___ 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 crashed !!!
There NO previous version of FS installed before, and FS 1.0.2 is also freshly installed. Anthony Minessale-2 wrote: please remove FS src and dest dir from your machine and recompile fresh from scratch. On Tue, Jan 13, 2009 at 4:48 AM, shehzad p pmh...@gmail.com wrote: Please find the output of bt from below pastebin link: http://pastebin.freeswitch.org/6757 Thanks, pms Michael S Collins wrote: Could you please do a backtrace and post it to a pastebin? If in Linux do this: gdb /path/to/freeswitch /path/to/corefile -MC Sent from my iPhone On Jan 12, 2009, at 5:23 AM, shehzad p pmh...@gmail.com wrote: Hi all, I am also testing FS release 1.0.2, but I faced strange problem. When I stop freeswitch (from CLI using ... or shutdown), Freeswitch ends with showing Segmentation fault: Below is the last 15 lines when fault occures. Sometimes this does not happen and FS shut down normally. === === === === === == 2009-01-12 16:52:56 [CONSOLE] switch_loadable_module.c:1244 do_shutdown() mod_esf unloaded. 2009-01-12 16:52:56 [CONSOLE] switch_core.c:1462 switch_core_destroy() Closing Event Engine. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 switch_event_shutdown() Stopping event queue 0 2009-01-12 16:52:56 [CONSOLE] switch_event.c:440 switch_event_shutdown() Stopping event queue 1 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 0 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:445 switch_event_shutdown() Stopping dispatch queue 0 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 1 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:247 switch_event_dispatch_thread() Dispatch Thread 0 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:305 switch_event_thread() Event Thread 2 Ended. 2009-01-12 16:52:56 [CONSOLE] switch_event.c:414 switch_core_memory_reclaim_events() Returning 23 recycled event(s) 1012 bytes 2009-01-12 16:52:56 [CONSOLE] switch_event.c:416 switch_core_memory_reclaim_events() Returning 331 recycled event header(s) 5296 bytes 2009-01-12 16:52:56 [CONSOLE] switch_core_sqldb.c:539 switch_core_sqldb_stop() Waiting for unfinished SQL transactions 2009-01-12 16:52:56 [NOTICE] switch_core_sqldb.c:199 switch_core_sql_thread() SQL thread ending 2009-01-12 16:52:56 [CONSOLE] switch_scheduler.c:303 switch_scheduler_task_thread_stop() Stopping Task Thread Segmentation fault (core dumped) === === === === === == What should be the cause of such crash. ahgindia wrote: Hi All, Recently I was testing the new freeswitch release 1.0.2 The system has Fedora 8 with 2 GB RAM with Intel(R) Core(TM)2 Duo CPU E7200 @ 2.53GHz processor. But it crashed, when there were 96 active calls in it (as can be seen from show calls on freeswitch cli) There is a dump file for it, in the folder from where i started the freeswitch. Let me know how can we know the cause of the crash. -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21414332.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 -- View this message in context: http://www.nabble.com/Freeswitch-crashed-%21%21%21-tp21386948p21433120.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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
[Freeswitch-users] OpenZAP hardware timers
Is there a way to use the hardware timers e.g. of a PRI card in fresswitch? Or other question: Is it recommended to use those if they are available? I have installed a dual PRI card, and show timer shows one soft timer. Best regards Peter ___ 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] Country specific tones - how to contribute?
Thanks, will go and have a look at the developers list.
Scott
Jason White wrote:
Scott Ellis scott.el...@novatex.com.au wrote:
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
Maybe by posting a patch to the bug tracking system or the development list?
I tried action application="set" data=""/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
us-ring and uk-ring are defined in vars.xml. Note that they are global
variables, referenced with the $${variable-name} syntax.
There's an ITU document referred to on the wiki with the official definitions
of ringback and other tones for various countries.
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[Freeswitch-users] FS 1.2 Windows Error
Hi, Im getting error on startup when executing freeSWITCH.exe , The procedure entry point_apr_...@12 could not be located in the dynamic library libaprutil.dll -- /Lito ___ 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] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent?
After poking around in the code, it looks like if I set param
name=enable-callerid value=false/ in openzap.conf.xml, it should
skip the GET_CALLERID state, and I should get the call answered straight
away.
mod_openzap.c
} else if (!strcasecmp(var, enable-callerid)) {
enable_callerid = val;
if (zap_configure_span(analog, span, on_analog_signal,
tonemap, tonegroup,
digit_timeout, to,
max_dialstr, max,
hotline, hotline,
enable_callerid, enable_callerid,
TAG_END) != ZAP_SUCCESS) {
zap_log(ZAP_LOG_ERROR, Error starting OpenZAP span
%d\n, span_id);
continue;
}
ozmod_analog.c
else if (!strcasecmp(var, enable_callerid)) {
if (!(val = va_arg(ap, char *))) {
break;
}
if (zap_true(val)) {
flags |= ZAP_ANALOG_CALLERID;
} else {
flags = ~ZAP_ANALOG_CALLERID;
}
and
case ZAP_OOB_RING_START:
{
if (event-channel-type != ZAP_CHAN_TYPE_FXO) {
zap_log(ZAP_LOG_ERROR, Cannot get a RING_START event on
a non-fxo channel, please check your config.\n);
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_DOWN);
goto end;
}
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN !zap_test_flag(event-channel,
ZAP_CHANNEL_INTHREAD))) {
if (zap_test_flag(analog_data, ZAP_ANALOG_CALLERID)) {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_GET_CALLERID);
} else {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_IDLE);
}
event-channel-ring_count = 1;
zap_mutex_unlock(event-channel-mutex);
locked = 0;
zap_thread_create_detached(zap_analog_channel_run,
event-channel);
} else {
event-channel-ring_count++;
}
}
break;
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [DOWN]
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:760 process_event() Changing
state on 1:1 from DOWN to GET_CALLERID
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run()
ANALOG CHANNEL thread starting.
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run()
Executing state handler on 1:1 for GET_CALLERID
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:44 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:45 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:47 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:48 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [GET_CALLERID]
2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:292 zap_analog_channel_run()
Changing state on 1:1 from GET_CALLERID to IDLE
2009-01-15 20:19:49 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run()
Executing state handler on 1:1 for IDLE
2009-01-15 20:19:49 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO
sig 1:1 [START]
The code all looks right, but I am not getting what I think should
happen. Anyone with any ideas?
Scott
Scott Ellis wrote:
Searched the wiki and mailing lists as best I can, but with no luck.
How do I get OpenZap to answer a call immediately? (I do not need caller id)
Scott
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Re: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Michael, it must not be the case here, but I had the same error, when incomming calles used a wrong numbering plan resp not the one, FS expected. Just a hint. regards Helmut Am 15.01.2009 09:20, schrieb Peter P GMX: Hello Michael, how much $$ are we talking about? I need this issue to be solved quickly and it's worth to spend some money. I've read the following post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html and have the same symptom with after hundreds of calls I start to get b channels that are stuck in states like TERMINATING or HANGUP Best regards Peter Michael Collins schrieb: I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -MC -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvFEkACgkQ4tZeNddg3dxitgCeIgNS+qUwYQ0ypc1KyXjRO3OV OFwAn1TeaNP466OWErmqEFr9H9p2Wam5 =2NfD -END PGP SIGNATURE- ___ 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] SQLExecute catches not all errors
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, today I moved the voicemail database from sqlite to mysql via odbc. FS started up and connected successfully to the empty database. Normally voicemail adds the neccessary database tables automaticly during startup. In this case I forgot to add permissions for creating, dropping and altering tables to the database user in mysql. So no SM table was createt in database. Unfortunately FS can not detect this. FS thinks everything is ok. After adding the permissions one table was created but the voicemail_prefs wasn't. This was, because I extented the create statement for this table - and the statement was wrong, so mysql couldn't execute it. This case wasn't detected by FS as well and FS resp. voicemail modul thought everything is fine Is there a way to detect those errors, because bug hunting can be last quite long without error messages especially when you have no way to access mysql.log to see the sql statements from FS. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvFuYACgkQ4tZeNddg3dwT5ACfd/ArMsLfsLrnc5peY1qxaDWu kbsAn00gvNJjXwtFYIX41lbbgGWF+m1P =GO2m -END PGP SIGNATURE- ___ 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] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
Helmut, can you give me a hint, how you worked around this? Best regards Peter Helmut Kuper schrieb: Hi Michael, it must not be the case here, but I had the same error, when incomming calles used a wrong numbering plan resp not the one, FS expected. Just a hint. regards Helmut Am 15.01.2009 09:20, schrieb Peter P GMX: Hello Michael, how much $$ are we talking about? I need this issue to be solved quickly and it's worth to spend some money. I've read the following post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html and have the same symptom with after hundreds of calls I start to get b channels that are stuck in states like TERMINATING or HANGUP Best regards Peter Michael Collins schrieb: I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -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 ___ 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] Sending SMS to SIPtoGSM gateway
Hi I have a IP to GSM gateway which supports SIP. How I can send SMS to the GSM phone using FreeSwitch + SIP GSM GW? Thanks Imthiyaz ___ 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] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent - a solution of sorts.
So I decided to hack the code to see if I could just get it to do what I
wanted - assuming some kind of error in the options setting.
First I changed the state change code to just skip straight to IDLE
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN !zap_test_flag(event-channel,
ZAP_CHANNEL_INTHREAD))) {
// if (zap_test_flag(analog_data,
ZAP_ANALOG_CALLERID)) {
// zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_GET_CALLERID);
// } else {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_IDLE);
// }
event-channel-ring_count = 1;
zap_mutex_unlock(event-channel-mutex);
locked = 0;
zap_thread_create_detached(zap_analog_channel_run, event-channel);
} else {
event-channel-ring_count++;
}
So we skip the GET_CALLERID state altogether.
This generated an illegal state change message cannot go from DOWN to IDLE
So then changed the code to
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN !zap_test_flag(event-channel,
ZAP_CHANNEL_INTHREAD))) {
// if (zap_test_flag(analog_data,
ZAP_ANALOG_CALLERID)) {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_GET_CALLERID);
// } else {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_IDLE);
// }
event-channel-ring_count = 1;
zap_mutex_unlock(event-channel-mutex);
locked = 0;
zap_thread_create_detached(zap_analog_channel_run, event-channel);
} else {
event-channel-ring_count++;
}
Allowing the state change to GET_CALLERID, then immediately to IDLE.
This works perfectly - the call is answered straight away. At the moment
I don't know enough about linux debugging to step through the parameter
code to see why setting get caller ID to false in openzap.conf.xml does
not get passed through, but even if it does the current code will still
run into the illegal state change error.
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [DOWN]
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:760 process_event() Changing
state on 1:1 from DOWN to GET_CALLERID
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:762 process_event() Changing
state on 1:1 from GET_CALLERID to IDLE
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run()
ANALOG CHANNEL thread starting.
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run()
Executing state handler on 1:1 for IDLE
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO
sig 1:1 [START]
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU
20ms
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1137 zap_channel_from_event()
Connect inbound channel OpenZAP/1:1/1
2009-01-15 21:59:18 [NOTICE] switch_channel.c:565
switch_channel_set_name() New Channel OpenZAP/1:1/1
[8e2a55c8-e2f3-11dd-adfd-6d934f226ffd]
Will go and put this into JIRA in the next couple of days.
Scott
Scott Ellis wrote:
After poking around in the code, it looks like if I set param
name=enable-callerid value=false/ in openzap.conf.xml, it should
skip the GET_CALLERID state, and I should get the call answered straight
away.
mod_openzap.c
} else if (!strcasecmp(var, enable-callerid)) {
enable_callerid = val;
if (zap_configure_span(analog, span, on_analog_signal,
tonemap, tonegroup,
digit_timeout, to,
max_dialstr, max,
hotline, hotline,
enable_callerid, enable_callerid,
TAG_END) != ZAP_SUCCESS) {
zap_log(ZAP_LOG_ERROR, Error starting OpenZAP span
%d\n, span_id);
continue;
}
ozmod_analog.c
else if (!strcasecmp(var, enable_callerid)) {
if (!(val = va_arg(ap, char *))) {
break;
}
if (zap_true(val)) {
flags |= ZAP_ANALOG_CALLERID;
} else {
flags = ~ZAP_ANALOG_CALLERID;
}
and
case ZAP_OOB_RING_START:
{
if (event-channel-type != ZAP_CHAN_TYPE_FXO) {
zap_log(ZAP_LOG_ERROR, Cannot get a RING_START
[Freeswitch-users] spidermonkey problems
Hello I got problems with hanging spidermonkey sessions and need some advice on how to debug them. I've made a javascript queue application that uses mod_spidermonkey_socket. It works fine for a while, but after some calls I noticed that calls didnt get transferred to agents. The reason was that earlier calls had not been terminated properly. freeswi...@test1 hupall 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 switch_core_session_hupall() Giving up with 8 sessions remaining API CALL [hupall()] output: +OK hangup all channels with cause MANAGER_REQUEST freeswi...@test1 show calls API CALL [show(calls)] output: 0 total. As you can see, 8 sessions are alive, but none of them are listed as calls. What kind of logs should I turn on to see what is happening with those sessions? Thanks, Jonas ___ 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] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Peter, it was simply a change in our TDM Voice Switch. It used a different numbering plan and we changed it to national to get it work with FS and openzap in Q921/Q931 mode. What I still search is a way to configure the numberplan in FS. To make it clear: In my case it didn't work from the second FS starts up. So this differs from your problem. To get an idea what's going on on the TDM link I used a TDM D-Channel monitoring device and traced the d-channel messages exchanged between FS and TDM. That should make it easier to see what's wrong when the problems occur. But you can also increase FS debug level to debug and trace the Q921 and Q931 messages in FS console via fs_cli during runtime. You have to set this in openzap.conf.xml: param name=q921loglevel value=debug/ param name=q931loglevel value=debug/ Unfortunately FS doesn't decode the whole Q931 messages, but it shows a hex representation of the message, so you can manually decode it with this documents: Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en I think for numberingplan issues you only have to track the Q.931 messages. The last idea I have to get some light into your problem and to avoid manually decoding, try to convert FS's q931 hexdump into wiresharks pcap format. Wireshark should be able to decode it :) http://wiki.wireshark.org/Q.931 Maybe it's a good idea to implement a wireshark export for those messages in FS. This will make debugging easy and cheap. Hope it helps a bit. regards helmut Am 15.01.2009 12:06, schrieb Peter P GMX: Helmut, can you give me a hint, how you worked around this? Best regards Peter Helmut Kuper schrieb: Hi Michael, it must not be the case here, but I had the same error, when incomming calles used a wrong numbering plan resp not the one, FS expected. Just a hint. regards Helmut Am 15.01.2009 09:20, schrieb Peter P GMX: Hello Michael, how much $$ are we talking about? I need this issue to be solved quickly and it's worth to spend some money. I've read the following post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html and have the same symptom with after hundreds of calls I start to get b channels that are stuck in states like TERMINATING or HANGUP Best regards Peter Michael Collins schrieb: I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -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 ___ 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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklvLHMACgkQ4tZeNddg3dxF0ACgpMqGf8hu1iSKbOG7nG2o1HZN qdEAoIpTY3Bgwv9wzhV7lq7IKtvDxO5/ =lDVf -END PGP SIGNATURE- ___ 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] No caller/called ID received (Wildcard X101P)
Thank you very much for your help, I've realized I was specting to receive my house phone number having a POTS line and that's not possible. So, I've put my house number in openzap.conf: [span zt] name = OpenZAP number = 91999 fxo-channel = 1 And I've added an extension on the default dialplan: extension name=public_did condition field=destination_number expression=^91999$ action application=answer/ action application=sleep data=2000/ action application=ivr data=demo_ivr/ /condition /extension So I was hopping the IVR answer the call when it is received but instead of that nothing happens, this is the log of one incoming call: 2009-01-15 21:16:56 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999[74cb661e-e341-11dd-acde-9740a65ca868] 2009-01-15 21:16:56 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:16:56 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 13 (OpenZAP/1:1/91999) Ended 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] 2009-01-15 21:17:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999 [78fe36b2-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:03 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:17:03 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 14 (OpenZAP/1:1/91999 ) Ended 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] 2009-01-15 21:17:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999 [7c60cfea-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:17:09 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/91999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 15 (OpenZAP/1:1/91999 ) Ended 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] Someone knows what's happening? Thank you. On Wed, Jan 14, 2009 at 4:42 PM, Jon Radel j...@radel.com wrote: Tomás wrote: Hi, Anthony, I think that's my problem, when I receive a call from the PSTN, FS receive number 1 instead of my house number and I don't know why. If you use SIP trunking or something like an ISDN-PRI line, the number the call is to is delivered as part of the signaling, which is only way to make use of many phone numbers on a single physical circuit or connection. When you put a POTS line into an FXO port, there is no such information provided, as there is only one number on the line. (Leaving aside various schemes found in some countries such as using different ring patterns to indicate different numbers having been called.) So, as Anthony keeps pointing out, if you want FS to know the number of the line plugged into the FXO port, you have to configure it yourself. --Jon Radel ___ 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] Announcing the FreeSWITCH Technology Preview VMWare Appliance.
FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken kr...@freeswitch.org kr...@rmktek.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] Announcing the FreeSWITCH Technology Preview VMWare Appliance.
Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken ___ 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] mod_sofia: NAT-ping RPID bounties
kokoska rokoska wrote: Hi all, I have just post two bounties Where did you post these bounties? We've started moving bounties away from the wiki and adding them to jira instead. ( so that progress can be followed more closely ) -Ray begin:vcard fn:intralanman n:Chandler;Raymond adr:;;630 Cooks Rd.;Farmville;VA;23901;United States email;internet:intralan...@freeswitch.org tel;cell:+1.434.315.4132 x-mozilla-html:TRUE version:2.1 end:vcard ___ 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] Changes in PlayAndGetDigits
I added another arg to the list. I'll have to revisit this today to make sure I did this right for your case. /b On Jan 15, 2009, at 1:43 AM, Juan Backson wrote: Hi, Is there a change in the playAndGetDigits api? In the old release, 11102, my lua script is working but is not working in the latest release. The error I am getting is Error in playAndGetDigits expected 10..10 args, got 9 . Thanks, JB ___ 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] spidermonkey problems
http://wiki.freeswitch.org/wiki/Report_Issue_Checklist Please open a jira and include your script and a test case. /b On Jan 15, 2009, at 5:20 AM, Jonas Gauffin wrote: Hello I got problems with hanging spidermonkey sessions and need some advice on how to debug them. I've made a javascript queue application that uses mod_spidermonkey_socket. It works fine for a while, but after some calls I noticed that calls didnt get transferred to agents. The reason was that earlier calls had not been terminated properly. freeswi...@test1 hupall 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 switch_core_session_hupall() Giving up with 8 sessions remaining API CALL [hupall()] output: +OK hangup all channels with cause MANAGER_REQUEST freeswi...@test1 show calls API CALL [show(calls)] output: 0 total. As you can see, 8 sessions are alive, but none of them are listed as calls. What kind of logs should I turn on to see what is happening with those sessions? Thanks, Jonas _ ___ 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] Country specific tones - how to contribute?
You can submit patches to http://jira.freeswitch.org
thanks,
/b
On Jan 15, 2009, at 1:16 AM, Scott Ellis wrote:
I have tracked down a set of au tones from the mailing list, which I
am
going to verify. How do I go about getting these added into the
default
build so that they are available for all in future?
I tried action application=set data=ringback=${au-ring}/ and
this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
Scott
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Re: [Freeswitch-users] Changes in PlayAndGetDigits
Update and try now... I think we fixed this to not break API compatibility. /b On Jan 15, 2009, at 1:43 AM, Juan Backson wrote: Hi, Is there a change in the playAndGetDigits api? In the old release, 11102, my lua script is working but is not working in the latest release. The error I am getting is Error in playAndGetDigits expected 10..10 args, got 9 . Thanks, JB ___ 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] mod_sofia: NAT-ping RPID bounties
Raymond Chandler napsal(a): kokoska rokoska wrote: Hi all, I have just post two bounties Where did you post these bounties? I have posted them to the Boutny wiki page (at the bottom of the page): http://wiki.freeswitch.org/wiki/Bounty We've started moving bounties away from the wiki and adding them to jira instead. ( so that progress can be followed more closely ) OK. Should I move my bounties or you'll be so kind and move them all (including mine)? :) Best regards, kokoska.rokoska ___ 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] mod_sofia: NAT-ping RPID bounties
kokoska rokoska wrote: Raymond Chandler napsal(a): kokoska rokoska wrote: Hi all, I have just post two bounties Where did you post these bounties? I have posted them to the Boutny wiki page (at the bottom of the page): http://wiki.freeswitch.org/wiki/Bounty We've started moving bounties away from the wiki and adding them to jira instead. ( so that progress can be followed more closely ) OK. Should I move my bounties or you'll be so kind and move them all (including mine)? :) anyone still interested in a bounty posted, should move it to jira and remove it from the wiki most of the ones on the wiki have been done already, iirc http://jira.freeswitch.org/browse/BOUNTY -Ray begin:vcard fn:intralanman n:Chandler;Raymond adr:;;630 Cooks Rd.;Farmville;VA;23901;United States email;internet:intralan...@freeswitch.org tel;cell:+1.434.315.4132 x-mozilla-html:TRUE version:2.1 end:vcard ___ 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] No caller/called ID received (Wildcard X101P)
Could you repeat this test with debug loglevel turned on? (Press F8 or type console loglevel 7). Please put the results in pastebin.freeswitch.org. -MC On Thu, Jan 15, 2009 at 4:35 AM, Tomás tomasborre...@gmail.com wrote: Thank you very much for your help, I've realized I was specting to receive my house phone number having a POTS line and that's not possible. So, I've put my house number in openzap.conf: [span zt] name = OpenZAP number = 91999 fxo-channel = 1 And I've added an extension on the default dialplan: extension name=public_did condition field=destination_number expression=^91999$ action application=answer/ action application=sleep data=2000/ action application=ivr data=demo_ivr/ /condition /extension So I was hopping the IVR answer the call when it is received but instead of that nothing happens, this is the log of one incoming call: 2009-01-15 21:16:56 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999[74cb661e-e341-11dd-acde-9740a65ca868] 2009-01-15 21:16:56 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:16:56 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 13 (OpenZAP/1:1/91999) Ended 2009-01-15 21:16:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] 2009-01-15 21:17:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999 [78fe36b2-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:03 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:17:03 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/914021339 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 14 (OpenZAP/1:1/91999 ) Ended 2009-01-15 21:17:03 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] 2009-01-15 21:17:09 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/91999 [7c60cfea-e341-11dd-acde-9740a65ca868] 2009-01-15 21:17:09 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP-91999 in context default 2009-01-15 21:17:09 [NOTICE] switch_core_state_machine.c:168 switch_core_standard_on_execute() Hangup OpenZAP/1:1/91999 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 15 (OpenZAP/1:1/91999 ) Ended 2009-01-15 21:17:09 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/91999 [CS_HANGUP] Someone knows what's happening? Thank you. On Wed, Jan 14, 2009 at 4:42 PM, Jon Radel j...@radel.com wrote: Tomás wrote: Hi, Anthony, I think that's my problem, when I receive a call from the PSTN, FS receive number 1 instead of my house number and I don't know why. If you use SIP trunking or something like an ISDN-PRI line, the number the call is to is delivered as part of the signaling, which is only way to make use of many phone numbers on a single physical circuit or connection. When you put a POTS line into an FXO port, there is no such information provided, as there is only one number on the line. (Leaving aside various schemes found in some countries such as using different ring patterns to indicate different numbers having been called.) So, as Anthony keeps pointing out, if you want FS to know the number of the line plugged into the FXO port, you have to configure it yourself. --Jon Radel ___ 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] VMWare voice quality
Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? Thanks, Remko -Oorspronkelijk bericht- Van: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Ken Rice Verzonden: donderdag 15 januari 2009 5:05 Aan: freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org Onderwerp: [Freeswitch-users] Announcing the FreeSWITCH Technology PreviewVMWare Appliance. Hey guys, I'm not trying to start 1 a day releases, Things just happened to fall that way... FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac. Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that was updated before we started testing it. 2) FreeSWITCH SVN Trunk is installed and operational. See /etc/motd on the running image for all the good information. We'll be unvailing a wiki page for this shortly. For now you can get the head start by downloading this at http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip Have fun guys! Ken ___ 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] VMWare voice quality
On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ 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] VMWare voice quality
If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice kr...@suspicious.org wrote: On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ 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] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
Thanks Helmut, I cross-checked with our provider. They use national numbering plan for our lines. So this didn't solve our problem. I also ensured that the local language is DE and ZAP timing is dedicated to span 1. I changed the configs to debug mode for OpenZAP, so I hopefully will get some more info on the next failure. Best regards Peter Helmut Kuper schrieb: Hi Peter, it was simply a change in our TDM Voice Switch. It used a different numbering plan and we changed it to national to get it work with FS and openzap in Q921/Q931 mode. What I still search is a way to configure the numberplan in FS. To make it clear: In my case it didn't work from the second FS starts up. So this differs from your problem. To get an idea what's going on on the TDM link I used a TDM D-Channel monitoring device and traced the d-channel messages exchanged between FS and TDM. That should make it easier to see what's wrong when the problems occur. But you can also increase FS debug level to debug and trace the Q921 and Q931 messages in FS console via fs_cli during runtime. You have to set this in openzap.conf.xml: param name=q921loglevel value=debug/ param name=q931loglevel value=debug/ Unfortunately FS doesn't decode the whole Q931 messages, but it shows a hex representation of the message, so you can manually decode it with this documents: Q.931: http://www.itu.int/rec/T-REC-Q.931-199805-I/en Q.921: http://www.itu.int/rec/T-REC-Q.921-199709-I/en I think for numberingplan issues you only have to track the Q.931 messages. The last idea I have to get some light into your problem and to avoid manually decoding, try to convert FS's q931 hexdump into wiresharks pcap format. Wireshark should be able to decode it :) http://wiki.wireshark.org/Q.931 Maybe it's a good idea to implement a wireshark export for those messages in FS. This will make debugging easy and cheap. Hope it helps a bit. regards helmut Am 15.01.2009 12:06, schrieb Peter P GMX: Helmut, can you give me a hint, how you worked around this? Best regards Peter Helmut Kuper schrieb: Hi Michael, it must not be the case here, but I had the same error, when incomming calles used a wrong numbering plan resp not the one, FS expected. Just a hint. regards Helmut Am 15.01.2009 09:20, schrieb Peter P GMX: Hello Michael, how much $$ are we talking about? I need this issue to be solved quickly and it's worth to spend some money. I've read the following post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg05792.html and have the same symptom with after hundreds of calls I start to get b channels that are stuck in states like TERMINATING or HANGUP Best regards Peter Michael Collins schrieb: I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -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 ___ 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] VMWare voice quality
I have been running FreeSWITCH on a VM ever since I got involved in the project. It's been almost a year now. I didn't do anything special - it works fine. I get audio problems if I go over 10 or 15 simultaneous calls. This is on the following setup: VMWare Server 1.0.6 and VMWare Server 2.0 (2.0 sucks, btw /rant) Dell Precision 360 (Desktop) Pentium 4 2.66Ghz 2.5GB RAM (512MB allocated to FS) Fedora Core 8, 2.6.23.1-42.fc8 stock kernel (a bit old) 7.2K 80GB hard drive Yes, fancy machine I have, huh? This is my normal day-to-day workstation as well as my VMWare Server. It works fine, I got occassional missed heartbeat alerts and timer sync notices, but they're rare. - Darren -Original Message- From: Michael Collins [mailto:m...@freeswitch.org] Sent: Thursday, January 15, 2009 9:32 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] VMWare voice quality If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice kr...@suspicious.org wrote: On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ 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-use rs 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] Using mod_managed Linux/Mono
Thanks Michael, that did get me a little further. I renamed mod_managed_lib.dll to FreeSWITCH.Managed.dll and that definitely had an effect. but now when I attempt to load mod_managed FreeSwitch core dumps now. I have tried mono 2.2 and mono 2.0.1 I am running . CentOS 5.2 x86 32bit [r...@sipcore-alpha conf]# uname -a Linux sipcore-alpha 2.6.18-92.el5 #1 SMP Tue Jun 10 18:49:47 EDT 2008 i686 athlon i386 GNU/Linux I attached the full output from the console in the txt doc attached above. I'm wondering if the problem is specific to this flavor of linux. perhaps more specifically the kernel. -Adam From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: Thursday, January 15, 2009 12:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 The managed assembly should be the same on both platforms. The correct name is FreeSWITCH.Managed.dll. I'll get a patch to the mod_managed/managed/Makefile. Meanwhile, simply renaming mod_managed_lib.dll should work. After that, make sure there's a managed subdirectory where the modules are. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Adam Long Sent: Wednesday, January 14, 2009 3:45 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Using mod_managed Linux/Mono 2.02 Has anyone had any luck using mod_managed under linux with mono yet? The Wiki looks to still be lacking some linux installation instructions. I feel like I'm close but missing something simple. I got as far as adding languages/mod_managed to the /usr/src/freeswitch-1.0.2/modules.conf without quotes obviously. My installed mono version is [r...@sipcore-alpha mod]# mono -V Mono JIT compiler version 2.2 (tarball Wed Jan 14 09:44:57 PST 2009) Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com TLS: __thread GC:Included Boehm (with typed GC) SIGSEGV: altstack Notifications: epoll Architecture: x86 Disabled: none I can successful compile freeswitch and it indeed compiles mod_managed.so I added load module=mod_managed / to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml I did also create the /usr/local/freeswitch/mod/managed directory as stated in the wiki as requirement. But when I start freeswitch I get the following in regards to the mod_managed loading. 2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading mod_managed (Common Language Infrastructure), Mono Version 2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling mono_assembly_loaded. 2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling mono_domain_assembly_open. 2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed. 2009-01-14 14:19:12 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so **Module load routine returned an error** One thing I think I may be missing is a FreeSWITCH.Managed.dll (this exists on windows environment but doesn't seem to be compiled under linux) I thought perhaps mod_managed_lib.dll was the linux equivalent but that exists and still no a no go. Any ideas would be very welcome? Thank you! Regards, -Adam freeswi...@sipcore-alpha load mod_managed 2009-01-15 09:46:59 [INFO] mod_managed.cpp:309 mod_managed_load() Loading mod_managed (Common Language Infrastructure), Mono Version 2009-01-15 09:46:59 [INFO] mod_managed.cpp:213 loadRuntime() Calling mono_assembly_loaded. 2009-01-15 09:46:59 [INFO] mod_managed.cpp:217 loadRuntime() Calling mono_domain_assembly_open. Stacktrace: Native stacktrace: /usr/lib/libmono.so.0 [0x3c020bd] /usr/lib/libmono.so.0 [0x3c21640] /usr/lib/libmono.so.0 [0x3ba349d] [0x661440] /usr/local/freeswitch/mod/mod_managed.so(mod_managed_load+0xe5) [0x1001365] /usr/local/freeswitch/lib/libfreeswitch.so.1 [0x8e7ab5] /usr/local/freeswitch/mod/mod_commands.so [0x3484b4] /usr/local/freeswitch/lib/libfreeswitch.so.1(switch_api_execute+0xbd) [0x8e3c5d] /usr/local/freeswitch/lib/libfreeswitch.so.1 [0x8c97a6] /usr/local/freeswitch/lib/libfreeswitch.so.1 [0x8c99da] /usr/local/freeswitch/lib/libfreeswitch.so.1 [0x93da26] /lib/libpthread.so.0 [0x52d45b] /lib/libc.so.6(clone+0x5e) [0x484e5e] Debug info from gdb: Using host libthread_db library /lib/libthread_db.so.1. [Thread debugging using libthread_db enabled] [New Thread -1208415792 (LWP 13930)] [New Thread 35236752 (LWP 13985)] [New Thread 8145808 (LWP 13984)] [New Thread -1420170352 (LWP
Re: [Freeswitch-users] mod_sofia: NAT-ping RPID bounties
Raymond Chandler napsal(a): kokoska rokoska wrote: Raymond Chandler napsal(a): kokoska rokoska wrote: Hi all, I have just post two bounties Where did you post these bounties? I have posted them to the Boutny wiki page (at the bottom of the page): http://wiki.freeswitch.org/wiki/Bounty We've started moving bounties away from the wiki and adding them to jira instead. ( so that progress can be followed more closely ) OK. Should I move my bounties or you'll be so kind and move them all (including mine)? :) anyone still interested in a bounty posted, should move it to jira and remove it from the wiki most of the ones on the wiki have been done already, iirc http://jira.freeswitch.org/browse/BOUNTY OK, I do it ASAP :-) Best regards, kokoska.rokoska ___ 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] waitForAnswer on the Socket Interface
Hi, Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? That is, when i call a device, i want to block until the call is completely answered (not just early media). Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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] waitForAnswer on the Socket Interface
Then originate the call with {ignore_early_media=true}sofia/blah/blah,
It will not return till its actually answered.
/b
On Jan 15, 2009, at 12:55 PM, Klaus Teller wrote:
Hi,
Can somebody tell me how to achieve the same behavuior as
session.waitForAnswer via the socket interface?
That is, when i call a device, i want to block until the call is
completely answered (not just early media).
Thanks,
Klaus.
--
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Re: [Freeswitch-users] waitForAnswer on the Socket Interface
Klaus, What is your dialstring? If you ignore_early_media=true then I believe it will have the same net effect, but it would be good to know exactly what you're hoping to accomplish. -MC On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? That is, when i call a device, i want to block until the call is completely answered (not just early media). Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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] Problem connecting with gtalk
Hello, I'm running fs 1.0.2 on CentOS 5.2 I've been trying to setup my fs to talk with googletalk following the instructions in http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration I got the error of TLS not supported so i: INSTALLED: yum install gnutls-devel gnutls REMOVED: rm -f /usr/src/freeswitch-1.0.2/libs/iksemel/.complete rm -f /usr/src/freeswitch-1.0.2/libs/libdingaling/.complete RE-INSTALLED cd /usr/src/freeswitch-1.0.2/ make sure make installall Even after this i keep getting the same error on the console. This is the error: 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: --- stream:stream from=gmail.com id=E607DA864F13575A version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client/stream:stream 2009-01-15 14:06:02 [DEBUG] libdingaling.c:1175 on_stream() TLS NOT SUPPORTED IN THIS BUILD! 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: --- stream:features starttls xmlns=urn:ietf:params:xml:ns:xmpp-tls required/required /starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-sasl mechanismX-GOOGLE-TOKEN/mechanism /mechanisms /stream:features and now when i *shutdown* my fs, the core gets dumped when trying to stop mod_dingaling: 2009-01-15 14:14:21 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_dingaling Segmentation fault Thank you for any help or suggestions you can give me. ___ 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 crashed !!!
On 01/13/2009 04:00 PM, Anthony Minessale wrote: So I can supply you with 250 thousand lines of C code that make your application possible. but you are not willing to show me the silly js code that may be the cause of your crash? What security purposes are you kidding? I just need to salute this! I get same silly reasons day by day, everyone wants their issues fixed in no time without proper (any) feedback. Maybe we should collect and build a top of such reasons... Cheers, Daniel just rename any sensitive data to something else or stop using js because without seeing the script code that's all I can tell you as the solution to your problem. -- Daniel-Constantin Mierla http://www.asipto.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] Problem connecting with gtalk
install gnutls and dev packages and reconfigure/recompile /b On Jan 15, 2009, at 1:21 PM, Milena wrote: Hello, I'm running fs 1.0.2 on CentOS 5.2 I've been trying to setup my fs to talk with googletalk following the instructions in http://wiki.freeswitch.org/wiki/Mod_dingaling#Sample_Configuration I got the error of TLS not supported so i: INSTALLED: yum install gnutls-devel gnutls REMOVED: rm -f /usr/src/freeswitch-1.0.2/libs/iksemel/.complete rm -f /usr/src/freeswitch-1.0.2/libs/libdingaling/.complete RE-INSTALLED cd /usr/src/freeswitch-1.0.2/ make sure make installall Even after this i keep getting the same error on the console. This is the error: 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: --- stream:stream from=gmail.com id=E607DA864F13575A version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client/stream:stream 2009-01-15 14:06:02 [DEBUG] libdingaling.c:1175 on_stream() TLS NOT SUPPORTED IN THIS BUILD! 2009-01-15 14:06:02 [INFO] libdingaling.c:1304 on_log() RECV: --- stream:features starttls xmlns=urn:ietf:params:xml:ns:xmpp-tls required/required /starttls mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-sasl mechanismX-GOOGLE-TOKEN/mechanism /mechanisms /stream:features and now when i *shutdown* my fs, the core gets dumped when trying to stop mod_dingaling: 2009-01-15 14:14:21 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_dingaling Segmentation fault Thank you for any help or suggestions you can give me. ___ 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] waitForAnswer on the Socket Interface
Thanks folks! ignore_early_media=true solves my problem. The dialstring was just sofia/gateway/blah/blah. Klaus. Original-Nachricht Datum: Thu, 15 Jan 2009 11:06:57 -0800 Von: Michael Collins m...@freeswitch.org An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] waitForAnswer on the Socket Interface Klaus, What is your dialstring? If you ignore_early_media=true then I believe it will have the same net effect, but it would be good to know exactly what you're hoping to accomplish. -MC On Thu, Jan 15, 2009 at 10:55 AM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Can somebody tell me how to achieve the same behavuior as session.waitForAnswer via the socket interface? That is, when i call a device, i want to block until the call is completely answered (not just early media). Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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 crashed !!!
Daniel-Constantin Mierla wrote: On 01/13/2009 04:00 PM, Anthony Minessale wrote: So I can supply you with 250 thousand lines of C code that make your application possible. but you are not willing to show me the silly js code that may be the cause of your crash? What security purposes are you kidding? I just need to salute this! I get same silly reasons day by day, everyone wants their issues fixed in no time without proper (any) feedback. Maybe we should collect and build a top of such reasons... Just set the bug to UTR and move on. No reason to berate the person who won't supply the script required to reproduce the problem... it just won't get fixed without both a way to reproduce and a developer who cares to dig into finding it. Could you imagine a large software company saying anything other than you have not supplied enough information for us to reproduce this bug? Between the time wasted writing a longer response, and the image it creates for clueless users/customers of the developers and the support process, it just isn't worth it. Matthew Kaufman ___ 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] Problem connecting with gtalk
Hello, Isn't that what I did? if not, what is the right way to install gnutls and dev packages and reconfigure/recompile thank you ___ 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] Problem connecting with gtalk
Well the right way is depends on your distro. Once you have it installed I would ./bootstrap.sh and ./configure again to be safe. /b On Jan 15, 2009, at 1:39 PM, Milena wrote: Hello, Isn't that what I did? if not, what is the right way to install gnutls and dev packages and reconfigure/recompile thank you ___ 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] Freeswitch and CELT:
Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much! ___ 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] VMWare voice quality
You'll never fix this. Voice is a latency specific application unless you figure out how to manipulate time. Any virtualization platform is going to provide less timing granularity than raw hardware. Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? ___ 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 crashed !!!
Matthew, I am not berating him, I am trying to convince him to give me the script that causes his crash. It seems ridiculous to me that he should be worried about what I will do with his js code when I am clearly only interested in finding out what causes his issue. And there is irony for him to think that I can write FS itself in C then need his js code for anything i want to accomplish. He can easily remove any sensitive information from the script before supplying it. Why exactly are you so abrasive in our community. We just spent like 3 days trying to help you with an issue didn't we? And it was not even a problem in FS itself. Also, why are you comparing us to a company we are not a company we are an open source project and everything is free, what exactly are you expecting? We all spend most of our day helping people here including you on multiple occasions and you seem to repeatedly criticize us for who knows why. I am not mad about your comment I just don't get it. On Thu, Jan 15, 2009 at 1:39 PM, Matthew Kaufman matt...@matthew.at wrote: Daniel-Constantin Mierla wrote: On 01/13/2009 04:00 PM, Anthony Minessale wrote: So I can supply you with 250 thousand lines of C code that make your application possible. but you are not willing to show me the silly js code that may be the cause of your crash? What security purposes are you kidding? I just need to salute this! I get same silly reasons day by day, everyone wants their issues fixed in no time without proper (any) feedback. Maybe we should collect and build a top of such reasons... Just set the bug to UTR and move on. No reason to berate the person who won't supply the script required to reproduce the problem... it just won't get fixed without both a way to reproduce and a developer who cares to dig into finding it. Could you imagine a large software company saying anything other than you have not supplied enough information for us to reproduce this bug? Between the time wasted writing a longer response, and the image it creates for clueless users/customers of the developers and the support process, it just isn't worth it. Matthew Kaufman ___ 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/ 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
Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality
That won't eliminate the problem. Just reduce the possibility of it happening. Trust me... I've got a large ESX infrastructure, and there is no way that a software based Voice platform is going to provide skip free audio in a virtualized environment. -Original Message- From: freeswitch-dev-boun...@lists.freeswitch.org [mailto:freeswitch- dev-boun...@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, January 15, 2009 12:15 PM To: freeswitch-users@lists.freeswitch.org; Remko Kloosterman; freeswitch-...@lists.freeswitch.org Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ Freeswitch-dev mailing list freeswitch-...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. ___ 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] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]
I did some more tests. When I sequentially setup calls (only one simultaneous call at a time), it works for hundreds of calls. As soon as I setup 2 calls in parallel ist fails aber a number of calls. Please find another debug ouput (now with Q.921 debug also). The log starts with the latest hangup of a successfull call. After this one I receive a 2009-01-15 20:26:46 [CRIT] ozmod_isdn.c:446 zap_isdn_931_34() Received Release with no matching channel 0 and later 2009-01-15 20:26:46 [DEBUG] ozmod_isdn.c:781 zap_isdn_921_23() 931 parse error [-3012] [Q931E_INVALID_CRV] Is there anyone to fix it? May I donate some money for fixing that? Best regards Peter Debug: 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame - Q.921 Packet [Outgoing] --- SAPI: 0, TEI: 0, C/R: Command (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] Q.921 state: Multiple Frame Mode Established (7) [flags: ] -- 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:813 state_advance() 2:3 STATE [TERMINATING] 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1121 state_advance() Terminating: Direction Inbound 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:1418 on_clear_channel_signal() got clear channel sig [STOP] 2009-01-15 20:26:44 [NOTICE] mod_openzap.c:1437 on_clear_channel_signal() Hangup OpenZAP/2:3/21658519 [CS_EXECUTE] [NORMAL_CLEARING] 2009-01-15 20:26:44 [DEBUG] switch_channel.c:1513 switch_channel_perform_hangup() Send signal OpenZAP/2:3/21658519 [KILL] 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal OpenZAP/2:3/21658519 [BREAK] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Receiving message from Layer4 (size: 184, type: 77) 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Sending message to Q.921 (size: 184) 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 126 (0x7e), CRVflag: 1 (0x1), MesType: 77 (0x4d) 2009-01-15 20:26:44 [DEBUG] ozmod_isdn.c:1529 q931_rx_32() WRITE 5 [08 02 80 7e 4d] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Got frame from Q.931, type: 4, tei: 0, size: 9 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Enqueueing I frame for TEI 0 [0] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() Sending frame - Q.921 Packet [Outgoing] --- SAPI: 0, TEI: 0, C/R: Command (0) Type: I Frame P/F: 0, N(S): 80, N(R): 81 [V(A): 80, V(R): 81, V(S): 80] Q.921 state: Multiple Frame Mode Established (7) [flags: ] -- 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 (timeout: 1000 msecs) started for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 stopped for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.931() Q931Rx43 return code: 1 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1922 listener_run() Session complete, waiting for children 2009-01-15 20:26:44 [DEBUG] mod_event_socket.c:1946 listener_run() Connection Closed 2009-01-15 20:26:44 [DEBUG] switch_ivr_play_say.c:1222 switch_ivr_play_file() done playing file 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (OpenZAP/2:3/21658519) State EXECUTE going to sleep 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (OpenZAP/2:3/21658519) Running State Change CS_HANGUP 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP 2009-01-15 20:26:44 [DEBUG] mod_openzap.c:472 channel_on_hangup() OpenZAP/2:3/21658519 CHANNEL HANGUP 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/2:3/21658519 Standard HANGUP, cause: NORMAL_CLEARING 2009-01-15 20:26:44 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (OpenZAP/2:3/21658519) State HANGUP going to sleep 2009-01-15 20:26:44 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Locked, Waiting on external entities 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 251 (OpenZAP/2:3/21658519) Ended 2009-01-15 20:26:44 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel OpenZAP/2:3/21658519 [CS_HANGUP] 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() New packet received (4 bytes) - Q.921 Packet [Incoming] --- SAPI: 0, TEI: 0, C/R: Response (0) Type: S Frame, SV: RR (Receive Ready) P/F: 0, N(R): 81 [V(A): 80, V(R): 81, V(S): 81] Q.921 state: Multiple Frame Mode Established (7) [flags: ] -- 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T200 stopped for TEI 0 2009-01-15 20:26:44 [DEBUG] Span:0 Q.921() T203 (timeout: 1 msecs) restarted for TEI 0 2009-01-15
Re: [Freeswitch-users] VMWare voice quality
Michael Collins wrote: If anyone figures this out please post it to this thread. I am working on a wiki page for the VMWare appliance and I would like to be able to inform people on how to handle this situation. Also, IIUC, those running VMWare Fusion on Macs are not experiencing this, correct? What about those using a hypervisor like ESXi? Any known issues? Thanks, MC On Thu, Jan 15, 2009 at 9:15 AM, Ken Rice kr...@suspicious.org wrote: On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ 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 Hi All, I'm using freeswitch in production environment running on ESXi . I have no issues with voice /probably because simply i leave the media to flow between endpoints/ . Performance is amazing and i'd recommend this setup to everybody. it is important though when you set your VM on ESXi to set in advance the number of CPUs. Changing # of CPUs later might affect your performance. My recommendation is NOT to use VMWARE server on top of other OS. ESXi as hipervisor is linux in its core that provides you with enough access to the HW and nothing more so the overhead is as minimal as possible /while this is not the case fro VMware server - it needs underlaying OS and so on/. I hope this info helps. If anybody is interested i'd be glad to share me experience on his matter. Best Regards Chav ___ 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] Problem connecting with gtalk
Oops, it took me a little while to realize what you meant and why make alone wouldn't work, thank you very much sir, it all works fine now. 2009/1/15 Milena testeado...@gmail.com Hello, Isn't that what I did? if not, what is the right way to install gnutls and dev packages and reconfigure/recompile thank you ___ 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-dev] VMWare voice quality
To the contrary, we have had quite good results in virtualized environments and you don't really need timing that is that accurate to make it work. We work quite well on amazon EC2 for example. There are 2 issues I know about with vmware, 1 is you need to set a setting on the host to extend somewhat sane clocks being available, the second is I have seen issues with the bridged network adapter actually doubling up all packets causing very strange issues, I suggest not using bridged networking if you experience this. Mike On Jan 15, 2009, at 3:12 PM, Gregory Boehnlein wrote: That won't eliminate the problem. Just reduce the possibility of it happening. Trust me... I've got a large ESX infrastructure, and there is no way that a software based Voice platform is going to provide skip free audio in a virtualized environment. -Original Message- From: freeswitch-dev-boun...@lists.freeswitch.org [mailto:freeswitch- dev-boun...@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, January 15, 2009 12:15 PM To: freeswitch-users@lists.freeswitch.org; Remko Kloosterman; freeswitch-...@lists.freeswitch.org Subject: Re: [Freeswitch-dev] [Freeswitch-users] VMWare voice quality On 1/15/09 11:01 AM, Remko Kloosterman r.klooster...@mtel.nl wrote: Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? There is a high resolution timer you need to enable on vmware... I'm not familiar enuff with all the versions of vmware to advise there that switch is, but they have a couple of articles on it in their knowledge base ___ 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-dev] VMWare voice quality
To the contrary, we have had quite good results in virtualized environments and you don't really need timing that is that accurate to make it work. If you don't handle RTP, I'm sure it is amazing. However, if you have to do voicemail, stream audio from the server or do any kind of actual time/latency/jitter sensitive processing, I don't care how much you tune your hypervisor, it's never going to scale. We work quite well on amazon EC2 for example. There are 2 issues I know about with vmware, 1 is you need to set a setting on the host to extend somewhat sane clocks being available, the second is I have seen issues with the bridged network adapter actually doubling up all packets causing very strange issues, I suggest not using bridged networking if you experience this. I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on Vmware Server or Workstation? ___ 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] VMWare voice quality
Hello all, let me also give some experience from the VirtualBox side (Community Version). Host machine == AMD X2 64 3800 with 8GB of RAM OS is a generic Debian 4.0R5 with Kernel 2.6.18-6-amd64 No special parameters in the Kernel. Started with VirtualBox 1.5 and now on 2.0.x Client machine (freeswitch) === Ubuntu 8.041 Generic Kernel 2.6.24-18-generic #1 SMP, 1 CPU Experience: === A single call produces about 20% CPU load. So this is not usefull for any production environment. I did not discover any dropouts in a normal call between internal and external UAs/gateways since 6 months. So for testing purposes its fine. Voice between User Agents is always fine. Seldomly I hear choppy voice when announcements are played. After some minutes these problems go away. Resume === For testing/development purposes, FS on VirtualBox is fine. For any productive environment it's not really usable in our environment. Comparison with Asterisk = Asterisk never worked in this environment: Choppe voice between UAs and when playing sound. 100% CPU load on a single call. == Best regards Peter Gregory Boehnlein schrieb: You'll never fix this. Voice is a latency specific application unless you figure out how to manipulate time. Any virtualization platform is going to provide less timing granularity than raw hardware. Hello Ken, hello all, I just read about the FreeSWITCH VMware applicance. I'm curious about your experiences with the audio quality on VMWare, so here's a new thread. I've installed freeswitch on VMware Server for Windows. The IVR audio always plays choppy, while the server itself has no performance issues. The same poor voice quality also goes for Asterisk or Yate, even on a very fast VMware ESX system. Did you experience the same and/or do you have pointers on how to troubleshoot and fix this? ___ 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-dev] VMWare voice quality
We have people running FreeSWITCH in vmware and xen with media and considerable load and it doesn't have a problem. We also work very well inside OpenVZ. /b On Jan 15, 2009, at 2:37 PM, Gregory Boehnlein wrote: If you don't handle RTP, I'm sure it is amazing. However, if you have to do voicemail, stream audio from the server or do any kind of actual time/latency/jitter sensitive processing, I don't care how much you tune your hypervisor, it's never going to scale. ___ 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-dev] VMWare voice quality
We have people running FreeSWITCH in vmware and xen with media and considerable load and it doesn't have a problem. We also work very well inside OpenVZ. I'd be very interested in seeing that, and knowing how it was done. ___ 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-dev] VMWare voice quality
Ok if can summarize a little of the intention of releasing this VMWare image. Its really there so you guys can get it and check it out. I personally don't believe in running such services on a virtual machine (too many nightmare stories from the 'day job' from such things) However, for testing and developing applications that ride on top of FreeSWITCH, this is a quick way to get up and running. Remember Voice application especially where you are interacting with the media streams will be affected by latency and jitter much more readily then store and forward things like IRC, Web, eMail and instant messaging. K On 1/15/09 2:12 PM, Gregory Boehnlein da...@nacs.net wrote: That won't eliminate the problem. Just reduce the possibility of it happening. Trust me... I've got a large ESX infrastructure, and there is no way that a software based Voice platform is going to provide skip free audio in a virtualized environment. ___ 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-dev] VMWare voice quality
On that note the OpenVZ instances could live migrate from box to box without dropping calls and usually had a small acceptable blip in audio. /b On Jan 15, 2009, at 2:59 PM, Gregory Boehnlein wrote: We have people running FreeSWITCH in vmware and xen with media and considerable load and it doesn't have a problem. We also work very well inside OpenVZ. I'd be very interested in seeing that, and knowing how it was don ___ 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-dev] VMWare voice quality
On Jan 15, 2009, at 1:02 PM, Brian West br...@freeswitch.org wrote: On that note the OpenVZ instances could live migrate from box to box without dropping calls and usually had a small acceptable blip in audio. I'd say a small blip is quite acceptable compared to the alternative! -MC /b On Jan 15, 2009, at 2:59 PM, Gregory Boehnlein wrote: We have people running FreeSWITCH in vmware and xen with media and considerable load and it doesn't have a problem. We also work very well inside OpenVZ. I'd be very interested in seeing that, and knowing how it was don ___ 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-dev] VMWare voice quality
On that note the OpenVZ instances could live migrate from box to box without dropping calls and usually had a small acceptable blip in audio. OpenVZ is not a hypervisor. It essentially runs all of it's applications natively on the CPU. I would expect that it would work under OpenVZ or other container based (chrooted / jailed setups) well. ___ 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] No Sound Heared
Hi, Need your help on this. I have the following Javascript statement: session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a file called gafachiDialout.js Then, i have the following extension in default.xml: extension name=6337 condition field=destination_number expression=^6337$ action application=javascript data=gafachiDialout.js / /condition /extension When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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] No Audio when dial out via gateway
Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. I just hope that it would be something easy to be pointed out by experienced users. Thank you, Will ___ 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] No Audio when dial out via gateway
did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. I just hope that it would be something easy to be pointed out by experienced users. Thank you, Will ___ 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] No Audio when dial out via gateway
Thank you so much for responding. Yes, I checked those, everything looks fine, and infact, if the audio stream is blocked by firewall or nat setting, how can I inject the audio file and hear it played on both ends. But as you suggest, I will doublecheck those values. Thanks again --- On Thu, 1/15/09, Brian West br...@freeswitch.org wrote: From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] No Audio when dial out via gateway To: freeswitch-users@lists.freeswitch.org Date: Thursday, January 15, 2009, 1:46 PM did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. I just hope that it would be something easy to be pointed out by experienced users. Thank you, Will ___ 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] Starting FS on OSX
Hello again FreeSwitchers, I have built the 1.02 on 10.4.11(OSX) and had no problems with that. I have never been able to build from the SVN, but that is another story. Now that I have migrated to 1.02 I was wondering if I can get some help on a long standing issue I have with starting FS at boot. I am hoping to use Launchd which is the standard on OSX 10.4 and I attempted to cobble together a script, but haven't had great results. I did search for wiki entries on this, but haven't found any help with it. Ideas? Thanks, Marty ___ 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-dev] VMWare voice quality
Lot's of experience and suggestions here. Thanks. I believe it should be theoretically possible to have blip-free RTP streaming through the appliance. Most windows ethernet drivers allow for QoS packet scheduling. If the VMware network bridge driver honors this and syncs the buffers at 20ms frames (or whatever frame size applies) you should be able to schale up a bit and maintain low jitter. Anyone knows how the VMware network bridge exactly works? -Oorspronkelijk bericht- Van: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Gregory Boehnlein Verzonden: donderdag 15 januari 2009 21:37 Aan: freeswitch-users@lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality To the contrary, we have had quite good results in virtualized environments and you don't really need timing that is that accurate to make it work. If you don't handle RTP, I'm sure it is amazing. However, if you have to do voicemail, stream audio from the server or do any kind of actual time/latency/jitter sensitive processing, I don't care how much you tune your hypervisor, it's never going to scale. We work quite well on amazon EC2 for example. There are 2 issues I know about with vmware, 1 is you need to set a setting on the host to extend somewhat sane clocks being available, the second is I have seen issues with the bridged network adapter actually doubling up all packets causing very strange issues, I suggest not using bridged networking if you experience this. I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on Vmware Server or Workstation? ___ 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] No Audio when dial out via gateway
I found this: When I call the outside number, first, cannot hear or be heard, then when I put the line on hold, the other party can hear the MOH, and when I switch it back, now we can talk. Something goes wrong here --- On Thu, 1/15/09, Brian West br...@freeswitch.org wrote: From: Brian West br...@freeswitch.org Subject: Re: [Freeswitch-users] No Audio when dial out via gateway To: freeswitch-users@lists.freeswitch.org Date: Thursday, January 15, 2009, 1:46 PM did you check our firewall? and various nat settings? /b On Jan 15, 2009, at 3:42 PM, Will Smith wrote: Hi, I have successfully installed and configured the FS thanks to the community help. Greatly appreciate all. Now I have some basic error: I can dial out from extension 1000 (all default ext) to any number not in the same network. I got the other number rung, and answered, but cannot hear anything from both ends. Strange thing is I can broadcast an audio into the conversation, and both ends can hear the audio, but just cannot talk. I just hope that it would be something easy to be pointed out by experienced users. Thank you, Will ___ 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] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent - a solution of sorts.
open a jira and attach a svn diff and we'll have a look
thanks
On Thu, Jan 15, 2009 at 5:14 AM, Scott Ellis scott.el...@novatex.com.auwrote:
So I decided to hack the code to see if I could just get it to do what I
wanted - assuming some kind of error in the options setting.
First I changed the state change code to just skip straight to IDLE
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN !zap_test_flag(event-channel,
ZAP_CHANNEL_INTHREAD))) {
// if (zap_test_flag(analog_data,
ZAP_ANALOG_CALLERID)) {
// zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_GET_CALLERID);
// } else {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_IDLE);
// }
event-channel-ring_count = 1;
zap_mutex_unlock(event-channel-mutex);
locked = 0;
zap_thread_create_detached(zap_analog_channel_run, event-channel);
} else {
event-channel-ring_count++;
}
So we skip the GET_CALLERID state altogether.
This generated an illegal state change message cannot go from DOWN to IDLE
So then changed the code to
if (!event-channel-ring_count (event-channel-state ==
ZAP_CHANNEL_STATE_DOWN !zap_test_flag(event-channel,
ZAP_CHANNEL_INTHREAD))) {
// if (zap_test_flag(analog_data,
ZAP_ANALOG_CALLERID)) {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_GET_CALLERID);
// } else {
zap_set_state_locked(event-channel,
ZAP_CHANNEL_STATE_IDLE);
// }
event-channel-ring_count = 1;
zap_mutex_unlock(event-channel-mutex);
locked = 0;
zap_thread_create_detached(zap_analog_channel_run, event-channel);
} else {
event-channel-ring_count++;
}
Allowing the state change to GET_CALLERID, then immediately to IDLE.
This works perfectly - the call is answered straight away. At the moment
I don't know enough about linux debugging to step through the parameter
code to see why setting get caller ID to false in openzap.conf.xml does
not get passed through, but even if it does the current code will still
run into the illegal state change error.
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][1:1] STATE [DOWN]
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:760 process_event() Changing
state on 1:1 from DOWN to GET_CALLERID
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:762 process_event() Changing
state on 1:1 from GET_CALLERID to IDLE
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:239 zap_analog_channel_run()
ANALOG CHANNEL thread starting.
2009-01-15 21:59:18 [DEBUG] ozmod_analog.c:410 zap_analog_channel_run()
Executing state handler on 1:1 for IDLE
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1165 on_fxo_signal() got FXO
sig 1:1 [START]
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:340 tech_init() Set codec PCMU
20ms
2009-01-15 21:59:18 [DEBUG] mod_openzap.c:1137 zap_channel_from_event()
Connect inbound channel OpenZAP/1:1/1
2009-01-15 21:59:18 [NOTICE] switch_channel.c:565
switch_channel_set_name() New Channel OpenZAP/1:1/1
[8e2a55c8-e2f3-11dd-adfd-6d934f226ffd]
Will go and put this into JIRA in the next couple of days.
Scott
Scott Ellis wrote:
After poking around in the code, it looks like if I set param
name=enable-callerid value=false/ in openzap.conf.xml, it should
skip the GET_CALLERID state, and I should get the call answered straight
away.
mod_openzap.c
} else if (!strcasecmp(var, enable-callerid)) {
enable_callerid = val;
if (zap_configure_span(analog, span, on_analog_signal,
tonemap, tonegroup,
digit_timeout, to,
max_dialstr, max,
hotline, hotline,
enable_callerid, enable_callerid,
TAG_END) != ZAP_SUCCESS) {
zap_log(ZAP_LOG_ERROR, Error starting OpenZAP span
%d\n, span_id);
continue;
}
ozmod_analog.c
else if (!strcasecmp(var, enable_callerid)) {
if (!(val = va_arg(ap, char *))) {
break;
}
if (zap_true(val)) {
flags |= ZAP_ANALOG_CALLERID;
} else {
flags = ~ZAP_ANALOG_CALLERID;
}
and
case ZAP_OOB_RING_START:
Re: [Freeswitch-users] Starting FS on OSX
Have you looked at creating a system level startup item in /Library/StartupItems ? Also, to build from source you need the latest DevTools Kit from apple installed. (I don't know if the latest will work w/ 10.4) Ken On 1/15/09 3:54 PM, Martin Joseph ast...@stillnewt.org wrote: Hello again FreeSwitchers, I have built the 1.02 on 10.4.11(OSX) and had no problems with that. I have never been able to build from the SVN, but that is another story. Now that I have migrated to 1.02 I was wondering if I can get some help on a long standing issue I have with starting FS at boot. I am hoping to use Launchd which is the standard on OSX 10.4 and I attempted to cobble together a script, but haven't had great results. I did search for wiki entries on this, but haven't found any help with it. Ideas? Thanks, Marty ___ 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] Starting FS on OSX
Your build issue is with your autotools install, I have seen issues if you have ever installed any of the autotools from macports or fink. If you want to build from svn you can run bootstrap on another box (a linux box perhaps) and then tar up that dir and move it to your mac. We pre-bootstrap the release tarballs which is why that is building fine for you. MIke On Jan 15, 2009, at 4:54 PM, Martin Joseph wrote: Hello again FreeSwitchers, I have built the 1.02 on 10.4.11(OSX) and had no problems with that. I have never been able to build from the SVN, but that is another story. Now that I have migrated to 1.02 I was wondering if I can get some help on a long standing issue I have with starting FS at boot. I am hoping to use Launchd which is the standard on OSX 10.4 and I attempted to cobble together a script, but haven't had great results. I did search for wiki entries on this, but haven't found any help with it. Ideas? Thanks, Marty ___ 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 crashed !!!
Could you imagine a large software company saying anything other than you have not supplied enough information for us to reproduce this bug? Between the time wasted writing a longer response, and the image it creates for clueless users/customers of the developers and the support process, it just isn't worth it. Well, when FS is a large company and it's a thing on paid support... ___ 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] mod_managed failing to load on CentOS 5.2
Hi Tim, I'm having exact same problem, try renaming mod_managed_lib.dll to FreeSWITCH.Managed.dll and then load. Michael confirmed this is supposed to be the case and is building a patch for the Makefile. However, when I do this on my Cent OS 5.2 it now loads successfully but immediately I get a core dump. I'm curious if you will have the same problem or not. Regards, -Adam From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Tim B Sent: Wednesday, January 14, 2009 8:13 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] mod_managed failing to load on CentOS 5.2 Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [r...@phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [r...@phone2 mod_managed]# make [r...@phone2 mod_managed]# make install 3) Added to modules.conf.xml : load module=mod_managed/ 4) Started freeswitch from command line ... Error: 2009-01-14 20:01:42 [ERR] mod_managed.cpp:220 loadRuntime() mono_domain_assembly_open failed. 2009-01-14 20:01:42 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_managed.so **Module load routine returned an error** 5) I know mono2 is working because I compiled and executed a helloworld test class on machine. Any ideas? _ Windows LiveT: Keep your life in sync. See how it works. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t1_allup_howitworks_012 009 ___ 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] mod_managed failing to load on CentOS 5.2
/freeswitch/mod/managed directory as stated in the
wiki as requirement. But when I start freeswitch I get the following in
regards to the mod_managed loading... 2009-01-14 14:19:12 [INFO]
mod_managed.cpp:309 mod_managed_load() Loading mod_managed (Common Language
Infrastructure), Mono Version 2009-01-14 14:19:12 [INFO] mod_managed.cpp:213
loadRuntime() Calling mono_assembly_loaded. 2009-01-14 14:19:12 [INFO]
mod_managed.cpp:217 loadRuntime() Calling mono_domain_assembly_open.
2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime()
mono_domain_assembly_open failed. 2009-01-14 14:19:12 [CRIT]
switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading
module /usr/local/freeswitch/mod/mod_managed.so **Module load routine
returned an error** One thing I think I may be missing is a
FreeSWITCH.Managed.dll (this exists on windows environment but doesn't seem
to be compiled under linux) I thought perhaps mod_managed_lib.dll was the
linux equivalent but that exists and still no a no go. Any ideas would be
very welcome? Thank you!Regards, -Adam -- next
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-- Message: 3 Date: Thu, 15 Jan 2009
17:50:30 +1100 From: Scott Ellis scott.el...@novatex.com.au Subject:
[Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch
equivalent? To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org Message-ID:
496edcb6.4020...@novatex.com.au Content-Type: text/plain;
charset=ISO-8859-1; format=flowed Searched the wiki and mailing lists as
best I can, but with no luck. How do I get OpenZap to answer a call
immediately? (I do not need caller id) Scott
-- Message: 4 Date: Thu, 15 Jan 2009 18:16:13
+1100 From: Scott Ellis scott.el...@novatex.com.au Subject:
[Freeswitch-users] Country specific tones - how to contribute? To:
freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org Message-ID:
496ee2bd.2050...@novatex.com.au Content-Type: text/plain;
charset=ISO-8859-1; format=flowed I have tracked down a set of au tones
from the mailing list, which I am going to verify. How do I go about
getting these added into the default build so that they are available for
all in future? I tried action application=set
data=ringback=${au-ring}/ and this did not work - where does it try and
load the ring tone from? I have entries in the tones.conf file, but these
do not seem to be used. Scott --
Message: 5 Date: Thu, 15 Jan 2009 18:24:05 +1100 From: Jason White
ja...@jasonjgw.net Subject: Re: [Freeswitch-users] Country specific tones
- how to contribute? To: freeswitch-users@lists.freeswitch.org Message-ID:
20090115072405.ga15...@jdc.jasonjgw.net Content-Type: text/plain;
charset=us-ascii Scott Ellis scott.el...@novatex.com.au wrote: I have
tracked down a set of au tones from the mailing list, which I am going to
verify. How do I go about getting these added into the default build so
that they are available for all in future? Maybe by posting a patch to the
bug tracking system or the development list?I tried action
application=set data=ringback=${au-ring}/ and this did not work -
where does it try and load the ring tone from? I have entries in the
tones.conf file, but these do not seem to be used. us-ring and uk-ring are
defined in vars.xml. Note that they are global variables, referenced with
the $${variable-name} syntax. There's an ITU document referred to on the
wiki with the official definitions of ringback and other tones for various
countries. -- Message: 6 Date: Thu,
15 Jan 2009 15:43:20 +0800 From: Juan Backson juanback...@gmail.com
Subject: [Freeswitch-users] Changes in PlayAndGetDigits To:
freeswitch-users@lists.freeswitch.org Message-ID:
27c25bc40901142343l34a3e99ftecf0df971e8e3...@mail.gmail.com Content-Type:
text/plain; charset=ISO-8859-1 Hi, Is there a change in the
playAndGetDigits api? In the old release, 11102, my lua script is working
but is not working in the latest release. The error I am getting is Error
in playAndGetDigits expected 10..10 args, got 9 . Thanks, JB
-- Message: 7 Date: Thu, 15 Jan 2009 09:20:18
+0100 From: Peter P GMX prometheus...@gmx.net Subject: Re:
[Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV] To:
freeswitch-users@lists.freeswitch.org Message-ID:
496ef1c2.8020...@gmx.net Content-Type: text/plain; charset=ISO-8859-1
Hello Michael, how much $$ are we talking about? I need this issue to be
solved quickly and it's worth to spend some money. I've read the
following post:
http://www.mail-archive.com
Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2
/freeswitch-1.0.2/modules.conf without quotes obviously.
My installed mono version is
[r...@sipcore-alpha mod]# mono -V
Mono JIT compiler version 2.2 (tarball Wed Jan 14 09:44:57 PST 2009)
Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com
TLS: __thread
GC: Included Boehm (with typed GC)
SIGSEGV: altstack
Notifications: epoll
Architecture: x86
Disabled: none
I can successful compile freeswitch and it indeed compiles mod_managed.so
I added load module=mod_managed /
to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
I did also create the /usr/local/freeswitch/mod/managed directory as
stated in the wiki as requirement.
But when I start freeswitch I get the following in regards to the
mod_managed loading...
2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading
mod_managed (Common Language Infrastructure), Mono Version
2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling
mono_assembly_loaded.
2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling
mono_domain_assembly_open.
2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime()
mono_domain_assembly_open failed.
2009-01-14 14:19:12 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_managed.so
**Module load routine returned an error**
One thing I think I may be missing is a FreeSWITCH.Managed.dll (this
exists on windows environment but doesn't seem to be compiled under linux)
I thought perhaps mod_managed_lib.dll was the linux equivalent but that
exists and still no a no go.
Any ideas would be very welcome? Thank you!
Regards,
-Adam
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Message: 3
Date: Thu, 15 Jan 2009 17:50:30 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
Freeswitch equivalent?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496edcb6.4020...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Searched the wiki and mailing lists as best I can, but with no luck.
How do I get OpenZap to answer a call immediately? (I do not need caller
id)
Scott
--
Message: 4
Date: Thu, 15 Jan 2009 18:16:13 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496ee2bd.2050...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
Scott
--
Message: 5
Date: Thu, 15 Jan 2009 18:24:05 +1100
From: Jason White ja...@jasonjgw.net
Subject: Re: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
Message-ID: 20090115072405.ga15...@jdc.jasonjgw.net
Content-Type: text/plain; charset=us-ascii
Scott Ellis scott.el...@novatex.com.au wrote:
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
Maybe by posting a patch to the bug tracking system or the development
list?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
us-ring and uk-ring are defined in vars.xml. Note that they are global
variables, referenced with the $${variable-name} syntax.
There's an ITU document referred to on the wiki with the official
definitions
of ringback and other tones for various countries.
--
Message: 6
Date: Thu, 15 Jan 2009 15:43:20 +0800
From: Juan Backson juanback...@gmail.com
Subject: [Freeswitch-users] Changes in PlayAndGetDigits
To: freeswitch-users@lists.freeswitch.org
Message-ID:
27c25bc40901142343l34a3e99ftecf0df971e8e3...@mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1
Hi,
Is there a change in the playAndGetDigits api? In the old release,
11102, my lua script is working but is not working in the latest
release.
The error I am getting
Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2
)
Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com
TLS: __thread
GC: Included Boehm (with typed GC)
SIGSEGV: altstack
Notifications: epoll
Architecture: x86
Disabled: none
I can successful compile freeswitch and it indeed compiles mod_managed.so
I added load module=mod_managed /
to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
I did also create the /usr/local/freeswitch/mod/managed directory as stated
in the wiki as requirement.
But when I start freeswitch I get the following in regards to the mod_managed
loading...
2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading
mod_managed (Common Language Infrastructure), Mono Version
2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling
mono_assembly_loaded.
2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling
mono_domain_assembly_open.
2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime()
mono_domain_assembly_open failed.
2009-01-14 14:19:12 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_managed.so
**Module load routine returned an error**
One thing I think I may be missing is a FreeSWITCH.Managed.dll (this exists
on windows environment but doesn't seem to be compiled under linux)
I thought perhaps mod_managed_lib.dll was the linux equivalent but that
exists and still no a no go.
Any ideas would be very welcome? Thank you!
Regards,
-Adam
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Message: 3
Date: Thu, 15 Jan 2009 17:50:30 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
Freeswitch equivalent?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496edcb6.4020...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Searched the wiki and mailing lists as best I can, but with no luck.
How do I get OpenZap to answer a call immediately? (I do not need caller id)
Scott
--
Message: 4
Date: Thu, 15 Jan 2009 18:16:13 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496ee2bd.2050...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
Scott
--
Message: 5
Date: Thu, 15 Jan 2009 18:24:05 +1100
From: Jason White ja...@jasonjgw.net
Subject: Re: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
Message-ID: 20090115072405.ga15...@jdc.jasonjgw.net
Content-Type: text/plain; charset=us-ascii
Scott Ellis scott.el...@novatex.com.au wrote:
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
Maybe by posting a patch to the bug tracking system or the development list?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
us-ring and uk-ring are defined in vars.xml. Note that they are global
variables, referenced with the $${variable-name} syntax.
There's an ITU document referred to on the wiki with the official definitions
of ringback and other tones for various countries.
--
Message: 6
Date: Thu, 15 Jan 2009 15:43:20 +0800
From: Juan Backson juanback...@gmail.com
Subject: [Freeswitch-users] Changes in PlayAndGetDigits
To: freeswitch-users@lists.freeswitch.org
Message-ID:
27c25bc40901142343l34a3e99ftecf0df971e8e3...@mail.gmail.com
Content-Type: text/plain; charset=ISO-8859-1
Hi,
Is there a change in the playAndGetDigits api? In the old release,
11102, my lua script is working but is not working in the latest
release.
The error I am getting is Error in playAndGetDigits expected 10..10
args, got 9 .
Thanks,
JB
--
Message: 7
Date: Thu, 15 Jan 2009 09:20:18 +0100
From: Peter P GMX prometheus...@gmx.net
Subject: Re
Re: [Freeswitch-users] Freeswitch and CELT:
try http://files.freeswitch.org/freeswitch.msi On Thu, Jan 15, 2009 at 2:03 PM, Terrance Harris tharris...@gmail.comwrote: Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much! ___ 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/ 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
Re: [Freeswitch-users] mod_managed failing to load on CentOS 5.2
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Using mod_managed Linux/Mono 2.02
Has anyone had any luck using mod_managed under linux with mono yet?
The Wiki looks to still be lacking some linux installation instructions.
I feel like I'm close but missing something simple.
I got as far as adding languages/mod_managed to the
/usr/src/freeswitch-1.0.2/modules.conf without quotes obviously.
My installed mono version is
[r...@sipcore-alpha mod]# mono -V
Mono JIT compiler version 2.2 (tarball Wed Jan 14 09:44:57 PST 2009)
Copyright (C) 2002-2008 Novell, Inc and Contributors. www.mono-project.com
TLS: __thread
GC: Included Boehm (with typed GC)
SIGSEGV: altstack
Notifications: epoll
Architecture: x86
Disabled: none
I can successful compile freeswitch and it indeed compiles mod_managed.so
I added load module=mod_managed /
to my /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml
I did also create the /usr/local/freeswitch/mod/managed directory as stated
in the wiki as requirement.
But when I start freeswitch I get the following in regards to the mod_managed
loading...
2009-01-14 14:19:12 [INFO] mod_managed.cpp:309 mod_managed_load() Loading
mod_managed (Common Language Infrastructure), Mono Version
2009-01-14 14:19:12 [INFO] mod_managed.cpp:213 loadRuntime() Calling
mono_assembly_loaded.
2009-01-14 14:19:12 [INFO] mod_managed.cpp:217 loadRuntime() Calling
mono_domain_assembly_open.
2009-01-14 14:19:12 [ERR] mod_managed.cpp:220 loadRuntime()
mono_domain_assembly_open failed.
2009-01-14 14:19:12 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_managed.so
**Module load routine returned an error**
One thing I think I may be missing is a FreeSWITCH.Managed.dll (this exists
on windows environment but doesn't seem to be compiled under linux)
I thought perhaps mod_managed_lib.dll was the linux equivalent but that
exists and still no a no go.
Any ideas would be very welcome? Thank you!
Regards,
-Adam
-- next part --
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Message: 3
Date: Thu, 15 Jan 2009 17:50:30 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
Freeswitch equivalent?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496edcb6.4020...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Searched the wiki and mailing lists as best I can, but with no luck.
How do I get OpenZap to answer a call immediately? (I do not need caller id)
Scott
--
Message: 4
Date: Thu, 15 Jan 2009 18:16:13 +1100
From: Scott Ellis scott.el...@novatex.com.au
Subject: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Message-ID: 496ee2bd.2050...@novatex.com.au
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
Scott
--
Message: 5
Date: Thu, 15 Jan 2009 18:24:05 +1100
From: Jason White ja...@jasonjgw.net
Subject: Re: [Freeswitch-users] Country specific tones - how to
contribute?
To: freeswitch-users@lists.freeswitch.org
Message-ID: 20090115072405.ga15...@jdc.jasonjgw.net
Content-Type: text/plain; charset=us-ascii
Scott Ellis scott.el...@novatex.com.au wrote:
I have tracked down a set of au tones from the mailing list, which I am
going to verify. How do I go about getting these added into the default
build so that they are available for all in future?
Maybe by posting a patch to the bug tracking system or the development list?
I tried action application=set data=ringback=${au-ring}/ and this
did not work - where does it try and load the ring tone from? I have
entries in the tones.conf file, but these do not seem to be used.
us-ring and uk-ring are defined in vars.xml. Note that they are global
variables, referenced with the $${variable-name} syntax.
There's an ITU document referred to on the wiki with the official definitions
of ringback and other tones for various countries.
--
Message: 6
Date: Thu, 15 Jan 2009 15:43:20 +0800
From: Juan Backson juanback...@gmail.com
Subject: [Freeswitch-users] Changes in PlayAndGetDigits
Re: [Freeswitch-users] No Sound Heared
is it only a problem in js what if you call the bridge app in the dialplan? On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Need your help on this. I have the following Javascript statement: session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a file called gafachiDialout.js Then, i have the following extension in default.xml: extension name=6337 condition field=destination_number expression=^6337$ action application=javascript data=gafachiDialout.js / /condition /extension When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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/ 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
Re: [Freeswitch-users] No Sound Heared
Hi Anthony, The problem exists also when i call session.execute(bridge,sofia/gateway/sip.gafachi.com/number); I tend to believe that this is a firewall issue. Would you confirm? Klaus. Original-Nachricht Datum: Thu, 15 Jan 2009 17:42:46 -0600 Von: Anthony Minessale anthony.miness...@gmail.com An: freeswitch-users@lists.freeswitch.org Betreff: Re: [Freeswitch-users] No Sound Heared is it only a problem in js what if you call the bridge app in the dialplan? On Thu, Jan 15, 2009 at 3:20 PM, Klaus Teller klaus.tel...@gmx.net wrote: Hi, Need your help on this. I have the following Javascript statement: session.execute(bridge,sofia/gateway/sip.gafachi.com/someNumber) in a file called gafachiDialout.js Then, i have the following extension in default.xml: extension name=6337 condition field=destination_number expression=^6337$ action application=javascript data=gafachiDialout.js / /condition /extension When i call this extension (6337), it rings as it should. But then there is NO sound going in either direction. Any idea what i'm doing wrong here? Thanks, Klaus. -- Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a ___ 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/ 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 -- Pt! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger ___ 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 and CELT:
Hello, From what I heard celt isn't included in the most recent windows builds. I would have to build FS and celt from the source to get it enabled. On Thu, Jan 15, 2009 at 5:31 PM, Anthony Minessale anthony.miness...@gmail.com wrote: try http://files.freeswitch.org/freeswitch.msi On Thu, Jan 15, 2009 at 2:03 PM, Terrance Harris tharris...@gmail.comwrote: Hello, I have recently found out about FS and how great it is. We are trying to use FS as a voip client for radio shows. We have been using Trixbox and Skype but Skype isn't getting it done. I have heard about how great the celt codec is but I don't have enough 'skill' to compile both FS and celt in MSVC++. Is there a binary out there that would make my day or a guide? Thanks much! ___ 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/ 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 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] mod_managed failing to load on CentOS 5.2
Yeah I compile mono ... i tried both 2.0.1 and 2.2 both error on loading the mod_managed. Tim _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009___ 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] Would like to pickup a call that is on hold on another extension
I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ 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] Would like to pickup a call that is on hold on another extension
You would use a combination of storing the UUID... in the internal
db... see insert in the default dialplan... then a code to get that
out of the db... then run intercept on it using the value returned
from the db. See default config's
Store it something like this:
action application=db data=insert/last_dial_ext/$
{dialed_extension}/${uuid}/
Then use it something like this:
extension name=intercept-ext
condition field=destination_number expression=^\*\*(\d+)$
action application=answer/
action application=intercept data=${db(select/
last_dial_ext/$1)}/
action application=sleep data=2000/
/condition
/extension
/b
On Jan 15, 2009, at 7:36 PM, Scott Ellis wrote:
I would like to be able to place a call on hold on one extension, walk
to another phone and then dial a sequence (like the barge sequence)
say
55+extension number and have the call taken off hold and transferred
to
the extension I am on.
Has anyone done this? (Before I try and work it out for myself!)
Scott
___
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] Would like to pickup a call that is on hold on another extension
Wouldnt that be call parking?? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park I have been told that would be better o use mod_fifo instead... It would be nice if someone would post something on mod_fifo wiki page about how to do fancy call parking with mod_fifo (even tho it might be pretty easy). Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ 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] Would like to pickup a call that is on hold on another extension
Well, sorry. That would be better, wouldnt it? http://wiki.freeswitch.org/wiki/Mod_fifo#Park_Time_Out_Example Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ 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] Would like to pickup a call that is on hold on another extension
It is kind of - but slightly different, and simpler for the users. Scott Joo Mesquita wrote: Wouldnt that be call parking?? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_park I have been told that would be better o use mod_fifo instead... It would be nice if someone would post something on mod_fifo wiki page about how to do fancy call parking with mod_fifo (even tho it might be pretty easy). Mesquita On Jan 15, 2009, at 11:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ 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] Would like to pickup a call that is on hold on another extension
Thanks Brian, I had started looking at this, and I think I was heading in the direction you describe - now I can pursue that with a bit more confidence! So even if we do not originate the call, the last dialled extension would still be valid as it would be set up during the bridging process? (I think I need another method to collect the UUID of the leg of the bridge that initiated the call - or just the UUID that is active for that extension) Scott Brian West wrote: You would use a combination of storing the UUID... in the internal db... see insert in the default dialplan... then a code to get that out of the db... then run intercept on it using the value returned from the db. See default config's Store it something like this: action application="db" data=""/ Then use it something like this: extension name="intercept-ext" condition field="destination_number" _expression_="^\*\*(\d+)$" action application="answer"/ action application="intercept" data=""/ action application="sleep" data=""/ /condition /extension /b On Jan 15, 2009, at 7:36 PM, Scott Ellis wrote: I would like to be able to place a call on hold on one extension, walk to another phone and then dial a sequence (like the barge sequence) say 55+extension number and have the call taken off hold and transferred to the extension I am on. Has anyone done this? (Before I try and work it out for myself!) Scott ___ 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] Would like to pickup a call that is on hold on another extension
The key is the uuid.. In FreeSWITCH the uuid is the only bit you really need to know to do anything with the session. /b On Jan 15, 2009, at 9:12 PM, Scott Ellis wrote: Thanks Brian, I had started looking at this, and I think I was heading in the direction you describe - now I can pursue that with a bit more confidence! So even if we do not originate the call, the last dialled extension would still be valid as it would be set up during the bridging process? (I think I need another method to collect the UUID of the leg of the bridge that initiated the call - or just the UUID that is active for that extension) Scott ___ 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] Would like to pickup a call that is on hold on another extension
So for this scenario, I think I need to store the UUID of both sides before every bridge that I do, that way it will always reflect the most recently connected call to an extension - either as source or destination. I found the log action, so now I can spit out debug information as I work this out! Scott p.s. Thanks for all your help, FreeSwitch (and the community) rock! Brian West wrote: The key is the uuid.. In FreeSWITCH the uuid is the only bit you really need to know to do anything with the session. /b On Jan 15, 2009, at 9:12 PM, Scott Ellis wrote: Thanks Brian, I had started looking at this, and I think I was heading in the direction you describe - now I can pursue that with a bit more confidence! So even if we do not originate the call, the last dialled extension would still be valid as it would be set up during the bridging process? (I think I need another method to collect the UUID of the leg of the bridge that initiated the call - or just the UUID that is active for that extension) Scott ___ 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] Starting FS on OSX
On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote: Your build issue is with your autotools install, I have seen issues if you have ever installed any of the autotools from macports or fink. I have never used Fink or Macports so that isn't it. In fact the supposed statements made to the effect that FS will build from SVN fine on 10.4 with the latest available apple dev tools is quite wrong in my experience. I setup a virgin 10.4 and updated everything and had many complaints from FS about tool versions. If you want to build from svn you can run bootstrap on another box (a linux box perhaps) and then tar up that dir and move it to your mac. Huh, interesting. We pre-bootstrap the release tarballs which is why that is building fine for you. Right, Thanks for all your efforts and an outstanding platform! Marty ___ 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] FS doesn't maitain PRI- D-channel state right
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, I found, that FS doesn't maintain D-Channel's state correctly. I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up I get this on console: 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread starting. 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008 (sapi: 0, tei: 0, mode: PTP TE) 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame - - Q.921 Packet [Outgoing] --- SAPI: 0, TEI: 0, C/R: Command (0) Type: U Frame (SABME) P/F: 1 Q.921 state: TEI Assigned (4) [flags: ] - -- 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'openzap' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239 switch_loadable_module_process() Adding Application 'disable_ec' 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'oz' 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes) - - Q.921 Packet [Incoming] --- SAPI: 0, TEI: 0, C/R: Response (0) Type: U Frame (DM (Disconnected Mode)) P/F: 1 Q.921 state: TEI Assigned (4) [flags: ] - -- So TDM side tells FS that the link is down: but when I check d-channel's state FS tells me that it is UP: freeswi...@ippbx-prod-node0 oz dump 1 16 API CALL [oz(dump 1 16)] output: span_id: 1 chan_id: 16 physical_span_id: 1 physical_chan_id: 16 type: DQ921 state: UP last_state: DOWN cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE When I do an outbound call FS throws no error or warning that the link isn't up and gives the call a try. regards helmut -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk =ycwj -END PGP SIGNATURE- ___ 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
