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Re: [Freeswitch-users] Hold Music not Playing when pressing Hold on Polycom Phone (Lyle Pratt)
by Lyle Pratt 08 Feb '14

08 Feb '14

An update on why this was happening so others will know: I disabled the Sip ALG on my router and everything now seems to be working fine. So, hold music will not work on some Polycom phones behind NAT with ALG connecting to remote Freeswitch servers. Thanks, Lyle Pratt On Thu, Feb 6, 2014 at 7:04 AM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: routing incoming SMS? (Privus P) > 2. Re: Filler Music in Call Recordings (Michael Jerris) > 3. Re: Hold Music not Playing when pressing Hold on Polycom > Phone (Michael Jerris) > > > ---------- Forwarded message ---------- > From: Privus P <privus007(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 12:58:35 +0000 > Subject: Re: [Freeswitch-users] routing incoming SMS? > Hi Donny, > > Perhaps you could share your javascript and lua script so that we could > try to figure out together what isn't working. > Judging by the apparent lack of response to this thread, it seems to be > the best way forward to try and resolve this. > > > On Thu, Feb 6, 2014 at 6:50 AM, Donny Hardyanto <hardyanto.donny(a)gmail.com > > wrote: > >> Hi, >> >> I have similar problem. I use gateway to GSM provider that support SIP >> simple . So far i can send sms from fs to gateway and receive sms in fs >> from gateway. But I was lost how to send the incoming sms from gateway to >> client. >> >> Also my client support sip simple also. When the sms from client arrive >> at fs, i dont know how to forward them to gateway. >> >> I use Javascript heavily and tried to produce the same thing as lua >> script in javascript but it always fail. The event fire command does not >> produce any thing. >> >> Can any one show us or point us how to forward sms? >> >> Dinny >> On Feb 4, 2014 8:06 PM, "Peter Villeneuve" <petervnv1(a)gmail.com> wrote: >> >>> Hi, >>> >>> Thanks for helping out. Indeed I would love to try out a lua script to >>> handle SMS. >>> Can anyone point to a sample script that I can study and play with? >>> >>> Any help is much appreciated. I'm still stuck with SMS hitting FS but >>> not being forwarded to the peer.... >>> >>> Thanks, >>> Peter >>> >>> >>> On Sun, Feb 2, 2014 at 5:46 PM, Privus P <privus007(a)gmail.com> wrote: >>> >>>> Hi Peter, >>>> >>>> I'm not really sure why your SMS isn't being routed correctly. >>>> I'm sure others in this list have a lot more experience and can help >>>> you out, but have you thought of using a lua script to handle SMS instead >>>> of relying on the send action? >>>> >>>> >>>> >>>> >>>> On Sat, Feb 1, 2014 at 2:00 PM, Peter Villeneuve <petervnv1(a)gmail.com>wrote: >>>> >>>>> Hi, >>>>> >>>>> I'm experimenting with GSMopen and I'm having trouble doing something >>>>> which is likely pretty simple. >>>>> I have enabled mod_sms and GSMopen in FS. I can send SIP simple >>>>> messages between 2 registered peers OK. >>>>> I can see that incoming GSM SMS messages arrive in FS as expected, but >>>>> I can't seem to get them converted into SIP SIMPLE format and routed to the >>>>> peer (1000). >>>>> Once they arrive in FS, I can see in the log: >>>>> >>>>> 2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text >>>>> message +4412398746->gsm01 in context default >>>>> Chatplan: gsm01 parsing [default->basic p2p] continue=true >>>>> Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ >>>>> break=on-false >>>>> Chatplan: gsm01 Action send() >>>>> >>>>> >>>>> My chatplan has: >>>>> >>>>> <?xml version="1.0" encoding="utf-8"?> >>>>> <include> >>>>> <context name="default"> >>>>> >>>>> <extension name="basic p2p" continue="true"> >>>>> <condition field="to" expression="^(.*)$"> >>>>>  >>>>> <action application="send"/> >>>>> </condition> >>>>> </extension> >>>>> >>>>> </context> >>>>> </include> >>>>> >>>>> And in my gsmopen.conf.xml: >>>>> >>>>> <configuration name="gsmopen.conf" description="GSMopen Configuration"> >>>>> <global_settings> >>>>> <param name="debug" value="8"/> >>>>> <param name="dialplan" value="XML"/> >>>>> <param name="context" value="default"/> >>>>> <param name="hold-music" value="$${moh_uri}"/> >>>>> <param name="destination" value="1000"/> >>>>> </global_settings> >>>>> >>>>> >>>>> I believe I'm doing something wrong in the chatplan. Is there any >>>>> transfer action like in the regular XML dialplan? >>>>> I tried creating in the default dialplan the following, hoping that it >>>>> would be similar to sofia calls and transfer the incoming SMS to peer 1000, >>>>> but no dice. >>>>> >>>>> <include> >>>>> <extension name="sms_inbound"> >>>>> <condition field="destination_number" expression="^(gsm01)$"> >>>>> <action application="transfer" data="1000 XML default"/> >>>>> </condition> >>>>> </extension> >>>>> </include> >>>>> >>>>> So, basically, how can I route the incoming SMS to peer 1000? >>>>> >>>>> Thanks >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting(a)freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users(a)lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 08:02:03 -0500 > Subject: Re: [Freeswitch-users] Filler Music in Call Recordings > You would have to make a media bug similar to the one for record session, > that has additional api commands to target the session by uuid that changes > a flag in a private struct of weather the recording is paused or not, and > then inside the callback for the recording, trigger different behavior when > its paused. As there is no audio being muxed, there is no reason at all to > use mod_conference. > > On Feb 5, 2014, at 7:42 PM, JP <jaykris(a)gmail.com> wrote: > > I'm trying to implement a feature that allows a user to selectively pause > and resume a call recording via FreeSWITCH's event socket interface. The > audio file should contain filler music for the duration of each pause > operation. Is there any simple straight forward way to accomplish this? I > would prefer avoiding the overhead incurred by using the conferencing > module for this if possible. > > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 6 Feb 2014 08:04:04 -0500 > Subject: Re: [Freeswitch-users] Hold Music not Playing when pressing Hold > on Polycom Phone > I don't see the phone actually sending the hold signal there, try turning > on sip trace and confirming what the phone is actually sending. Compare > that to your dev environment where it works. > > Mike > > On Feb 6, 2014, at 12:11 AM, Lyle Pratt <lylepratt(a)gmail.com> wrote: > > 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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FSClient Compile
by Nyamul Hassan 08 Feb '14

08 Feb '14

Hi, Is this the right forum to ask about FSClient compile problems? If not, can you please guide me to the forum which is? If yes, I would be grateful if you could help me out here. I downloaded FreeSwitch and the build on VS2012 completed successfully (without mod_managed, mod_cephstral, mod_v8, mod_v8_skel & libv8). Whenever I start compiling FSClient, I get the following error: Error 1 The command "if not exist ..\..\"external items\Release\" md ..\..\"external items\Release\" if exist "%FREESWITCH_SRC_LOCATION%" copy "%FREESWITCH_SRC_LOCATION%"\Win32\Release\mod\FreeSWITCH.Managed.dll ..\..\"external items\Release\"" exited with code 1. FSClient Could you please help with what I am doing wrong? I am following the instructions on http://wiki.freeswitch.org/wiki/FSClient. I must be misunderstanding / missing something. Regards HASSAN

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Filler Music in Call Recordings
by JP 08 Feb '14

08 Feb '14

I'm trying to implement a feature that allows a user to selectively pause and resume a call recording via FreeSWITCH's event socket interface. The audio file should contain filler music for the duration of each pause operation. Is there any simple straight forward way to accomplish this? I would prefer avoiding the overhead incurred by using the conferencing module for this if possible. Thanks -JP

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jitter issue with FreeSWITCH Version 1.2.10
by Yuriy Nasida 08 Feb '14

08 Feb '14

Hello, We have noted jitter issue with this version of FS. The only ERR i see in logs now is:debce20e-eb3e-4388-89df-07e412000777 2014-02-06 14:25:21.397009 [ERR] switch_core_session.c:2762 Error sofia/external/dest_num@IP:5060 too many stacked extensionsNot sure that this means exactly Problem can be temporary solved by means of restarting of FS but after some time this happens again. I switch all calls from FS node before restating and also noted that FS eats ~10% CPU without any calls. FreeSWITCH Version 1.2.10+git~20130521T182905Z~b73e2c51bb (git b73e2c5 2013-05-21 18:29:05Z) In my opinion looks like that issue connected with this version of FS. Please advice. Thanks.

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[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

1 0

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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