- FreeSWITCH-users - FreeSWITCH Mailing Lists

Call drops after 30 seconds
by Matthew Ralston 14 Jul '11

14 Jul '11

Hi, I'm having a problem at the moment with calls being successfully set up, with two-way audio, being terminated by FreeSWITCH after 30 seconds. Internal calls (i.e. between SIP phones on the same LAN segment as the FreeSWITCH box) work flawlessly. The problem arises when at least one of the handsets is located elsewhere on the Internet. This behaviour is exhibited under the following circumstances: - A-leg only call, e.g. to voicemail when the handset is at another location on the Internet - A-leg-B-leg call if one or both of the handsets are at another location on the Internet - Inbound calls from our external SIP provider - Outbound calls to our external SIP provider So it is obvious that the problem is related to the SIP going via the Internet, but I'm having trouble understanding why. Whilst debugging this problem I have placed the FreeSWITCH box is in the DMZ on our router, so there should not be any ports blocked. The FreeSWITCH box itself is not running a software firewall. The calls themselves are absolutely fine for the first 30 seconds - each party can hear the other talking fine. The fact that the call is consistently dropped after 30 seconds (give or take a second or two for PDD) suggests that some timeout is being triggered. When FreeSWITCH terminates the call, the following is logged to the console: 2011-06-22 13:33:50.514941 [DEBUG] sofia.c:4787 Channel sofia/internal/1006(a)public.ip.removed entering state [terminating][0] 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2641 (sofia/internal/1006(a)public.ip.removed) Callstate Change ACTIVE -> HANGUP 2011-06-22 13:33:50.514941 [NOTICE] sofia.c:5508 Hangup sofia/internal/1006(a)public.ip.removed [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2011-06-22 13:33:50.514941 [DEBUG] switch_channel.c:2657 Send signal sofia/internal/1006(a)public.ip.removed [KILL] 2011-06-22 13:33:50.514941 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.534966 [DEBUG] switch_ivr_play_say.c:1649 done playing file 2011-06-22 13:33:50.625988 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-press.wav] (en:en) 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:2063 sofia/internal/1006(a)public.ip.removed skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1006(a)public.ip.removed) State EXECUTE going to sleep 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_HANGUP 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006(a)public.ip.removed) State HANGUP 2011-06-22 13:33:50.727027 [DEBUG] mod_sofia.c:458 Channel sofia/internal/1006(a)public.ip.removed hanging up, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1006(a)public.ip.removed Standard HANGUP, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1006(a)public.ip.removed) State HANGUP going to sleep 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1006(a)public.ip.removed) State Change CS_HANGUP -> CS_REPORTING 2011-06-22 13:33:50.727027 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_REPORTING 2011-06-22 13:33:50.727027 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006(a)public.ip.removed) State REPORTING 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1006(a)public.ip.removed Standard REPORTING, cause: NORMAL_UNSPECIFIED 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1006(a)public.ip.removed) State REPORTING going to sleep 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1006(a)public.ip.removed) State Change CS_REPORTING -> CS_DESTROY 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1118 Send signal sofia/internal/1006(a)public.ip.removed [BREAK] 2011-06-22 13:33:50.740064 [DEBUG] switch_core_session.c:1290 Session 5 (sofia/internal/1006(a)public.ip.removed) Locked, Waiting on external entities 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1308 Session 5 (sofia/internal/1006(a)public.ip.removed) Ended 2011-06-22 13:33:50.740064 [NOTICE] switch_core_session.c:1310 Close Channel sofia/internal/1006(a)public.ip.removed [CS_DESTROY] 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1006(a)public.ip.removed) Callstate Change HANGUP -> DOWN 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1006(a)public.ip.removed) Running State Change CS_DESTROY 2011-06-22 13:33:50.740064 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006(a)public.ip.removed) State DESTROY 2011-06-22 13:33:50.740064 [DEBUG] mod_sofia.c:363 sofia/internal/1006(a)public.ip.removed SOFIA DESTROY 2011-06-22 13:33:50.780056 [DEBUG] switch_nat.c:570 unmapped public port 31484 protocol UDP to localport 31484 2011-06-22 13:33:50.840070 [DEBUG] switch_nat.c:570 unmapped public port 31485 protocol UDP to localport 31485 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1006(a)public.ip.removed Standard DESTROY 2011-06-22 13:33:50.840070 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1006(a)public.ip.removed) State DESTROY going to sleep The above example was from an externally situated SIP phone ringing voicemail (4000) on FreeSWITCH. I have experimented changing various timers and timeouts in the config of FreeSWITCH (one at a time, being careful to put them back afterwards!) but been unable to resolve the issue. Incidentally, we have no long term intention of running off-site SIP phones with the PBX and I'm hoping not to have to leave it in the DMZ either, it's just like that for debugging. What is a real issue is the calls to our external SIP provider (i.e. outbound calls) being dropped. Any suggestions would be greatly appreciated. Thanks, Matthew Ralston Web Developer & IT Consultant matt(a)mralston.co.uk www.mralston.com

6 8

DISA, originate & bridge
by Boris Kovalenko 14 Jul '11

14 Jul '11

Hello! I'm trying to modify DISA javascript example to use 'originate' instead of 'transfer'. So the problem is - after originated call is answered I see no voice. Thinking this is becase call is not bridged to the current session. How may I do this? My peace of code: osession = new Session("{origination_caller_id_number=7343523000 1,v_ats_dstport=50002}sofia/epbx/89979049898XXX@X.X.X.42:5061"); while(osession.ready()) { // Don't know what to write here } -- Regards, Boris

1 0

How to configure for FS playing voice prompt in case the called extension is in a call.
by fieldpeak 14 Jul '11

14 Jul '11

Hi Gurus, Could anyone advise how to realize have FS to play a voice prompt e.g. 'the extension you dialed is busy now, please dial the other exsention' to replacing busy tone when the called extension is in a call... the details is below, When calling to FS, the FS will play IVR "welcome to call us, please input the extension number, for operator please press 0", then the caller input the extension number, in case the extension is in a call, the caller will hear busy tone... it needs the system play "the extension you dailed is busy, please dial the other extension", currently my dial plan as following, could anyone advise how to change based on below dial plan or any other new dail plan can realize it... Thanks a lot! <extension name="Local_Extension"> <condition field="destination_number" expression="^(1\d{3})$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=false"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="bridge" data="sofia/internal/${dialed_extension}%${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> <extension name="IVR" > <condition field="destination_number" expression="^888$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="ivr" data="IVR"/> </condition> </extension>  <menu name="IVR" greet-long="C:/VSWITCH/recordings/greeting_tts.wav" greet-short="C:/VSWITCH/recordings/greeting_tts.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="5"> <entry action="menu-exec-app" digits="/(^\*\d{3,5}$|^\d{3,5}$)/" param="transfer $1 XML default"/> </menu> Regards, Charles

3 6

SPA-2102 ring issue
by Joseph L. Casale 13 Jul '11

13 Jul '11

I have an SPA-2102 a client uses that I can't replace with something good. It appears to ring for several seconds after another user answers the incoming call from some other sip phone. The SPA-3102 appears to have some params that look applicable to tune this but those don't appear in this unit. I don't have an easy way to setup a syslog server remotely, so short of driving out to this place, does anyone know off hand what may cause this so I may give it a try? thanks, jlc

1 0

How to set the member-flags in mod_conference
by abubacker 13 Jul '11

13 Jul '11

Dear all, My requirement is to organise a conference where I want some of the users to be mute , and some of the members to be deaf and some of the member to be endconf I was trying to set the variable "member-flags" as mute or deaf or endconf but its not working whether I am doing the right thing ! Dialplan : <extension name="Conference"> <condition field="destination_number" expression="^3001$"> <action application="answer" /> <action application="set" data="member-flags=mute"/> <action application="conference" data="room1"/> </condition> </extension> <extension name="Conference"> <condition field="destination_number" expression="^3002$"> <action application="answer" /> <action application="set" data="member-flags=endconf"/> <action application="conference" data="room1"/> </condition> </extension> -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

2 2

FreeSWITCH Conference Call Today
by Michael Collins 13 Jul '11

13 Jul '11

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_13 We have a few things to discuss as a community, plus a cool video to share. We are especially interested in your thoughts on the subject of FreeSWITCH "config sets" like what's here: http://svn.freeswitch.org/svn/configs/ Please be ready to share your config ideas with the group. Talk to you soon! -Michael

1 0

Configuration Server
by Roger Castaldo 13 Jul '11

13 Jul '11

Hi everyone, I have been following freeswitch for quite some time now and working on an ever evolving configuration server for it, which has now taken yet another branch, that instead of writing xml configuration files now is attempting to control the call flow itself using the outbound socket. Currently it is a closed source project, originally because of some of my closed source libraries I was using to develop it, but since they have become open source I was wondering if there is any interest in working with me on the development/testing of the product. It is written in C# and is entirely web based, using ajax calls and web services for everything. I am using a Relational Mapping library of my own design to map the data to the database (currently a firebirdsql database).

4 7

WIKI and command printout explanation
by Philippe Le Toquin 13 Jul '11

13 Jul '11

Hello, I am trying to get more familiar wit FS cli and found the page that list the commands and their usage. What I am looking for is an explanation of their printout. Some of them are easy to understand but other not so (not to me at least) Is there a wiki page that explains For example the simple status command give me that: status UP 0 years, 29 days, 17 hours, 38 minutes, 3 seconds, 716 milliseconds, 375 microseconds 512 session(s) since startup 0 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 I had 512 session since startup.....fair enough but then it says "1000 session(s) max" What does it means? that FS will stop working after 10000 sessions or that I can't get more than 1000 in parallel? Regards /Philippe

4 4

Does anybody implemented the FreeSWITHC as an SBC?
by Aram Ter-Martirosyan 13 Jul '11

13 Jul '11

Does anybody implemented the FreeSWITHC as an SBC? I already have our SIP switch and it does almost all the telephony functionalities, including registration, call handling, and different call features like hunting, call forwarding, etc. I want the SBC to translate the SIP registrations and the INVITEs to our existing SIP registrar coming in from the remote VoIP users. I also want to create redundant call and registration handling by means of several SBCs, maybe having one Outbound server and several SBCs behind it, so when one is down the others can handle the same traffic. Is this possible with FreeSWITCH? sbc's - can be instructed to rewrite headers by rules or scripts. they also have different security functions and abilities for remote user NAT traversal, some can also transcode. Thanks in advance, Aram Ter-Martirosyan ConnectTo Communications, Inc. <http://www.connectto.com/> http://www.ConnectTo.com 555 Riverdale, Suite A Glendale, CA 91204 aram(a)ConnectTo.com tel 818.546.4601 fax 818.546.4617 Turning Technology Into Business Solutions

2 1

An easy way to test a list of numbers in condition?
by Mario G 13 Jul '11

13 Jul '11

I am trying to find an easy way to test multiple numbers in a condition without putting all the numbers on one line. There can be 25 numbers or more. Preferably have comments. I looked at many web sites on XML, etc. but could not find a way to continue lines. Freeswitch treats everything in the expression including all spaces as part of the expression. Anyone know of a good way to do this? Thanks, Mario G <extension name="out.group1"> <condition field="${destination_number_w}" expression="^1111111111|  2222222222|3333333333|  4444444444|5555555555  $"> <action application="set" data="destination_number_w=1$1"/> <action application="bridge" data="sofia/gateway/group1/1$1"/> </condition> </extension>

3 5