February 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[REQUESTED_CHAN_UNAVAIL] error with Sangoma FreeTDM
by Neil Patel 08 Feb '14

08 Feb '14

Hi All, Several of my PRIs in India have recently been getting this error from time to time. Every time it happens, I can temporarily fix it by restarting FS: 2014-02-03 13:11:25.713142 [NOTICE] mod_freetdm.c:2728 Hangup FreeTDM/3:4/0942847XXXX [CS_INIT] [REQUESTED_CHAN_UNAVAIL] 2014-02-03 13:11:25.713142 [DEBUG] switch_ivr_originate.c:3639 Originate Resulted in Error Cause: 44 [REQUESTED_CHAN_UNAVAIL] Our system sends 30 calls at a time to the PRI gateway, and this error comes up on any one of the channels. The channel is not in use at the time, based on 'show channels'. FreeTDM does not bypass the channel and try another, so if it's channel 14 that's unavailable it doesn't move on to 15, and only 1-14 get used (until I restart FS). Sangoma support has examined a trace and found the d-channel on affected lines fluctuating. Also I notice that around the time this problem crops up the line suddenly flicks signals: 2014-02-03 17:25:00.877663 [DEBUG] ftmod_sangoma_isdn_cntrl.c:45 [s2c1][2:1] Signalling link status changed to UP 2014-02-03 17:25:00.877663 [DEBUG] ftmod_sangoma_isdn_cntrl.c:45 [s2c2][2:2] Signalling link status changed to UP 2014-02-03 17:25:00.877663 [DEBUG] ftmod_sangoma_isdn_cntrl.c:45 [s2c3][2:3] Signalling link status changed to UP 2014-02-03 17:25:00.877663 [DEBUG] ftmod_sangoma_isdn_cntrl.c:45 [s2c4][2:4] Signalling link status changed to UP 2014-02-03 17:25:00.877663 [DEBUG] ftmod_sangoma_isdn_cntrl.c:45 [s2c5][2:5] Signalling link status changed to UP ... I'm also wondering whether I should give a time gap between originate commands? Perhaps inserting a half second or so between calls will prevent the channels from getting jammed up. I'm not sure where the issue lies. With the telco (fluctuating d-channel), with FreeTDM, perhaps with my Sangoma hardware or driver (currently running 7.0.8 and libsng_isdn 7.28.0)? Thanks, Neil

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separate log files
by Andy Kliman' 08 Feb '14

08 Feb '14

Hi! Faced with the issue. I need to save detailed log for each call in separate txt file. Have any ideas about it? Will be very appreciate for some easy sollution. In any case thanks for any help! -- Best Regards Tel: +38 068 252 13 76 Skype: NewAndy3

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FS 1.2.18 won't compile on FreeBSD 9.2-RELEASE
by Tim St. Pierre 08 Feb '14

08 Feb '14

Hello, I'm trying to do an update to a current version, and I can't even get the latest GIT to bootstrap. Lots of automake an autoconf errors. It seems to find all the dependencies: bootstrap: autoconf version 2.69 (ok) bootstrap: automake version 1.14 (ok) bootstrap: aclocal version 1.14 (ok) bootstrap: libtool version 2.4.2 (ok) But it's constantly complaining about definitions in every Makefile it encounters. It eventually stops here: src/mod/asr_tts/mod_unimrcp/Makefile.am:34: error: ISMAC does not appear in AM_CONDITIONAL src/mod/codecs/mod_com_g729/Makefile.am:6: error: ISLINUX does not appear in AM_CONDITIONAL src/mod/endpoints/mod_portaudio/Makefile.am:14: error: ISMAC does not appear in AM_CONDITIONAL src/mod/endpoints/mod_sofia/Makefile.am:35: error: ISMAC does not appear in AM_CONDITIONAL src/mod/formats/mod_portaudio_stream/Makefile.am:14: error: ISMAC does not appear in AM_CONDITIONAL src/mod/formats/mod_portaudio_stream/Makefile.am:9: warning: source file '$(MODPA_DIR)/pablio.c' is in a subdirectory, src/mod/formats/mod_portaudio_stream/Makefile.am:9: but option 'subdir-objects' is disabled src/mod/formats/mod_portaudio_stream/Makefile.am:9: warning: source file '$(MODPA_DIR)/pa_ringbuffer.c' is in a subdirectory, src/mod/formats/mod_portaudio_stream/Makefile.am:9: but option 'subdir-objects' is disabled src/mod/languages/mod_spidermonkey/Makefile.am:50: error: HAVE_ODBC does not appear in AM_CONDITIONAL There is a configure script, but if I run it, all I get is: ./configure: 2719: Syntax error: word unexpected (expecting ")") Has anyone else been able to compile a recent version on FreeBSD? Thanks! -Tim

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SIP + IPv6
by Kristian Kielhofner 08 Feb '14

08 Feb '14

All, What IPv6 accessible SIP resources (echo test, conference, etc) are publicly available on the internet? I found this but every wiki page seems to direct to full-page ads: http://qxip.net/mediawiki/index.php/Main_Page Thanks! -- Kristian Kielhofner

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Hold Music not Playing when pressing Hold on Polycom Phone
by Lyle Pratt 08 Feb '14

08 Feb '14

Hi Guys, When I press the HOLD button (or start to transfer someone, join a conference, etc) the configured hold music will not play. I previously had this working in production, but all of the sudden it is no longer working I can't figure out why. It works fine in my development environment on my local network, but not when I'm registering to Public IPs. I first thought it was a Polycom specific NAT issue, but then I tried it with a different phone (Cisco) from a different network with no success either. Here is a Pastebin of the Freeswitch console logs when I press the hold button on my device and no hold music occurs. http://pastebin.com/p6xtwyz7 Does anyone have any ideas on how on earth I messed this up? Its the last piece of the puzzle I have in completely getting this working! Any help appreciated. Thanks, Lyle Pratt

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Frequent connect/disconnect on PRI line leaves mod_freetdm in stuck state
by Deepika Agarwal 08 Feb '14

08 Feb '14

Hello All, We are facing a peculiar problem in which the frequent connect/disconnect events on the PRI line results in FreeSwitch not able to handle any further incoming calls. The workaround to the above problem is to *restart mod_freetdm* or the complete freeswitch. I'm providing some of the logs with my analysis: 30111 2014-01-31 10:32:53.522513 [NOTICE] mod_freetdm.c:1889 Alarm raised on channel 13:15 *30112 2014-01-31 10:32:53.522513 [DEBUG] ftmod_wanpipe.c:1630 [s3c7][3:7] read wanpipe event 1730113 2014-01-31 10:32:53.522513 [NOTICE] mod_freetdm.c:1889 Alarm raised on channel 3:2230114 2014-01-31 10:32:53.522513 [INFO] ftmod_sangoma_isdn_stack_rcv.c:663 [SNGISDN PHY] D-chan 3 : T1/E1/BRI Disconnected30115 2014-01-31 10:32:53.522513 [DEBUG] mod_freetdm.c:2416 got clear channel sig [ALARM_TRAP]* 30116 2014-01-31 10:32:53.522513 [DEBUG] ftmod_wanpipe.c:1385 [s3c7][3:7] Ignoring wanpipe link disconnected event 30117 2014-01-31 10:32:53.522513 [INFO] ftmod_sangoma_isdn_stack_rcv.c:663 [SNGISDN PHY] D-chan 3 : T1/E1/BRI Disconnected 30118 2014-01-31 10:32:53.522513 [DEBUG] mod_freetdm.c:2416 got clear channel sig [ALARM_TRAP] 30119 2014-01-31 10:32:53.522513 [DEBUG] ftmod_wanpipe.c:1630 [s3c8][3:8] read wanpipe event 17 30313 2014-01-31 10:32:53.682516 [INFO] ftmod_sangoma_isdn_stack_rcv.c:1004 sng_isdn->s16: Invalid Q.921/Q.931 frame - ignoring len:1 30314 2014-01-31 10:32:53.722506 [DEBUG] mod_freetdm.c:866 Dropping frame! (write note ready) in channel FreeTDM/3:8/43850327 device 3:8! 30315 2014-01-31 10:32:53.722506 [DEBUG] mod_freetdm.c:866 Dropping frame! (write note ready) in channel FreeTDM/3:4/43850118 device 3:4! 30316 2014-01-31 10:32:53.722506 [DEBUG] ftmod_wanpipe.c:945 [s14c11][14:11] Rx Queue length exceeded 80% hreshold (9/10) 30317 2014-01-31 10:32:53.722506 [DEBUG] ftmod_wanpipe.c:945 [s23c11][23:11] Rx Queue length exceeded 80% hreshold (9/10) 30318 2014-01-31 10:32:53.722506 [DEBUG] mod_freetdm.c:866 Dropping frame! (write note ready) in channel FreeTDM/3:5/43851158 device 3:5! 30319 2014-01-31 10:32:53.722506 [DEBUG] ftmod_wanpipe.c:945 [s24c2][24:2] Rx Queue length exceeded 80% hreshold (9/10) 30320 2014-01-31 10:32:53.722506 [DEBUG] mod_freetdm.c:866 Dropping frame! (write note ready) in channel FreeTDM/3:6/43850118 device 3:6! 30321 2014-01-31 10:32:53.722506 [DEBUG] ftmod_wanpipe.c:945 [s6c17][6:18] Rx Queue length exceeded 80% hreshold (9/10) 30322 2014-01-31 10:32:53.742514 [DEBUG] mod_freetdm.c:866 Dropping frame! (write note ready) in channel FreeTDM/3:3/43850523 device 3:3! *30323 2014-01-31 10:32:53.762508 [WARNING] mod_freetdm.c:763 Too many timeouts while waiting I/O in channel FreeTDM/3:6/43850118 device 3:6!30324 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:6/43850118 device 3:6!* 30325 2014-01-31 10:32:53.762508 [WARNING] mod_freetdm.c:763 Too many timeouts while waiting I/O in channel FreeTDM/3:1/43852488 device 3:1! 30326 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:1/43852488 device 3:1! Because of the timeout in the read channel, the TFLAG_IO gets cleared and leads to a recurrent logs w.r.t the TFLAG_IO flag (as shown in line 30413) 30403 2014-01-31 10:32:53.762508 [WARNING] mod_freetdm.c:763 Too many timeouts while waiting I/O in channel FreeTDM/3:8/43850327 device 3:8! 30404 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:8/43850327 device 3:8! 30405 2014-01-31 10:32:53.762508 [DEBUG] switch_ivr_bridge.c:505 FreeTDM/3:8/43850327 ending bridge by request from read function 30406 2014-01-31 10:32:53.762508 [WARNING] mod_freetdm.c:763 Too many timeouts while waiting I/O in channel FreeTDM/3:4/43850118 device 3:4! 30407 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:4/43850118 device 3:4! 30408 2014-01-31 10:32:53.762508 [DEBUG] switch_ivr_bridge.c:505 FreeTDM/3:4/43850118 ending bridge by request from read function 30409 2014-01-31 10:32:53.762508 [WARNING] mod_freetdm.c:763 Too many timeouts while waiting I/O in channel FreeTDM/3:10/43115865 device 3:10! 30410 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:10/43115865 device 3:10! 30411 2014-01-31 10:32:53.762508 [DEBUG] switch_ivr_play_say.c:1678 done playing file /srv/sounds/usr/159d/0491/-a6f/0-48/e4-8/1df-/1823/c766/9 6e8.mp3 30412 2014-01-31 10:32:53.762508 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels 20ms *30413 2014-01-31 10:32:53.762508 [DEBUG] mod_freetdm.c:747 TFLAG_IO is not set in channel FreeTDM/3:10/43115865 device 3:10!* 30414 2014-01-31 10:32:53.762508 [ERR] mod_freetdm.c:808 clearing IO in channel FreeTDM/3:10/43115865 device 3:10! Upon further looking at the code, it appears that the TFLAG_IO gets cleared because of the timeout (as a result of PRI connect/disconnect), but as this flag is not inturn set upon PRI line recovery, the channel_read_frame (in mod_freetdm.c) is not able to read any further frames. Please provide your comments in this regard and any further inputs would be much appreciated. Thanks Deepika

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Help debug "Failed to write to Sangoma decoder RTP session"
by Antonio Silva 08 Feb '14

08 Feb '14

Hi, I have transconding card sangoma D100 and it works very well, but sometimes (1 or two times during the day on a server with 6calls/seconds) i get the error messages: " 2014-02-07 15:07:51.490715 [ERR] mod_sangoma_codec.c:517 Failed to write to Sangoma encoder RTP session. ae118c22-9d7a-4455-865e-1e9ea42ad487 2014-02-07 15:07:51.490715 [ERR] switch_core_io.c:1376 Codec Sangoma G729 encoder error! " What could be the cause for this problem? How can i debug it and try to fix it? Thanks, António

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BSD Maintainers needed...
by Brian West 08 Feb '14

08 Feb '14

FreeSWITCHers, Is anyone willing and able to maintain any of the BSD platforms? Currently you can not compile on Net/Open/FreeBSD with ease... OpenBSD and NetBSD are already well on their way to the unsupported due to no activity in the community to assist in maintaining these platforms. If you wish to assist in maintaining any or all of these platforms please let me know. Thoughts? Feedback? Feb. 12th we’ll be talking about this during our weekly call. Thanks, -- Brian West brian(a)freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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Redirecting traffic - Freeswitch vs Kamailio
by Alex Lake 07 Feb '14

07 Feb '14

If I want to set up a SIP redirector for small call volumes (eg. <1/second), can Freeswitch do as good a job as Kamailio? The idea is that we get a call in from a PSTN-SIP gateway and then re-route to another box depending on the number being called. Just wondering if there's any difference in the signalling, which I presume is either a 302 or a SIP re-invite (but please correct me if I'm wrong!). At what kind of call volumes might one choose to switch to a more dedicated SIP proxy?

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sip_history_info value
by Giuseppe Cardone 07 Feb '14

07 Feb '14

Hi, I have FS 1.2.14 running and I'm testing it's behaviour in call forwarding scenarios. I'd like to include the SIP History-Info-Header as per RFC 4244 in forwarded calls, so I guess the sip_history_info channel variable should hold this data. However, when I show its contents in the fs_cli debugs by using the log action in the relevant part of the dialplan with: <action application="log" data="LOG SIP History info is '${sip_history_info}'"/> it appears to be empty: [DEBUG] mod_dptools.c:1595 Says SIP History info is '' Any clues as to how this variable should be used ? Thanks in advance, Giuseppe

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