August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to match a message sent using chat api with MESSAGE event
by Leonardo 02 Aug '12

02 Aug '12

1 0

Conditional testing on caller_id_name
by Phil Quesinberry 02 Aug '12

02 Aug '12

Thanks Brian, I think the lights just came on... I think I finally see what's going on here. Looking through the wiki some more, I came across the following: "The XML Dialplan has the ability to test a number of conditions based upon variables with expressions; however, it needs to be understood that some variables may not be available for conditional testing until the first transfer or execute_extension is performed." When I do a conditional test such as the following: <condition field="caller_id_name" expression="^Currently running a lookup"> the console output shows the ani instead of the CNAM in parentheses as shown, and fails: Dialplan: sofia/external/4105551212(a)140.239.xx.xx Regex (FAIL) [currently_running] caller_id_name(4105551212) =~ /^Currently running a lookup/ break=on-false BUT when I ask for the variable as follows: <action application="log" data="${caller_id_name}"/> I get the expected value in the console output: EXECUTE sofia/external/4105551212(a)140.239.xx.xx log(SMITH,JOHN) 2012-08-02 12:12:50.679592 [DEBUG] mod_dptools.c:1458 SMITH,JOHN (the names and numbers have been changed to protect the clueless) So just to be sure, is this in fact what's going on here? I learned something today... now that I understand it, I can deal with it. By next year I'm bound to comprehend things well enough to get something useful from ClueCon. Thanks! The rest of the message that follows was written before I came across this little tidbit of information but I'm keeping it in here for reference: <clueless banter follows> I have no idea how to use Perl in FS but after a bit of looking around it appears that I would need to do something like: <action application="perl" data="/usr/src/freeswitch/scripts/perl/cnam.cgi $1"/> Is that correct? Better yet, is there any way to get to the CID name variable without having to resort to Perl? Doing an 'info' in the dialplan during an incoming call shows the following: Caller-Caller-ID-Name: [JOHN SMITH] How can I get to that data? 'caller_id_name' just seems to contain the number instead of the CNAM info, which doesn't make any sense. The wiki shows the following example to set a variable directly: <action application="set" data="cid_name=${cidlookup(${caller_id_number})}"/> and this looks like it might work, but I'd like to understand why the variables aren't already set after the cidlookup is done in public.xml. <clueless banter ends/> Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com ---------- [Freeswitch-users] Setting effecting_caller_id_name Brian West Sat Jul 28 17:46:14 MSD 2012 Sounds like your doing openCNAM, check the cnam.cgi in tree, you should be checking the response code and not the text returned. I built the wrapper to clean up the input before I query because of this. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 27, 2012, at 10:06 AM, Ken Rice <krice at freeswitch.org> wrote: Re: [Freeswitch-users] Setting effecting_caller_id_name This exmaple should be correct provided that caller_id_name containts "Currently runnint a lookup" with this pinned to that start of the field... On that extension in your dialplan toss a <anti-action application="info"/> If your regex isnt matching on the caller_id_name field the above will show you whats in there (along with every thing else too heh) K

3 2

Freeswitch Auto Retrying a call for 3 times , if it is not answered
by ramesh 02 Aug '12

02 Aug '12

Hi Team, I made a script to dial a user , and if not answered the second user should get the dial. Everything works file , Except the first user is dialed for 3 times , if the user is busy or unanswered , i was trying to figure out what could be the problem for this issue, but i couldn't . Can Anyone Figure out what is the reason for this issue. below is my lua dial script session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919xxxxxxxx|sofia/gateway/bandwidth.com/+919xxxxxxxx"); following is the log i can get. Any Help or suggestion would be really helpful! [NOTICE] sofia.c:5594 Ring-Ready sofia/internal/+919677080275! 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2012-08-01 07:09:13.212635 [DEBUG] switch_core_codec.c:216 rtmp/default/+542914850488 Push codec L16:70 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5513 Remote SDP: v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/+919677080275 PCMU/8000 20 ms 160 samples 64000 bits 2012-08-01 07:09:16.552637 [DEBUG] switch_core_codec.c:111 sofia/internal/+919677080275 Original read codec set to PCMU:0 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/internal/+919677080275] 10.244.15.45 port 30506 -> 65.115.130.14 port 10382 codec: 0 ms: 20 2012-08-01 07:09:16.552637 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-08-01 07:09:16.552637 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/internal/+919677080275! 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2936 (sofia/internal/+919677080275) Callstate Change RINGING -> EARLY 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2978 Send signal rtmp/default/+542914850488 [BREAK] 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec 480,620 index 0 hits 1 2012-08-01 07:09:16.572648 [DEBUG] switch_core_media_bug.c:457 Attaching BUG to sofia/internal/+919677080275 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec 1776.7 index 1 hits 2 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2877 sofia/internal/+919677080275 bug already running 2012-08-01 07:09:16.572648 [DEBUG] switch_rtp.c:3205 Correct ip/port confirmed. 2012-08-01 07:09:16.572648 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:38.512638 [DEBUG] libdingaling.c:1624 Sent keep alive signal 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:10:11.012635 [DEBUG] switch_core_codec.c:241 rtmp/default/+542914850488 Restore previous codec SPEEX:99. 2012-08-01 07:10:11.012635 [DEBUG] switch_channel.c:2852 (sofia/internal/+919677080275) Callstate Change EARLY -> HANGUP 2012-08-01 07:10:11.012635 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/+919677080275 [CS_CONSUME_MEDIA] [NO_ANSWER] Thanks Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Auto-Retrying-a-ca… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 1

Am I using play_and_detect_speech correctly?
by Liu, Xingkun 02 Aug '12

02 Aug '12

Hello, I am using play_and_detect_speech with Java ESL in my IVR applications. Previously I call it again each time after I receive any recognition event, like recognition complete, no-input-timeout, or recognition-timeout, it seems to work fine. Now I have changed my app to issue play_and_detect_speech command based on my available system utterances as well as the speech event. I.e., I use a separate thread to constantly check if there is a system utterance coming in from another component of my application, if there is any utterance I issue the command which will speak the new utterance and listen to user input no matter whether or not previous command has finished. And if there is any speech event (recognition result, timeout etc.) the play_and_detect_speech command is also issued but with playing silence. Obviously the new command will stop the utterance speaking of the previous command if it is not finished. My question is will the new play_and_detect_speech command also stop the previous ASR listening or will there be many ASR listening channel and sending speech data (or silence) to ASR server? Do I need to explicitly issue a "stop" commnad before issuing a new play_and_detect_speech? If yes, how to do that, by "detect_speech stop"? Recently there is a network traffice problem (lots of connections /data transportation to the ASR server machine) when I am running my application. I am not sure if this is because of other issues or because of my new changes to the way of using play_and_detect_speech. Please any one could shed a light on this? Many thanks! Xing -- Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278.

4 8

ESL - phpmod compile Error
by Lloyd Aloysius 02 Aug '12

02 Aug '12

Hi All: when I compile the ESL module for php . OS - Centos 6.2 / 64 bit libs/esl make phpmod I got the following Errors. make phpmod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/src/freeswitch.git/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/src/freeswitch.git/libs/esl/php' g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib6464 -lcrypt -lcrypt -ledit -lncurses -lm -ldl -lnsl -lm -ldl -ldl -lm -lcrypt -lm -lcrypt -lpthread -o ESL.so -L. /usr/bin/ld: cannot find -ledit collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/usr/src/freeswitch.git/libs/esl/php' make: *** [phpmod] Error 2 How to solve this issue. Thanks Lloyd

2 3

Re: [Freeswitch-users] multi-tenant setup - call goes to wrong place
by Mr Nathan Downes 02 Aug '12

02 Aug '12

I had setup shared line appearance on one tenant, so had to re-enable it for them to make calls, further investigation shows that manage-presence causes the problem and it only affects domain listed in presence-hosts. I took domain3.com out of presence-hosts and when I call from 101(a)domain3.com to 201 I do not get 201(a)domain2.com If 300(a)domain1.com calls 201 they get sent to 201(a)domain2.com Is this a bug? I tried updated to latest head but there seems to be an issue with that that even when people are registered as per normal, calling them internally or routing to them from a did reports user_not_registered and they just get voicemail, doesn't affect outbound though.. I did have time to confirm the issue is still in latest head though From: Mr Nathan Downes [mailto:nathandownes@hotmail.com] Sent: Wednesday, 1 August 2012 11:04 PM To: 'FreeSWITCH Users Help' Subject: FW: multi-tenant setup - call goes to wrong place Hi, I ended up locating the problem.. it appears when I was playing with shared_presence.. to try and get BLF working for certain tenants. What I thought would help from googling caused this issue somehow, below is what I enabled to cause the problem. In internal.xml  <param name="manage-presence" value="false"/> this one true   this one enabled  this one true + enabled    this one + enabled <param name="presence-hosts" value="domain1.com,domain2.com,domain3.com,$${local_ip_v4}"/> Is this actually required to use BLF for a certain domain? From: Mr Nathan Downes [mailto:nathandownes@hotmail.com] Sent: Friday, 27 July 2012 9:17 PM To: 'FreeSWITCH Users Help' Subject: multi-tenant setup - call goes to wrong place Hi, I have a multi tenant setup with fusionpbx, for one domain I use extensions that don't register just for voicemail i.e 201(a)domain1.com , when someone from xx(a)domain1.com calls 201 everything appears to go correctly, domain is set and call goes to 201(a)domain1.com but when it first becomes an ip it sends to 201(a)domain2.ip.address. It doesn't seem to happen with all extensions numbered the same in different domains.. Any idea what to look for?? Log as follows. 2012-07-26 09:37:35.178096 [NOTICE] switch_channel.c:926 New Channel sofia/internal/306(a)domain1.com [b62db898-d6b1-11e1-be15-73f389528c7f] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306(a)domain1.com) Running State Change CS_NEW 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/306(a)domain1.com) State NEW 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5838 Channel sofia/internal/306(a)domain1.com entering state [received][100] 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5849 Remote SDP: v=0 o=- 60255882 60255882 IN IP4 115.64.93.203 s=- c=IN IP4 115.64.93.203 t=0 0 m=audio 55938 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/306(a)domain1.com PCMA/8000 20 ms 160 samples 64000 bits 2012-07-26 09:37:35.178096 [DEBUG] switch_core_codec.c:111 sofia/internal/306(a)domain1.com Original read codec set to PCMA:8 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5155 Set 2833 dtmf send/recv payload to 101 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:6077 (sofia/internal/306(a)domain1.com) State Change CS_NEW -> CS_INIT 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306(a)domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306(a)domain1.com) Running State Change CS_INIT 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/306(a)domain1.com) State INIT 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:85 sofia/internal/306(a)domain1.com SOFIA INIT 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:125 (sofia/internal/306(a)domain1.com) State Change CS_INIT -> CS_ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306(a)domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/306(a)domain1.com) State INIT going to sleep 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306(a)domain1.com) Running State Change CS_ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1919 (sofia/internal/306(a)domain1.com) Callstate Change DOWN -> RINGING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/306(a)domain1.com) State ROUTING 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:148 sofia/internal/306(a)domain1.com SOFIA ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:104 sofia/internal/306(a)domain1.com Standard ROUTING 2012-07-26 09:37:35.178096 [INFO] mod_dialplan_xml.c:485 Processing 0286249343 <306>->201 in context domain1.com Dialplan: sofia/internal/306(a)domain1.com parsing [domain1.com->unloop] continue=false Dialplan: sofia/internal/306(a)domain1.com parsing [domain1.com->Local_Extension] continue=false Dialplan: sofia/internal/306(a)domain1.com Regex (PASS) [Local_Extension] destination_number(201) =~ /(^\d{2,7}$)/ break=on-false Dialplan: sofia/internal/306(a)domain1.com Action set(dialed_extension=201) Dialplan: sofia/internal/306(a)domain1.com Action export(dialed_extension=201) Dialplan: sofia/internal/306(a)domain1.com Action limit(hash ${domain_name} 201 ${limit_max} ${limit_destination}) Dialplan: sofia/internal/306(a)domain1.com Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/306(a)domain1.com Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s trftime(%b)}/${strftime(%d)}/${uuid}.wav) Dialplan: sofia/internal/306(a)domain1.com Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/306(a)domain1.com Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/306(a)domain1.com Action set(ringback=${us-ring}) Dialplan: sofia/internal/306(a)domain1.com Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/306(a)domain1.com Action set(call_timeout=30) Dialplan: sofia/internal/306(a)domain1.com Action set(hangup_after_bridge=true) Dialplan: sofia/internal/306(a)domain1.com Action set(continue_on_fail=true) Dialplan: sofia/internal/306(a)domain1.com Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/306(a)domain1.com Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/306(a)domain1.com Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/306(a)domain1.com Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/306(a)domain1.com Action bridge(user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}) Dialplan: sofia/internal/306(a)domain1.com Action answer() Dialplan: sofia/internal/306(a)domain1.com Action sleep(1000) Dialplan: sofia/internal/306(a)domain1.com Action voicemail(default ${domain_name} ${dialed_extension}) 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/306(a)domain1.com) State Change CS_ROUTING -> CS_EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306(a)domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/306(a)domain1.com) State ROUTING going to sleep 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306(a)domain1.com) Running State Change CS_EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/306(a)domain1.com) State EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:241 sofia/internal/306(a)domain1.com SOFIA EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:196 sofia/internal/306(a)domain1.com Standard EXECUTE EXECUTE sofia/internal/306(a)domain1.com set(call_direction=local) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [call_direction]=[local] EXECUTE sofia/internal/306(a)domain1.com set(open=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [open]=[true] EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-spymap/306/b62db898-d6b1-11e1-be15-73f389528c7f) EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-last_dial/306/201) EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-last_dial/global/b62db898-d6b1-11e1-be15-73f389528c7 f) EXECUTE sofia/internal/306(a)domain1.com set(RFC2822_DATE=Thu, 26 Jul 2012 09:37:35 +1000) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [RFC2822_DATE]=[Thu, 26 Jul 2012 09:37:35 +1000] EXECUTE sofia/internal/306(a)domain1.com set(dialed_extension=201) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [dialed_extension]=[201] EXECUTE sofia/internal/306(a)domain1.com export(dialed_extension=201) 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [dialed_extension]=[201] EXECUTE sofia/internal/306(a)domain1.com limit(hash domain1.com 201 2 !BUSY) 2012-07-26 09:37:35.178096 [INFO] switch_limit.c:126 incr called: domain1.com_201 max:2, interval:0 2012-07-26 09:37:35.178096 [INFO] mod_hash.c:202 Usage for domain1.com_201 is now 1/2 EXECUTE sofia/internal/306(a)domain1.com bind_meta_app(1 b s execute_extension::dx XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/306(a)domain1.com bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89 8-d6b1-11e1-be15-73f389528c7f.wav) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89 8-d6b1-11e1-be15-73f389528c7f.wav EXECUTE sofia/internal/306(a)domain1.com bind_meta_app(3 b s execute_extension::cf XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/306(a)domain1.com bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/306(a)domain1.com set(ringback=%(2000, 4000, 440.0, 480.0)) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [ringback]=[%(2000, 4000, 440.0, 480.0)] EXECUTE sofia/internal/306(a)domain1.com set(transfer_ringback=local_stream://moh) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/306(a)domain1.com set(call_timeout=30) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [call_timeout]=[30] EXECUTE sofia/internal/306(a)domain1.com set(hangup_after_bridge=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/306(a)domain1.com set(continue_on_fail=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [continue_on_fail]=[true] EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-call_return/201/306) EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-last_dial_ext/201/b62db898-d6b1-11e1-be15-73f389528c 7f) EXECUTE sofia/internal/306(a)domain1.com set(called_party_callgroup=) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306(a)domain1.com SET [called_party_callgroup]=[UNDEF] EXECUTE sofia/internal/306(a)domain1.com hash(insert/domain1.com-last_dial//b62db898-d6b1-11e1-be15-73f389528c7f) EXECUTE sofia/internal/306(a)domain1.com bridge(user/201(a)domain1.com) 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047 sofia/internal/306(a)domain1.com EXPORTING[export_vars] [domain_name]=[domain1.com] to event 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047 sofia/internal/306(a)domain1.com EXPORTING[export_vars] [dialed_extension]=[201] to event 2012-07-26 09:37:35.178096 [DEBUG] switch_ivr_originate.c:1958 Parsing global variables 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047 sofia/internal/306(a)domain1.com EXPORTING[export_vars] [domain_name]=[domain1.com] to event 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047 sofia/internal/306(a)domain1.com EXPORTING[export_vars] [dialed_extension]=[201] to event 2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:1958 Parsing global variables 2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable [sip_invite_domain]=[domain1.com] 2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable [presence_id]=[201(a)domain1.com] 2012-07-26 09:37:35.198097 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:201@110.143.31.101:1178 [b630218c-d6b1-11e1-be1d-73f389528c7f] 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:4734 (sofia/internal/sip:201@110.143.31.101:1178) State Change CS_NEW -> CS_INIT 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201@110.143.31.101:1178) Running State Change CS_INIT 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:201@110.143.31.101:1178) State INIT 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:85 sofia/internal/sip:201@110.143.31.101:1178 SOFIA INIT 2012-07-26 09:37:35.198097 [DEBUG] sofia_glue.c:2602 Local SDP: v=0 o=FreeSWITCH 1343234313 1343234314 IN IP4 203.174.163.226 s=FreeSWITCH c=IN IP4 203.174.163.226 t=0 0 m=audio 25142 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:201@110.143.31.101:1178) State Change CS_INIT -> CS_ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:201@110.143.31.101:1178) State INIT going to sleep 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201@110.143.31.101:1178) Running State Change CS_ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1919 (sofia/internal/sip:201@110.143.31.101:1178) Callstate Change DOWN -> RINGING 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/sip:201@110.143.31.101:1178) State ROUTING 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:148 sofia/internal/sip:201@110.143.31.101:1178 SOFIA ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:201@110.143.31.101:1178) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/sip:201@110.143.31.101:1178) State ROUTING going to sleep 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201@110.143.31.101:1178) Running State Change CS_CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:201@110.143.31.101:1178) State CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:201@110.143.31.101:1178) State CONSUME_MEDIA going to sleep 2012-07-26 09:37:35.198097 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:201@110.143.31.101:1178 entering state [calling][0] 2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201@110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.338097 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:201@110.143.31.101:1178 entering state [proceeding][180] 2012-07-26 09:37:35.338097 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:201@110.143.31.101:1178! 2012-07-26 09:37:35.338097 [INFO] switch_ivr_originate.c:1156 Sending early media 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/306(a)domain1.com] 203.174.163.226 port 26698 -> 115.64.93.203 port 55938 codec: 8 ms: 20 2012-07-26 09:37:35.338097 [DEBUG] switch_rtp.c:1680 Starting timer [soft] 160 bytes per 20ms 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 60ms (3 frames) 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3566 sofia/internal/306(a)domain1.com Set rtp dtmf delay to 40 2012-07-26 09:37:35.338097 [DEBUG] mod_sofia.c:2606 Ring SDP: v=0 o=FreeSWITCH 1343232757 1343232758 IN IP4 203.174.163.226 s=FreeSWITCH c=IN IP4 203.174.163.226 t=0 0 m=audio 26698 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-07-26 09:37:35.338097 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/306(a)domain1.com! 2012-07-26 09:37:35.338097 [DEBUG] switch_channel.c:3042 (sofia/internal/306(a)domain1.com) Callstate Change RINGING -> EARLY 2012-07-26 09:37:35.338097 [DEBUG] switch_core_session.c:777 Send signal sofia/internal/306(a)domain1.com [BREAK]

1 0

No call_uuid in ringing state unix odbc for PostgreSQL
by cogs66 01 Aug '12

01 Aug '12

Hello all I use ODBC in the core with PostgreSQL and have an intercept application that looks for the call_uuid when in a ringing state. The issue i have is that only 'null' is displayed not the uuid so the application fails. Once the call has been answered the call_uuid appears as it should. I have tested on other FreeSWITCH installations and everything works great so i know it is to do with this rogue setup. I ran a FreeSWITCH upgrade a few days ago in case this was the issue. Using: Ubuntu 10.04LTS PostgreSQL 9.1 FreeSWITCH Version 1.2.0-rc2+git~20120726T181739Z~696fb9c28b (1.2.0-rc2; git at commit 696fb9c28b on Thu, 26 Jul 2012 18:17:39 Z) I wonder if you are able to shed some light please. Thanks Andy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-call-uuid-in-ringing-state… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 2

Please help - How to use different variables in different bridge groups
by Mario G 01 Aug '12

01 Aug '12

What I need: to have different variable/settings for each target group in a bridge command, and allow leg_delay_start to work. It's related to trying to fix other issues and holding up progress for them. Thanks for any help, Mario G I scoured the wiki and tried many things. Normally, enterprise syntax would solve this, but there is a catch: you can't use leg_delay_start which I need, it is ignored when using enterprise. I found that using brackets [] only applies the variable to the first user in the group, not the whole group so that won't work. What I have now but need to have alert_info moved/apply to the mgt group only: <action application="bridge" data="{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@${domain_name}+A)},${group_call(mgt@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/> This solves variables for groups but breaks leg_delay_start, etc.: <action application="bridge" data="<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:{alert_info=n=${lua_ringtone}${group_call(mgt@${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>

1 0

Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ?
by fieldpeak 01 Aug '12

01 Aug '12

Hi Masters, I'm testing the TLS on FS to work with softphone. followed the wiki (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup ), I generated the CA (root) certificate by below command, however, when i install the root certificate on windows,* it prompt me that the valid period is for only one month*. I tried to change the "DAYS=2190" inside the "gentls_cert" script, but it only effect on server certificate(agent.pem) but not root certificate *(cafile.pem*), Could anyone please help me, appreciated for your any advise!! *./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS:fs.audiocodes.com.cn-org audiocodes.com.cn* * * *below is from wiki. * *This will create CA certificate and key along with in conf/ssl/CA directory(cacert.pem, cakey.pem) and certificate in the conf/ssl folder(cafile.pem).* * [ Note: The name given for -cn and -alt should be the same as the DNS name of your freeswitch installation and used as the registrar name on the phone (at least on Polycoms). ] You can change the "DAYS=2190" line in the gentls_cert file to make the certificate valid for longer time. However making it too long has some wrap around problem, it appears.* *To short, I want to change the valid period longer for the cafile.pem , thanks!!** * -- Regards, Charles

7 11

send_silence_when_idle=10000 Breaks inherit_codec?
by Jerry Richards 01 Aug '12

01 Aug '12

Hello All, We want to configure Freeswitch with the following tags, but there are undesired side-effects. In conf/vars.xml: <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=10000"/> In conf/sip_profiles/internal.xml: <param name="inbound-late-negotiation" value="true"/> In conf/dialplan/default.xml: <param name="inherit_codec" value="true"/> The problem is that setting send_silence_when_idle in vars.xml (above) causes ringback_data to get set in switch_ivr_originate.c (around line 2070) which causes 183 Session Progress w/SDP which prevents inherit_codec in internal.xml (above) to work as expected. This looks like a bug? Thanks, Jerry

2 1