April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to leave voicemail message longer than a few seconds
by Mark Campbell-Smith 14 Apr '11

14 Apr '11

HI! I am having an issue which must have been introduced at my last upgrade of FS (FreeSWITCH Version 1.0.head (git-b737851 2011-03-25 00-13-39 +0100)) Callers are unable to leave voicemail messages of any length; the voicemail application terminates the recording and transfers to the ivr. My dialplan is quite simple. Extract here: <action application="bridge" data="${group_extension}"/> <action application="set_user" data="${dialed_extension}@${domain}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> And the only place the message length value is defined is in voicemail.conf.xml in the autoloads conf directory. It is set to default. <param name="max-record-len" value="300"/> Below is an extract of the debug logs. I am only allowed to leave a message in this case for 6 seconds (between times 2011-04-13 21:26:41.098532 and 2011-04-13 21:26:47.37119) Dialplan: sofia/internal/1000(a)192.168.1.120 Action set(call_timeout=22) Dialplan: sofia/internal/1000(a)192.168.1.120 Action set(sip_contact_user=${caller_id_number}) Dialplan: sofia/internal/1000(a)192.168.1.120 Action export(sip_contact_user=${caller_id_number}) Dialplan: sofia/internal/1000(a)192.168.1.120 Action info() Dialplan: sofia/internal/1000(a)192.168.1.120 Action bind_meta_app(1 a s execute_extension::4001 XML default) Dialplan: sofia/internal/1000(a)192.168.1.120 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1000(a)192.168.1.120 Action bind_meta_app(2 a s execute_extension::checkpin XML features) Dialplan: sofia/internal/1000(a)192.168.1.120 Action bind_meta_app(9 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1000(a)192.168.1.120 Action bridge(${group_extension}) Dialplan: sofia/internal/1000(a)192.168.1.120 Action set_user(${dialed_extension}@${domain}) Dialplan: sofia/internal/1000(a)192.168.1.120 Action answer() Dialplan: sofia/internal/1000(a)192.168.1.120 Action sleep(1000) Dialplan: sofia/internal/1000(a)192.168.1.120 Action voicemail(default ${domain_name} ${dialed_extension}) : : 2011-04-13 21:26:29.611121 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000(a)192.168.1.120 [BREAK] 2011-04-13 21:26:29.611121 [DEBUG] switch_channel.c:2813 (sofia/internal/ 1000(a)192.168.1.120) Callstate Change EARLY -> ACTIVE 2011-04-13 21:26:29.611121 [NOTICE] mod_dptools.c:929 Channel [sofia/internal/1000(a)192.168.1.120] has been answered EXECUTE sofia/internal/1000(a)192.168.1.120 sleep(1000) 2011-04-13 21:26:29.702581 [DEBUG] sofia.c:4744 Channel sofia/internal/ 1000(a)192.168.1.120 entering state [ready][200] 2011-04-13 21:26:29.758583 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed. EXECUTE sofia/internal/1000(a)192.168.1.120 voicemail(default markcs.dyndns.org 1000) 2011-04-13 21:26:30.750645 [DEBUG] switch_ivr_play_say.c:1291 Codec Activated L16@8000hz 1 channels 30ms 2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:1635 done playing file 2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en] 2011-04-13 21:26:41.098532 [DEBUG] switch_ivr_play_say.c:581 Raw Codec Activated 2011-04-13 21:26:41.098532 [DEBUG] switch_core_codec.c:116 sofia/internal/ 1000(a)192.168.1.120 Push codec L16:70 2011-04-13 21:26:47.371195 [DEBUG] switch_core_codec.c:141 sofia/internal/ 1000(a)192.168.1.120 Restore previous codec PCMU:0. 2011-04-13 21:26:47.371195 [DEBUG] switch_ivr_play_say.c:63 No language specified - Using [en] 2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en) 2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:1291 Codec Activated L16@8000hz 1 channels 30ms 2011-04-13 21:26:47.758529 [DEBUG] switch_ivr_play_say.c:1635 done playing file 2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1] (en:en) 2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:1291 Codec Activated L16@8000hz 1 channels 30ms Any ideas why this would be? I checked Jira but couldn't find any reference to bugs there. Thanks!

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Cannot compile freetdm
by Valery Kalinin 14 Apr '11

14 Apr '11

# cd /usr/local/freeswitch/libs/freetdm # ./configure --with-libisdn # make bla-bla-bla cc1: warnings being treated as errors src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34': src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen' src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span': src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of 'Q931SetLogCB' from incompatible pointer type make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1 Why? libisdn-0.0.1 installed

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fax and jpg to tiff
by Madovsky 14 Apr '11

14 Apr '11

sorry I don't have the personal email of Steve Underwood (I saw a thread of him related of this subject), it's about a problem of spandsp txfax and jpeg files. I didn't succeed to find the right conversion to make the far end accept the fax. I get always "far end cannot receive at the size of the image". this problems occurs only from jpeg to tiff conversion. I use imagemagick for that : sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184 -units pixelsperinch -monochrome -compress Fax <fileToConvert> <fileToSave> I tried numrous options but no luck. Thanks

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Call state notifications
by Rogelio Perez 13 Apr '11

13 Apr '11

Hi everyone, I'm building a simple callshop panel for FS and I need to show the call state for every booth on a web page. What is the best way to notify my application about call state changes (ringing, connected, disconnected)? I'm using mod_python to send the notifications, but I'm not sure how to call the python script inside the dial plan. Are there any examples out there? Thanks! Rogelio Perez

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occasional ~5s delay during bind_meta_app execute_extenstion
by elijah 13 Apr '11

13 Apr '11

I have setup a feature where users who have dialed outbound have the option to transfer the b-leg to a separate outbound number, like so: <extension name="outbound_9"> <condition field="destination_number" expression="^9(\d{10})$"> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bind_meta_key=*"/> <action application="bind_meta_app" data="1 a s execute_extension::dx XML mycontext"/> <action application="transfer" data="to_outbound XML mycontext"/> </condition> </extension> <extension name="dx"> <condition field="destination_number" expression="^dx$"> <action application="log" data="INFO transferring the b-leg..."/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="playback" data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/transfer2.wav"/> <action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="execute_extension" data="is_transfer XML mycontext"/> </condition> </extension> A delay sometimes occurs after users have pressed *1 and before the message "transferring the b-leg..." appears in the console. This delay happens ~1/3 of the time and lasts for ~5s. Do you know how I should execute this functionality in a way that will not cause such a delay? thanks, elijah

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codec negotiation
by Sameer Khan 13 Apr '11

13 Apr '11

hello every 1 i need help regarding codec negotiation I set abs codec string in my dialplan $xml_output .='<action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/>'; but still leg B is carrying the same codecs as leg A disable_transcoding is false in my internal sip profile

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Re: [Freeswitch-users] Event queue is full.
by Gary Chen 13 Apr '11

13 Apr '11

I am not doing load test and my sipp is fairly simple. It just send call and wail for short time period and then send bye to end the call. The system I am using has plenty of power. Last night it only used about 15% of cpu for 300 concurrent calls. We are in the process of exploring options to replace our aging commercial sip feature server. I just want to make sure FS does not have any memory leak issue and can run long enough without rebooting. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, April 13, 2011 11:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Event queue is full. you are doing something abusive with sipp we do not entertain and assist with Ddos or load testing related subjects. On Wed, Apr 13, 2011 at 8:56 AM, Gary Chen <gchen00(a)insightbb.com> wrote: > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian > West > Sent: Wednesday, April 13, 2011 9:34 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Event queue is full. > > > > Sounds like an old code base.. you sure that other box is using the latest > code? > > > > /b > > > > On Apr 13, 2011, at 7:44 AM, Gary Chen wrote: > > > >> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500) > >> > >> Last night I start Sipp to sent calls to my test FS1 and FS1 in turn >> forward calls to another FS2 playing Music On Hold. I got about 300 >> concurrent calls with media going through FS1 before I left. > >> This morning, the FS1 is not responding. I fs_cli into it and got a lot of >> messages like this: > >> 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is full! > >> x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is >> full! > >> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full! > >> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full! > >> 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is full! > >> 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is full! > > > > Call Flow: Sipp ---> FS1 --->FS2(play music on hold) > > FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 > -0500) > > FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z) > > FS2 has older version. But 'Event queue is full' only happen on FS1. Should > have nothing to do with FS2. Right now FS2 still show 304 sessions. That > means that it still playing music on 304 calls. > > I can not run fs_cli -x status because the event queue problem. > > Gary > > _______________________________________________ > > 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 > > _______________________________________________ > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900_______________________________________________ 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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called number rewrite
by budi wibowo 13 Apr '11

13 Apr '11

hi is it possible for freeswitch to change / rewrite called number. say, i call 1234567 and FS will change the called number to 44444444 thx budi

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Re: [Freeswitch-users] Event queue is full.
by Gary Chen 13 Apr '11

13 Apr '11

-----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, April 13, 2011 9:34 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Event queue is full. Sounds like an old code base.. you sure that other box is using the latest code? /b On Apr 13, 2011, at 7:44 AM, Gary Chen wrote: > FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500) > > Last night I start Sipp to sent calls to my test FS1 and FS1 in turn forward calls to another FS2 playing Music On Hold. I got about 300 concurrent calls with media going through FS1 before I left. > This morning, the FS1 is not responding. I fs_cli into it and got a lot of messages like this: > 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is full! > x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is full! > 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full! > 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full! > 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is full! > 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is full! Call Flow: Sipp ---> FS1 --->FS2(play music on hold)FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)FS2 has older version. But 'Event queue is full' only happen on FS1. Should have nothing to do with FS2. Right now FS2 still show 304 sessions. That means that it still playing music on 304 calls.I can not run fs_cli -x status because the event queue problem. Gary _______________________________________________ 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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Lua session originate
by Eric Beard 13 Apr '11

13 Apr '11

When I try to use lua to originate a call, I get this: freeswitch@internal> lua test_originate.lua -ERR encountered freeswitch@internal> 2011-04-12 13:03:17.806648 [ERR] mod_lua.cpp:191 Error in originate expected 4. .4 args, got 3 stack traceback: [C]: in function 'originate' /usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk I'm using the following line of code: local new_session = freeswitch.Session(); new_session.originate(session, "sofia/gateway/mygatewayname/mynumber", 60); This is in the Wiki: session:originate local new_session = freeswitch.Session(); new_session.originate(session, dest[, timeout]); dest - quoted dialplan destanation. For example: "sofia/internal/1000(a)10.0.0.1" or "sofia/gateway/my_sip_provider/my_dest_number" timeout - origination timeout in seconds Note: in recent git versions, session.originate expects 4 arguments; the example above does not work (anymore?). What is the 4th arg? ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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