December 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Segmentation fault (core dumped)
by Miha Zoubek 07 Dec '11

07 Dec '11

Hi, I have added this in mod_radius_cdr.c and run make&make install. if (channel){ const char *no_radius_start = switch_channel_get_variable(channel, "no_radius_start"); if (switch_true(no_radius_start)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Start\n"); return SWITCH_STATUS_SUCCESS; } } and for radius_stop. if (channel){ const char *no_radius_stop = switch_channel_get_variable(channel, "no_radius_stop"); if (switch_true(no_radius_stop)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Stop\n"); return SWITCH_STATUS_SUCCESS; } } When I run call a get this: Segmentation fault (core dumped) Do you know why I am getting this? I have also remove this lines and compile mod_radius_cdr.c again but I am still getting this. Regrdas, Miha

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Re: [Freeswitch-dev] hangup detection takes long
by murthy 06 Dec '11

06 Dec '11

In addition to what I wrote at " http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-December/005450.h… " I am not disconneting the call in the middle. I want to get the result immediatly when the other party rejects the call. I am using freeswitch-1.0.7. I have added a single call trace bellow and it took around 60 seconds. 2011-12-06 16:10:08.691084 [NOTICE] switch_channel.c:808 New Channel sofia/external/xxxxxxxxxx(a)192.168.0.213[2ca97a57-0536-40a8-b009-fb4161d75f71] 2011-12-06 16:10:08.691084 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_CREATE] **** 2011-12-06 16:10:08.691084 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_UUID] **** 2011-12-06 16:10:08.694071 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_OUTGOING] **** 2011-12-06 16:10:08.694071 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_ORIGINATE] **** 2011-12-06 16:10:08.694071 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:10:08.694071 [WARNING] sofia_glue.c:213 Codec PCMU payload 0 added to sdp wanting ptime 20 but it's already 30 (PCMU:0:30), disabling ptime. 2011-12-06 16:10:08.694071 [WARNING] sofia_glue.c:213 Codec PCMU payload 0 added to sdp wanting ptime 20 but it's already 30 (PCMU:0:30), disabling ptime. 2011-12-06 16:10:08.697237 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_CALLSTATE] **** 2011-12-06 16:10:08.697237 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:10:08.697237 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:10:11.439650 [INFO] sofia.c:729 sofia/external/ xxxxxxxxxx(a)192.168.0.213 Update Callee ID to "Outbound Call" <xxxxxxxxxx> 2011-12-06 16:10:11.439650 [NOTICE] mod_dummy.c:1856 **** event [CALL_UPDATE] **** 2011-12-06 16:10:11.439650 [NOTICE] mod_dummy.c:1856 **** event [CODEC] **** 2011-12-06 16:10:11.439650 [NOTICE] mod_dymmy.c:1856 **** event [CODEC] **** 2011-12-06 16:10:11.442639 [NOTICE] sofia_glue.c:3710 Pre-Answer sofia/external/xxxxxxxxxx(a)192.168.0.213! 2011-12-06 16:10:11.442639 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_CALLSTATE] **** 2011-12-06 16:10:11.442639 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_PROGRESS_MEDIA] **** 2011-12-06 16:10:25.033559 [NOTICE] mod_dummy.c:1856 **** event [HEARTBEAT] **** 2011-12-06 16:10:25.033559 [NOTICE] mod_dummy.c:1856 **** event [RE_SCHEDULE] **** 2011-12-06 16:10:45.030539 [NOTICE] mod_dummy.c:1856 **** event [HEARTBEAT] **** 2011-12-06 16:10:45.030539 [NOTICE] mod_dummy.c:1856 **** event [RE_SCHEDULE] **** 2011-12-06 16:11:05.026502 [NOTICE] mod_dummy.c:1856 **** event [HEARTBEAT] **** 2011-12-06 16:11:05.026502 [NOTICE] mod_dummy.c:1856 **** event [RE_SCHEDULE] **** 2011-12-06 16:11:08.026039 [NOTICE] switch_ivr_originate.c:3338 Hangup sofia/external/xxxxxxxxxx(a)192.168.0.213 [CS_CONSUME_MEDIA] [NO_ANSWER] 2011-12-06 16:11:08.026039 [INFO] mod_dummy.c:1135 ******** cancle cause: NONE 2011-12-06 16:11:08.029100 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/external/xxxxxxxxxx(a)192.168.0.213) Ended 2011-12-06 16:11:08.029100 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/xxxxxxxxxx(a)192.168.0.213 [CS_DESTROY] 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_CALLSTATE] **** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_HANGUP] **** 2011-12-06 16:11:08.029100 [INFO] mod_dummy.c:1863 *** got SWITCH_EVENT_CHANNEL_HANGUP with cause [NO_ANSWER] *** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_HANGUP_COMPLETE] **** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_DESTROY] **** 2011-12-06 16:11:08.029100 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_CALLSTATE] **** 2011-12-06 16:11:08.032037 [NOTICE] mod_dummy.c:1856 **** event [CHANNEL_STATE] **** 2011-12-06 16:11:08.789914 [ERR] mod_dummy.c:1607 Originate Failed. Cause: NO_ANSWER murthy.

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[Freeswitch-dev] hangup detection takes long
by murthy 05 Dec '11

05 Dec '11

When I originate a call from a module in FS using switch_ivr_originate(), if the callee rejects the call in the middle of the ringing, FS is not detecting the hangup cause and it is not getting the response immediatly. After some time (may be 60 seconds default value or the value of originate_timeout seconds) only it is getting NO_ANSWER (hangup cause) and comming out from the originate function. I have tried two ways to capture the result. One is event socket and other is using switch_ivr_originate function. Both are taking same time. How can I get the response/hangup cause immediatly when the calee rejects the call. Regards, Murthy.

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[Freeswitch-dev] freeswitch still core dumps on Debian SPARC
by daemonserj TVC 05 Dec '11

05 Dec '11

Subj happens while earlier start. root@tornado:/usr/local/freeswitch/bin# ./freeswitch 2011-12-05 16:48:31.251614 [INFO] switch_event.c:631 Activate Eventing Engine. 2011-12-05 16:48:31.264171 [DEBUG] switch_event.c:610 Create event dispatch thread 0 2011-12-05 16:48:32.228712 [INFO] switch_nat.c:419 Scanning for NAT 2011-12-05 16:48:32.229292 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2011-12-05 16:48:32.229907 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2011-12-05 16:48:32.229989 [DEBUG] switch_nat.c:424 Checking for UPnP 2011-12-05 16:48:44.230533 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2011-12-05 16:48:44.238417 [INFO] switch_core_sqldb.c:1908 Opening DB Bus error (core dumped) Machine: SunFire V240 Is it going to solve ?

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[Freeswitch-dev] mod_radius_cdr
by Miha Zoubek 05 Dec '11

05 Dec '11

Hi, I would like to disable mod_radius_cdr for incoming calls as we are getting cdr for incoming calls from SBC. So I would like to have mod_radius_cdr just for outgoing calls. I found this on Freeswithc JIRA: http://jira.freeswitch.org/browse/FS-1325 but after I make a call I get: Segmentation fault (core dumped). Thank you for all your help! Regards, Miha

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[Freeswitch-dev] Multicast listen for FS?
by Drew Terenzini 03 Dec '11

03 Dec '11

Good morning, I'm working on an integration of ETC speakers with FS (a stripped down install of FS running on Ubuntu 10.4). The speakers in use are Multicast-capable only (PCMU, PCMA, and G.729 codecs) and I've successfully gotten to broadcast audio from FS out through the speakers using a multi-cast paging group (PCMU). However, I'm not able to _receive_ multicast audio into FS and send into a conference or to other (non-speaker) attached devices. Tcpdump confirms that multicast packets from the speakers are seen at the eth0 interface on the Ubuntu 10.4 box. I've scoured the docs, the Wiki and the mailing lists, without success. Is there a multicast listening module that I've not yet found or am I chasing a concept that is not yet developed? Thanks for any help you could provide.

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[Freeswitch-dev] FOSDEM 2012 Telephony/Communications Devroom Call for Presenters
by Bryan M. Johns 02 Dec '11

02 Dec '11

Greetings, This is a call for presenters (with presentations!) for the telephony and communications devroom at FOSDEM 2012 - http://www.fosdem.org/. We will be holding a day full of presentations on development topics in the area of open source telephony and communications on Sunday, February 5th. The schedule allows for presentations from 9:00 to 17:00. The room we will be using will have a projector, wifi, and 100 seats. Please submit all proposals no later than 2011-12-09. Notification of accepted speakers will be provided by 2011-12-16. We will then work to have a schedule finalized by 2012-01-06. Talks should be submitted by subscribing to and then posting to the telephony-devroom mailing list hosted on http://lists.fosdem.org/. If you would like to contact the devroom organizer directly, please contact Kevin P. Fleming <kpfleming at digium.com>. Feel free to forward this along to any people or mailing lists that you think would be interested in this event. Thank you! Bryan M. Johns Digium, Inc. | Community Director Direct: 256.428.6007 http://twitter.com/bryanmjohns Check us out at : www.asterisk.org or www.digium.com

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