Re: [Freeswitch-users] api_hangup_hook not actually executing the command
Thank you. Scott Brian West wrote: I would use mod_limit and not futz with global anything. /b On Jan 22, 2009, at 2:07 PM, Mathieu Rene wrote: Its global_setvar not set_global Math ___ 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] Hang up not received
I had a similar problem, you can use
action application="set" data=""/ (I added
an "au" ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the
system generated ring tone, which was confusing some of our (not to
bright) callers.
As we don't use callerID I turned that flag off in the openzap.conf.xml
file - I thought that this would do what I wanted (answer the instant
the call is detected), but the change in the config file does not make
it all the way down to the point where it takes action. At this point I
hacked the code to get what I wanted. I have to create a JIRA entry
with the details yet.
As far as I understand, this is the right place for OpenZap, as it is a
product of the FS project.
Scott
Toms wrote:
Scott, I imagined that it could be an OpenZap problem, but
I didn't find an OpenZap mailing list, so I sent the email to FS list.
Do you know where can I find more information about OpenZap hardware
support and developement status (I have special interest in Loop
Start)??
Anthony and Ognjen, I've tried tone detection and thanks to that FS is
detecting hung up, but I faced the problem that tone detector answer
the call...
That's my dialplan:
extension name="extension_name"
condition field="destination_number"
_expression_="^91999$"
action application="tone_detect" data=""/
action application="bridge"
data=""/
/condition
/extension
When I receive a call from PSTN, tone detection answer the call (the
caller hears only one first tone and then hears "nothing" until I pick
up the call on softphone).
So, I think that tone detection solution does not resolve my problem...
Is there any other possibility to detect hang up without answering the
call (using Loop Start signaling) or have we to wait until OpenZap is
completely developed?
Thanks in advance.
On Tue, Jan 20, 2009 at 10:43 PM, Ognjen
Seslija osesl...@gmail.com
wrote:
Ok, as discussed with Tony on IRC channelI followed his
directions which lead to a successfull outcome (like it always does I
might add :).
One has to use tone_detect app in FreeSWITCH dialplan in order
to check for busy tones coming from the PSTN side and ifmatched fire a
hangup application. This is the snippet of my test dp that does the
trick (from extension Local_extensions in default.xml):
anti-action application="tone_detect" data=""/
anti-action application="bridge" data=""true"
href="mailto:user/$%7bdialed_extension...@$%7bdomain_name%7d%22/"
target="_blank">user/${dialed_extensi...@${domain_name}"/
This means that FS will listen to freq of 425 Hz and wait for
4 positive detection to fire up hangup app with code 16 which is
NORMAL_CLEARING (425 Hz is the freq telco here uses; for other
countries I suggest getting the ITU world tones pdf file and check
there):
2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 1/4
2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 2/4
2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 3/4
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262
tone_detect_callback() TONE busy HIT 4/4
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268
tone_detect_callback() TONE busy DETECTED
2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584
hangup_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]
Regards,
Ognjen
On Tue, Jan 20, 2009 at 8:45 PM, Ognjen
Seslija osesl...@gmail.com
wrote:
I tried similar setup with my analog card (X100P) and I'm
having same issue. Call is not hungup on the oz side once the caller
ends. My telco doesn't use polarity reversal for singalling hang up so
I'm guess I'm stuck to detecting busy tone from the telco side. I'll
try to modify tones.conf accordingly.
Regards,
Ognjen
(sekil)
On Tue, Jan 20, 2009 at 6:05 PM, Anthony
Minessale anthony.miness...@gmail.com
wrote:
This
is a common issue with analog phones even traditional answering
machines suffer from it.
I'm sure you must have had an answering machine at some point that has
dial tone as the message it receives.
Unless FreeSWITCH has some hint that the call has hungup it will not
stop trying to complete the call.
If the other side is sending a busy tone to indicate hangup it's
possible to use the tone_detect app to pick
up on the tones and abort the call.
Another thing you could do if you have unlimited inbound is explicitly
answer the call in the dialplan before
you call your sip phones this will give you a more profound hangup
detection but it will make every call count
even when nobody answers.
On Tue, Jan 20, 2009 at 10:46 AM, Toms tomasborre...@gmail.com wrote:
Hi all,
Re: [Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent?
Yes and yes, I will get the full details to you next week, snowed under
a bit at the moment.
Scott
Anthony Minessale wrote:
did you answer the call in your dialplan?
do you have a full debug log of a call with that parameter enabled on
the analog span in question?
On Thu, Jan 15, 2009 at 4:17 AM, Scott Ellis
scott.el...@novatex.com.au
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 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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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
GTAL
[Freeswitch-users] Can I restrict a gateway to 1 call at a time?
I need to restrict the outbound calls to an SPA3000. Is there a way of marking a gateway to only allow one call at a time? Or, as the SPA3000 registers with FS, allowing the extension to only allow one outbound call. (Problem here is I have not worked out how to send a call to the FXS using the extension information, I have just been sending it as sofia/internal/$...@10.0.0.17:5061 ) If I do this action application="bridge" data=""/ I get a dial tone, but I can't figure how to send the number in this case. Obviously my understanding of dial plans is less than what it could be! This is all necessary, as if you send a second outbound call to the SPA3000, it will then think that there is an inbound call 1min 30 seconds later. 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] Hang up not received
A lot of the hardware has problems detecting hangups, and the OpenZap drivers are not totally sorted yet. So I would be looking at hardware/openzap issue, not FS. Scott Toms wrote: Hi all, I'm configuring my home PBX using FreeSwitch. I'm using a X101P card configured as FXO (conected to analog PSTN line) and I have several IP phones and softphones conected to FreeSwitch. I can call from an IP phone to other IP phone (the same with the softphones) and also from an IP phone (or softphone) to an external number thought PSTN. When I call from an external analog phone to FreeSwitch, I bridge the call to all internal IP phones and softphones and they ring, but the problem is that when I hang up the call in the external phone, all internal phones (IP phones and softphones) keeps ringing... I'm pretty sure the problem is that FreeSwitch don't receive the hang up, because I cann't see anything on the log. I've also created my own tones.conf for my country (Spain) but I'm not sure if it's ok (but I have the same problem with hang up) I've googled the list, and I've found several people with a similar problem but no solution... That's my pastebin with the most importants printouts and config files: http://pastebin.freeswitch.org/6822 Thank you very much in advance. ___ 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] api_hangup_hook not actually executing the command
The on answer is fine, the api_hangup_hook gets to the point where it wants to execute, but then nothing happens. Any thoughts? 2009-01-21 03:08:00 [DEBUG] switch_channel.c:1773 switch_channel_perform_mark_answered() sofia/internal/lin...@10.0.0.9 execute on answer: set_global(10.0.0.19_INCALL=true)2009-01-21 03:08:00 [DEBUG] mod_dptools.c:726 set_global_function() SET GLOBAL [10.0.0.19_INCALL]=[true] 2009-01-21 03:08:09 [DEBUG] switch_core_state_machine.c:416 switch_core_session_run() Hangup Command set_global(10.0.0.19_INCALL=false): It does not execute - the global variable is not set...quite odd. 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] api_hangup_hook not actually executing the command
coming Brian West wrote: I have just tested this: extension name="api_hangup_hook_test" condition field="destination_number" _expression_="^877$" action application="answer"/ action application="set" data=""moz-txt-link-abbreviated" href="mailto:user/1...@bkw.org">user/1...@bkw.org "/ action application="playback" data=""/ /condition /extension When I hangup I see: 2009-01-20 11:22:41 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() Hangup Command originate(user/1...@bkw.org ): Works So yes we need to see the full log. /b On Jan 20, 2009, at 11:20 AM, Michael Collins wrote: Can you pastebin the entire call from start to finish? Also, pastebin your dialplan extension. -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
Re: [Freeswitch-users] OpenZap detect tones question
No, I need to look through the code to work it out. Will get back to you when I have. Scott Michael Collins wrote: Scott, Did you get the answer to this question yet? Either way, could you post your openzap.conf and tones.conf files? I'd like to get this one documented on the wiki so that non-US PSTN tones are properly documented. -MC On Sat, Jan 17, 2009 at 1:51 AM, Scott Ellis scott.el...@novatex.com.au wrote: Quick question, when specifying a "detect-busy" tone in the tones.conf file - is the cadence used? (The US examples to not have cadence) In the tests I have does it does not seem to be. This is a problem in Australia, as we have managed to have our busy tone 425Hz, 375ms on 375ms off, also in our dial tone 400+425+450. So when I go to dial a call, I often get the Zap channel hanging up again as it thinks the line is busy. 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
[Freeswitch-users] Interesting Problem with SPA 3000
I have an interesting problem with a set up using two (or more) SPA 3000's. I make a call out on the unit to a PSTN line - great. I then try and make another call out on that line - it fails and moves on to the next one in the bridge call statement. It then goes through on the second unit. Almost always 1:30 later, I get an inbound call from the PSTN showing up in FreeSwitch, which goes to an extension - and when answered bridges that extension onto the existing call. Now I am sure that this behaviour from the SPA is a little odd, but does anyone have any tips for dealing with it from the dialplan? Most obvious being a way to not call the first unit when it already is active... I do not have it defined as a gateway - just using sofia/internal/$...@10.0.0.18:5061 so make a call. 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] Interesting Problem with SPA 3000
No the hang ups seem to be ok. It seems to be that after FS tries to make a second call, 1:30 later I get the phantom inbound call. If I do not try and make that second call everything works fine. I think I can make a dial plan to manage the outbound calls, but the same thing happens if the first call is inbound PSTN, then we make outbound. So I need to have a db variable to indicate "device in use" effectively so the call attempt is not made. Scott The second outbound being made 2009-01-20 06:07:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/43517...@10.0.0.18:5061 [5b9d6822-e65c-11dd-b2e1-993799172013] 2009-01-20 06:07:03 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/43517...@10.0.0.18:5061 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8 (sofia/internal/43517...@10.0.0.18:5061) Ended 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/43517...@10.0.0.18:5061 [CS_HANGUP] 2009-01-20 06:07:04 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/43517...@10.0.0.17:5061 [5babe9ba-e65c-11dd-b2e1-993799172013] The phantom inbound, goes to 500 is not answered, and then the original call above is closed. 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/lin...@10.0.0.9] has been answered 2009-01-20 06:07:42 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/500 [726fc392-e65c-11dd-b2e1-993799172013] 2009-01-20 06:07:42 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/500 [CS_CONSUME_MEDIA] [USER_BUSY] 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 11 (sofia/internal/500) Ended 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/500 [CS_HANGUP] 2009-01-20 06:07:42 [INFO] mod_dptools.c:1984 audio_bridge_function() Originate Failed. Cause: USER_BUSY 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:2011 audio_bridge_function() Hangup sofia/internal/lin...@10.0.0.9 [CS_EXECUTE] [USER_BUSY] 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 10 (sofia/internal/lin...@10.0.0.9) Ended 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/lin...@10.0.0.9 [CS_HANGUP] 2009-01-20 06:11:02 [NOTICE] mod_sofia.c:681 sofia_read_frame() Hangup sofia/internal/49633...@10.0.0.18:5061 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2009-01-20 06:11:02 [NOTICE] switch_ivr_bridge.c:955 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/4...@10.0.0.9 [CS_EXECUTE] [MEDIA_TIMEOUT] 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 5 (sofia/internal/4...@10.0.0.9) Ended 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/4...@10.0.0.9 [CS_HANGUP] 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 6 (sofia/internal/49633...@10.0.0.18:5061) Ended 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/49633...@10.0.0.18:5061 [CS_HANGUP] Michael Collins wrote: Scott, Is it possible that your SPA is not properly detecting hangups? I'm wondering if there is a setting in the SPA3000 for your country. (Australia, no?) Please check your SPA settings and report back. Thanks, MC On Mon, Jan 19, 2009 at 10:27 AM, Scott Ellis scott.el...@novatex.com.au wrote: I have an interesting problem with a set up using two (or more) SPA 3000's. I make a call out on the unit to a PSTN line - great. I then try and make another call out on that line - it fails and moves on to the next one in the bridge call statement. It then goes through on the second unit. Almost always 1:30 later, I get an inbound call from the PSTN showing up in FreeSwitch, which goes to an extension - and when answered bridges that extension onto the existing call. Now I am sure that this behaviour from the SPA is a little odd, but does anyone have any tips for dealing with it from the dialplan? Most obvious being a way to not call the first unit when it already is active... I do not have it defined as a gateway - just using sofia/internal/$...@10.0.0.18:5061 so make a call. 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/freesw
[Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102)
If so, could you please share your set up? directory files, and dial plan details (gateway details if configured this way)? 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] Interesting Problem with SPA 3000
Will do. Scott Michael Collins wrote: Scott, I think you're at the point where you'll need to collect more info and start a pastebin or two. I'm actually working on the checklist and how-to for reporting bugs and troubleshooting. You can be my first test case!:) Please visit here and start collecting data: http://wiki.freeswitch.org/wiki/Reporting_Bugs Let me know if any of my instructions seem wrong or confusing or just plain don't work. Thanks, MC (mercutioviz) On Mon, Jan 19, 2009 at 11:21 AM, Scott Ellis scott.el...@novatex.com.au wrote: No the hang ups seem to be ok. It seems to be that after FS tries to make a second call, 1:30 later I get the phantom inbound call. If I do not try and make that second call everything works fine. I think I can make a dial plan to manage the outbound calls, but the same thing happens if the first call is inbound PSTN, then we make outbound. So I need to have a db variable to indicate "device in use" effectively so the call attempt is not made. Scott The second outbound being made 2009-01-20 06:07:03 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/43517...@10.0.0.18:5061 [5b9d6822-e65c-11dd-b2e1-993799172013] 2009-01-20 06:07:03 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/43517...@10.0.0.18:5061 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8 (sofia/internal/43517...@10.0.0.18:5061) Ended 2009-01-20 06:07:04 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/43517...@10.0.0.18:5061 [CS_HANGUP] 2009-01-20 06:07:04 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/43517...@10.0.0.17:5061 [5babe9ba-e65c-11dd-b2e1-993799172013] The phantom inbound, goes to 500 is not answered, and then the original call above is closed. 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/lin...@10.0.0.9] has been answered 2009-01-20 06:07:42 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/500 [726fc392-e65c-11dd-b2e1-993799172013] 2009-01-20 06:07:42 [NOTICE] sofia.c:3124 sofia_handle_sip_i_state() Hangup sofia/internal/500 [CS_CONSUME_MEDIA] [USER_BUSY] 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 11 (sofia/internal/500) Ended 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/500 [CS_HANGUP] 2009-01-20 06:07:42 [INFO] mod_dptools.c:1984 audio_bridge_function() Originate Failed. Cause: USER_BUSY 2009-01-20 06:07:42 [NOTICE] mod_dptools.c:2011 audio_bridge_function() Hangup sofia/internal/lin...@10.0.0.9 [CS_EXECUTE] [USER_BUSY] 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 10 (sofia/internal/lin...@10.0.0.9) Ended 2009-01-20 06:07:42 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/lin...@10.0.0.9 [CS_HANGUP] 2009-01-20 06:11:02 [NOTICE] mod_sofia.c:681 sofia_read_frame() Hangup sofia/internal/49633...@10.0.0.18:5061 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT] 2009-01-20 06:11:02 [NOTICE] switch_ivr_bridge.c:955 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/4...@10.0.0.9 [CS_EXECUTE] [MEDIA_TIMEOUT] 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 5 (sofia/internal/4...@10.0.0.9) Ended 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/4...@10.0.0.9 [CS_HANGUP] 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 6 (sofia/internal/49633...@10.0.0.18:5061) Ended 2009-01-20 06:11:02 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/49633...@10.0.0.18:5061 [CS_HANGUP] Michael Collins wrote: Scott, Is it possible that your SPA is not properly detecting hangups? I'm wondering if there is a setting in the SPA3000 for your country. (Australia, no?) Please check your SPA settings and report back. Thanks, MC On Mon, Jan 19, 2009 at 10:27 AM, Scott Ellis scott.el...@novatex.com.au wrote: I have an interesting problem with a set up using two (or more) SPA 3000's. I make a call out on the unit to a PSTN line - great. I then try and make another call out on that line - it fails and moves on to the next one in the bridge call statement. It then goes through on the second unit. Almost always 1:30 later, I get an inbound call from the PSTN showing up in FreeSwitch, which goes to an extension - and when answered bridges that extension onto the existing call. Now I am sure that this behaviour from the SPA is a little odd, but does anyone have any tips for dealing with it from the dialplan? Most obvious being a way to not call the first unit when it already is active... I d
[Freeswitch-users] OpenZap detect tones question
Quick question, when specifying a detect-busy tone in the tones.conf file - is the cadence used? (The US examples to not have cadence) In the tests I have does it does not seem to be. This is a problem in Australia, as we have managed to have our busy tone 425Hz, 375ms on 375ms off, also in our dial tone 400+425+450. So when I go to dial a call, I often get the Zap channel hanging up again as it thinks the line is busy. 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] 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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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] 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] 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
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
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
[Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call
I have an inbound call via OpenZap, when I attempt to bridge to a SIP
extension, I get the ring tone (inbound line) up until the bridge fails
(for timeout or do not disturb). At this point the call is answered and
then my dial plan moves on to attempt another bridge to different
extensions. So I no longer have the ring tone for the person dialing in.
The call can still be answered and everything works ok, but I would
rather not answer the call until someone actually picks up. Failing that
simulating a ring tone would be good enough.
Have searched around, but at a bit of a loss on how to dothis.
Any suggestions greatly appreciated.
Scott
From my dialplan
extension name=LandLine IN
condition field=source expression=mod_openzap/
condition field=caller_id_number expression=^[1-8]$
!-- Ring reception for 30 seconds --
!--action application=set data=call_timeout=30/ --
action application=set data=continue_on_fail=true/
!--action application=set data=hangup_after_bridge=true/--
action application=bridge
data={leg_timeout=30}sofia/$${domain}/500/
!--action application=playback
data=sounds/ReceptionBusy.wav/ --
!-- Ring second group for 15 seconds --
action application=set data=call_timeout=15/
action application=set data=continue_on_fail=true/
action application=set data=hangup_after_bridge=true/
action application=ring_ready/
action application=bridge
data=${group_call(ringgro...@${domain_name})/
!-- Ring everybody --
action application=set data=call_timeout=15/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=${group_call(every...@${domain_name})/
action application=hangup data=NO_ANSWER/
/condition
/extension
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Re: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call (partial call log added)
I have an inbound call via OpenZap, when I attempt to bridge to a SIP
extension, I get the ring tone (inbound line) up until the bridge fails
(for timeout or do not disturb). At this point the call is answered and
then my dial plan moves on to attempt another bridge to different
extensions. So I no longer have the ring tone for the person dialing in.
The call can still be answered and everything works ok, but I would
rather not answer the call until someone actually picks up. Failing that
simulating a ring tone would be good enough.
Have searched around, but at a bit of a loss on how to dothis.
Any suggestions greatly appreciated.
Scott
From my dialplan
extension name=LandLine IN
condition field=source expression=mod_openzap/
condition field=caller_id_number expression=^[1-8]$
!-- Ring reception for 30 seconds --
!--action application=set data=call_timeout=30/ --
action application=set data=continue_on_fail=true/
!--action application=set data=hangup_after_bridge=true/--
action application=bridge
data={leg_timeout=30}sofia/$${domain}/500/
!--action application=playback
data=sounds/ReceptionBusy.wav/ --
!-- Ring second group for 15 seconds --
action application=set data=call_timeout=15/
action application=set data=continue_on_fail=true/
action application=set data=hangup_after_bridge=true/
action application=ring_ready/
action application=bridge
data=${group_call(ringgro...@${domain_name})/
!-- Ring everybody --
action application=set data=call_timeout=15/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=${group_call(every...@${domain_name})/
action application=hangup data=NO_ANSWER/
/condition
/extension
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A call log
009-01-14 22:47:10 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][2:1] STATE [IDLE]
2009-01-14 22:47:12 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][2:1] STATE [IDLE]
2009-01-14 22:47:13 [DEBUG] ozmod_analog.c:744 process_event() EVENT
[RING_START][2:1] STATE [IDLE]
2009-01-14 22:47:13 [NOTICE] switch_ivr_originate.c:206
check_per_channel_timeouts() Hangup sofia/internal/500
[CS_CONSUME_MEDIA] [ALLOTTED_TIMEOUT]
2009-01-14 22:47:13 [DEBUG] switch_channel.c:1517
switch_channel_perform_hangup() Send signal sofia/internal/500 [KILL]
2009-01-14 22:47:13 [DEBUG] switch_core_session.c:810
switch_core_session_signal_state_change() Send signal sofia/internal/500
[BREAK]
2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:466
switch_core_session_run() (sofia/internal/500) State CONSUME_MEDIA going
to sleep
2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:379
switch_core_session_run() (sofia/internal/500) Running State Change
CS_HANGUP
2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:410
switch_core_session_run() (sofia/internal/500) State HANGUP
2009-01-14 22:47:13 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel
sofia/internal/500 hanging up, cause: ALLOTTED_TIMEOUT
2009-01-14 22:47:13 [DEBUG] mod_sofia.c:351 sofia_on_hangup() Sending
CANCEL to sofia/internal/500
2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:46
switch_core_standard_on_hangup() sofia/internal/500 Standard HANGUP,
cause: ALLOTTED_TIMEOUT
2009-01-14 22:47:13 [DEBUG] switch_core_state_machine.c:410
switch_core_session_run() (sofia/internal/500) State HANGUP going to sleep
2009-01-14 22:47:13 [DEBUG] switch_core_session.c:942
switch_core_session_thread() Session 403 (sofia/internal/500) Locked,
Waiting on external entities
2009-01-14 22:47:13 [DEBUG] switch_ivr_originate.c:1705
switch_ivr_originate() Originate Resulted in Error Cause: 602
[ALLOTTED_TIMEOUT]
2009-01-14 22:47:13 [NOTICE] switch_core_session.c:960
switch_core_session_thread() Session 403 (sofia/internal/500) Ended
2009-01-14 22:47:13 [NOTICE] switch_core_session.c:962
switch_core_session_thread() Close Channel sofia/internal/500 [CS_HANGUP]
2009-01-14 22:47:13 [DEBUG] switch_ivr.c:59 switch_ivr_sleep()
OpenZAP/2:1/2 receive message [PROGRESS]
2009-01-14 22:47:13 [DEBUG] mod_openzap.c:785
channel_receive_message_fxo() Changing state on 2:1 from IDLE to UP
2009-01-14 22:47:13 [DEBUG] switch_core_session.c:513
switch_core_session_perform_receive_message() Send signal OpenZAP/2:1/2
[BREAK]
2009-01-14 22:47:13 [NOTICE] switch_ivr.c:59 switch_ivr_sleep()
Ring-Ready OpenZAP/2:1/2!
2009-01-14 22:47:13 [NOTICE] switch_ivr.c:59 switch_ivr_sleep()
Pre-Answer OpenZAP/2:1/2!
2009-01-14 22:47:13 [DEBUG] switch_channel.c:177
switch_channel_audio_sync() OpenZAP/2:1/2 receive message [AUDIO_SYNC]
Re: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call
Thanks Anthony, got to say I am hugely impressed with the software - I am another Asterisk refugee :-) So the answering of the call even though the bridge fails is correct operation for the system? (Just curious) Scott Anthony Minessale wrote: Have a look here: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones On Wed, Jan 14, 2009 at 4:03 AM, Scott Ellis scott.el...@novatex.com.au wrote: I have an inbound call via OpenZap, when I attempt to bridge to a SIP extension, I get the ring tone (inbound line) up until the bridge fails (for timeout or do not disturb). At this point the call is answered and then my dial plan moves on to attempt another bridge to different extensions. So I no longer have the ring tone for the person dialing in. The call can still be answered and everything works ok, but I would rather not answer the call until someone actually picks up. Failing that simulating a ring tone would be good enough. Have searched around, but at a bit of a loss on how to dothis. Any suggestions greatly appreciated. Scott From my dialplan extension name="LandLine IN" condition field="source" _expression_="mod_openzap"/ condition field="caller_id_number" _expression_="^[1-8]$" !-- Ring reception for 30 seconds -- !--action application="set" data=""/ -- action application="set" data=""/ !--action application="set" data=""/-- action application="bridge" data=""/ !--action application="playback" data=""/ -- !-- Ring second group for 15 seconds -- action application="set" data=""/ action application="set" data=""/ action application="set" data=""/ action application="ring_ready"/ action application="bridge" data=""/ !-- Ring everybody -- action application="set" data=""/ action application="set" data=""/ action application="bridge" data=""/ action application="hangup" data=""/ /condition /extension ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent?
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 ___ 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] Country specific tones - how to contribute?
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] Pennytel Gateway Registration problem
c:265 sofia_reg_check_gateway() Registering sip.pennytel.com nua: nua_handle_bind: entering nua: nua_register: entering nua(0x9413228): sent signal r_register nua(0x9413228): recv signal r_register nua: nua_stack_set_params: entering soa_clone(static::0x93f2a10, 0x93f34f0, 0x9413228) called soa_set_params(static::0x9413318, ...) called soa_set_params(static::0x9413318, ...) called nta_leg_tcreate(0x9416e30) nua(0x9413228): adding register usage nta: selecting scheme sip nta: for "sip.pennytel.com" query "_sip._udp.sip.pennytel.com" SRV nta: for "sip.pennytel.com" query "sip.pennytel.com" A (cached) nta: sip.pennytel.com. IN A 202.85.243.87 tport_tsend(0x93f0088) tpn = udp/202.85.243.87:5060 tport_resolve addrinfo = 202.85.243.87:5060 tport_by_addrinfo(0x93f0088): not found by name udp/202.85.243.87:5060 tport_vsend(0x93f0088): 646 bytes of 646 to udp/202.85.243.87:5060 tport_vsend returned 646 send 646 bytes to udp/[202.85.243.87]:5060 at 17:02:02.171924: REGISTER sip:sip.pennytel.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 203.113.255.140:5080;rport;branch=z9hG4bKKD27r9g81N1DF Max-Forwards: 70 From: sip:8xxx...@sip.pennytel.com;transport=udp;tag=KccgNKcX2yy4m To: sip:8xxx...@sip.pennytel.com;transport=udp Call-ID: 4b7a9924-cd25-11dd-976a-1b8d30580e2c CSeq: 108691493 REGISTER Contact: sip:8xxx...@203.113.255.140:5080;transport=udp Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10760 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 nta: sent REGISTER (108691493) to udp/202.85.243.87:5060 tport_pend(0x93f0088): pending 0x9415108 for udp/192.168.0.5:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms tport_wakeup_pri(0x93f0088): events IN tport_recv_event(0x93f0088) tport_recv_iovec(0x93f0088) msg 0x94388a8 from (udp/192.168.0.5:5080) has 518 bytes, veclen = 1 recv 518 bytes from udp/[202.85.243.87]:5060 at 17:02:02.218845: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 203.113.255.140:5080;rport=5080;branch=z9hG4bKKD27r9g81N1DF From: sip:8xxx...@sip.pennytel.com;transport=udp;tag=KccgNKcX2yy4m To: sip:8xxx...@sip.pennytel.com;transport=udp;tag=abda4710fbd488d9ce6d01bba5c3e23b-ed81 Call-ID: 4b7a9924-cd25-11dd-976a-1b8d30580e2c CSeq: 108691493 REGISTER WWW-Authenticate: Digest realm="sip.pennytel.com", nonce="494ad54ae2c4bbea62dadcd6a8b620332dcbd7d2" Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 tport_deliver(0x93f0088): msg 0x94388a8 (518 bytes) from udp/202.85.243.87:5080/sip next=(nil) nta: received 401 Unauthorized for REGISTER (108691493) nta: 401 Unauthorized is going to a transaction nta_outgoing: RTT is 48.883 ms tport_release(0x93f0088): 0x9415108 by 0x9439188 with 0x94388a8 nta: timer set next to 4530 ms nta: timer K fired, terminate REGISTER (108691493) outgoing_reclaim_all((nil), (nil), 0xb2d2b1e8) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free nta: timer not set 2008-12-19 04:02:31 [WARNING] sofia_reg.c:307 sofia_reg_check_gateway() sip.pennytel.com Failed Registration, setting retry to 30 seconds. Anthony Minessale wrote: can you press f8 to set the FS console to DEBUG and take the same capture. On Wed, Dec 17, 2008 at 8:45 PM, Scott Ellis scott.el...@novatex.com.au wrote: After further checking, it does not seem like the authentication after the challenge is being sent... Are there any other settings I should be aware of other than placing the file in external and setting register to true? Scott 2008-12-18 13:32:28 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway() Registering sip.pennytel.com nua: nua_handle_bind: entering nua: nua_register: entering nua(0x89b08e0): sent signal r_register nua(0x89b08e0): recv signal r_register nua: nua_stack_set_params: entering soa_clone(static::0x8977798, 0x89792f8, 0x89b08e0) called soa_set_params(static::0x89c52c0, ...) called soa_set_params(static::0x89c52c0, ...) called nta_leg_tcreate(0x89c4948) nua(0x89b08e0): adding register usage nta: selecting scheme sip nta: for "sip.pennytel.com" query "_sip._udp.sip.pennytel.com" SRV nta: for "sip.pennytel.com" query "sip.pennytel.com" A (cached) nta: sip.pennytel.com. IN A 202.85.243.87 tport_tsend(0x8976740) tpn = udp/202.85.243.87:5060 tport_resolve addrinfo = 202.85.243.87:5060 tport_by_addrinfo(0x8976740): not found by name udp/202.85.243.87:5060 tport_vsend(0x8976740): 646 bytes of 646 to udp/202.85.243.87:5060 tport_vsend returned 646 send 646 bytes to udp/[20
Re: [Freeswitch-users] Pennytel Gateway Registration problem
After further checking, it does not seem like the authentication after the challenge is being sent... Are there any other settings I should be aware of other than placing the file in external and setting register to true? Scott 2008-12-18 13:32:28 [NOTICE] sofia_reg.c:265 sofia_reg_check_gateway() Registering sip.pennytel.com nua: nua_handle_bind: entering nua: nua_register: entering nua(0x89b08e0): sent signal r_register nua(0x89b08e0): recv signal r_register nua: nua_stack_set_params: entering soa_clone(static::0x8977798, 0x89792f8, 0x89b08e0) called soa_set_params(static::0x89c52c0, ...) called soa_set_params(static::0x89c52c0, ...) called nta_leg_tcreate(0x89c4948) nua(0x89b08e0): adding register usage nta: selecting scheme sip nta: for "sip.pennytel.com" query "_sip._udp.sip.pennytel.com" SRV nta: for "sip.pennytel.com" query "sip.pennytel.com" A (cached) nta: sip.pennytel.com. IN A 202.85.243.87 tport_tsend(0x8976740) tpn = udp/202.85.243.87:5060 tport_resolve addrinfo = 202.85.243.87:5060 tport_by_addrinfo(0x8976740): not found by name udp/202.85.243.87:5060 tport_vsend(0x8976740): 646 bytes of 646 to udp/202.85.243.87:5060 tport_vsend returned 646 send 646 bytes to udp/[202.85.243.87]:5060 at 02:32:30.322198: REGISTER sip:sip.pennytel.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 203.113.255.140:5080;rport;branch=z9hG4bKt232eUFUNXr2e Max-Forwards: 70 From: sip:8...@sip.pennytel.com;transport=udp;tag=t0Umc83St29ND To: sip:8x...@sip.pennytel.com;transport=udp Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d CSeq: 108665407 REGISTER Contact: sip:8xx...@203.113.255.140:5080;transport=udp Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10760 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Content-Length: 0 nta: sent REGISTER (108665407) to udp/202.85.243.87:5060 tport_pend(0x8976740): pending 0x89f2d50 for udp/192.168.0.5:5080 (already 0) nta: timer set to 32000 ms nta: timer shortened to 500 ms tport_wakeup_pri(0x8976740): events IN tport_recv_event(0x8976740) tport_recv_iovec(0x8976740) msg 0x89eeeb8 from (udp/192.168.0.5:5080) has 518 bytes, veclen = 1 recv 518 bytes from udp/[202.85.243.87]:5060 at 02:32:30.370072: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 203.113.255.140:5080;rport=5080;branch=z9hG4bKt232eUFUNXr2e From: sip:8...@sip.pennytel.com;transport=udp;tag=t0Umc83St29ND To: sip:8...@sip.pennytel.com;transport=udp;tag=abda4710fbd488d9ce6d01bba5c3e23b-cec7 Call-ID: d25d6f36-ccab-11dd-900f-67e92a02be7d CSeq: 108665407 REGISTER WWW-Authenticate: Digest realm="sip.pennytel.com", nonce="4949b76bf622961d78acb213b5556104938ecd6e" Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 tport_deliver(0x8976740): msg 0x89eeeb8 (518 bytes) from udp/202.85.243.87:5080/sip next=(nil) nta: received 401 Unauthorized for REGISTER (108665407) nta: 401 Unauthorized is going to a transaction nta_outgoing: RTT is 49.89 ms tport_release(0x8976740): 0x89f2d50 by 0x89a4640 with 0x89eeeb8 nta: timer set next to 4531 ms nta: timer K fired, terminate REGISTER (108665407) outgoing_reclaim_all((nil), (nil), 0xb2c6d1e8) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free nta: timer not set 2008-12-18 13:32:59 [WARNING] sofia_reg.c:307 sofia_reg_check_gateway() sip.pennytel.com Failed Registration, setting retry to 60 seconds. Michael Jerris wrote: We send authentication after we get a challenge because on startup we need the nonce from them to build the hash in the Auth header properly. Mike On Dec 16, 2008, at 6:03 AM, Scott Ellis wrote: I have a standard install, and I am trying to get a Pennytel gateway to register. After looking at Wireshark traces of x-lite registering and FreeSwitch registering, FreeSwitch is not sending any authentication information with the registration request. I am obviously missing something here! I understand for incoming calls you don't want authentication, but for outgoing it is obviously required. Is there a flag somewhere that I am supposed to set? The file was taken from the wiki page, and looks like it was previously tested when using the obsolete outbound directory structure. The following file is in the conf/sip_profiles/external directory. include gateway name="PennyTel" param name="username" value="8xxx"/ param name="password" value="xxx"/ param name="realm" value="sip.pennytel.com"/ param name="proxy" va
[Freeswitch-users] Pennytel Gateway Registration problem
I have a standard install, and I am trying to get a Pennytel gateway to register. After looking at Wireshark traces of x-lite registering and FreeSwitch registering, FreeSwitch is not sending any authentication information with the registration request. I am obviously missing something here! I understand for incoming calls you don't want authentication, but for outgoing it is obviously required. Is there a flag somewhere that I am supposed to set? The file was taken from the wiki page, and looks like it was previously tested when using the obsolete outbound directory structure. The following file is in the conf/sip_profiles/external directory. include gateway name=PennyTel param name=username value=888917/ param name=password value=xxx/ param name=realm value=sip.pennytel.com/ param name=proxy value=sip.pennytel.com/ param name=register value=true/ param name=expire-seconds value=60/ /gateway /include Thanks. 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] Jitter + Packet Loss
This is something that would make a great deal of the trouble shooting easier. The Linksys SPA942's that we use have some stats available for this, but it would be better to have it available centrally. Scott Jonathan Palley wrote: I'm curious to start a discussion on being able to query a channel and get statistics on the incoming jitter and packet loss (calculated from the RTP, not RTCP). Is this on the roadmap? Is it hard to do? Would be very useful for us indeed! Thanks - JP ___ 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
