January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Connect three parties together using bridge?
by Ali Pey 17 Feb '14

17 Feb '14

Hello, Is it possible to connect three parties together using bridge or uuid_bridge? Thanks, Ali Pey

5 8

Customized CDR
by Abdullah 11 Feb '14

11 Feb '14

Hi all genius Developer's, How i get free-switch , Answer-time ,billsec and duration variables in Perl script with out using API hangup hook and using other Module of Freeswitch (CVS,XML-CDR etc). i have try to get Global and local variable like this but not getting any value after hangup call. $billsec = $session->getVariable("billsec"); but when i get Dial status variable value return like SUCCESS But why i cant get Billsec using this ?? Suggestion are Welcome.... Thanks Abdullah -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Customized-CDR-tp7596058.html Sent from the freeswitch-users mailing list archive at Nabble.com.

3 2

FreeSwitch Proxy + RTPProxy Media server
by adahary 09 Feb '14

09 Feb '14

I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. <action application="set"> </action> <action application="set"> </action> Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Med… Sent from the freeswitch-users mailing list archive at Nabble.com.

5 16

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

3 9

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

3 3

1.4/master openssl requirement change.
by Michael Jerris 05 Feb '14

05 Feb '14

Please note, I just pushed a change that now requires openssl version 1.0.1e or later to build. I also added for osx to automatically look in the homebrew location for installing that library, as you will likely need to install homebrew openssl to build now. Please let me know if you see any issues. Mike

11 43

codec configuration ignored by freeswitch
by Giuseppe Cardone 05 Feb '14

05 Feb '14

Hello, I have tried to configure a limited set of two codecs for outbound calls. I first tried with the parameters described in the documentation in a sip_profile and then in a dialplan and then in vars.xml. However, each time the set of codecs seems to be overwritten and the original list offered by the calling UA is applied. I am new to FS and suspect I am missing something quite simple, can anyone lend a hand in suggesting what would cause the described behaviour ? Here is the essential part of the log showing the absolute_codec_string first applied: EXECUTE sofia/internal/390742850020@217.72.102.128:5070 set(absolute_codec_string=PCMA,G729) 2014-01-29 17:17:40.290655 [DEBUG] mod_dptools.c:1402 sofia/internal/390742850020@217.72.102.128:5070 SET [absolute_codec_string]=[PCMA,G729] EXECUTE sofia/internal/390742850020@217.72.102.128:5070 bridge(sofia/gateway/unidata_dev/+390144750000) 2014-01-29 17:17:40.290655 [DEBUG] switch_channel.c:1198 sofia/internal/390742850020@217.72.102.128:5070 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 29 Jan 2014 17:17:40 +0100] to event 2014-01-29 17:17:40.290655 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2014-01-29 17:17:40.290655 [NOTICE] switch_channel.c:1052 New Channel sofia/external/+390144750000 [e00a6dde-8900-11e3-8b08-0b2b96c9cf60] 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5231 (sofia/external/+390144750000) State Change CS_NEW -> CS_INIT 2014-01-29 17:17:40.290655 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/+390144750000 [BREAK] and then overwritten: 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5301 [zrtp_passthru] Setting a-leg inherit_codec=true 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5304 [zrtp_passthru] Setting b-leg absolute_codec_string='G722@8000h@20i@64000b,PCMU@8000h@20i@64000b,PCMA@8000h@20i@64000b,GSM@8000h@20i@13200b' 2014-01-29 17:17:40.290655 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+390144750000) Running State Change CS_INIT 2014-01-29 17:17:40.290655 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+390144750000) State INIT 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:87 sofia/external/+390144750000 SOFIA INIT 2014-01-29 17:17:40.290655 [DEBUG] sofia_glue.c:2739 Local SDP: Thanks in advance for your help. Regards, Giuseppe

2 3

Using park_timeout with uuid_park
by Siri MM 04 Feb '14

04 Feb '14

Hi, I have a lua script which parks a leg using uuid_park I would like to restrict the amount of time that the leg is parked. In order to achieve this, I tried using park_timeout as follows: cmd = "uuid_setvar "..uuid.." park_timeout 10"; api:executeString(cmd); api:execute("uuid_park", uuid); However, the parked call doesn't seem to timeout. Am I missing something? Thanks!

2 9

Re: [Freeswitch-users] Play IVR Before Answer
by Shahzad Bhatti 01 Feb '14

01 Feb '14

i observe on my phone that this is possible as when i on my mobile when bill is not paid and outgoing is blocked then when try to call some number then Message is played and call hangup. just like above Scenario i want to play message using IVR or may with some other application to tell the caller what is his current credit on leg A and then bridge call to leg B without answer. any help will be appreciated you can get the log of cli here http://pastebin.freeswitch.org/21912 and example of dialplan here http://pastebin.freeswitch.org/21913 Regards Shahzad Bhatti > > ---------- Forwarded message ---------- > From: Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 29 Jan 2014 18:13:34 +0500 > Subject: Re: [Freeswitch-users] Play IVR Before Answer > Thanks Mario G for your reply i worked through it using ringback and play > file using local stream and find that freeswitch send early media but > unable to listen on leg A so i remove the line of answer and play a sample > ivr for test and also remove bridge the call to leg B for testing and found > that during early media state ivr is playing and log show that it plays but > on leg A i am unable to listen anything from that ivr but a ring tone and > when the ivr completed the channel state is returned to hangup > so i think i am near but missing some thing in configuration so anyone can > reply that what i do to listen ivr before answer through early media > here is my new log > http://pastebin.freeswitch.org/21912 > and here is my dialplan for the above call > http://pastebin.freeswitch.org/21913 > Regards > Shahzad Bhatti > > > ---------- Forwarded message ---------- >> From: Mario G <mario_fs(a)mgtech.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Tue, 28 Jan 2014 13:08:24 -0800 >> Subject: Re: [Freeswitch-users] Play IVR Before Answer >> I don't know how to do IVR this way but... this may help you get started: >> What you want is to dynamically create early media. Our calls use early >> media to say hello "name" to the person and change instructions and the >> music in early media. This is done by setting the "ringback" variable. >> Before the bridge I call a LUA program that looks up the incoming number >> and creates the early media string and sets the "ringback" variable before >> returning (it does many other things as well). If all you want to do when >> someone calls in is to say "Hi Mario, you have a credit of 34.64", it can >> be done with a little work. This is NOT IVR though. >> >> Before messing with IVR note this IMPORTANT FACT: Most (all?) ITSPs limit >> how long they will ring (send early media) before doing something else, in >> my case the max I can set ringing to is 60 seconds before they send to VM, >> etc on their end. That means I am limited to early media of 60 seconds >> which my LUA program dynamically calculates and ensures we don't go past 55 >> seconds. >> Good luck >> Mario G >> >> On Jan 28, 2014, at 8:26 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> >> wrote: >> >> Hello i want to play an ivr using phrase to say the caller credit, i do >>> this after answer and it is working i want to know that is it also possible >>> to play IVR before answer using execute_on_preanswer like >>> >> *<action application="set" data="execute_on_preanswer=phrase >> saycredit,34:64:97" />* >> here is my xml_curl xml for the call >> http://pastebin.freeswitch.org/21909 >> i want to know is it possible and if possible then what is the correct >> way i want to play IVR on legA but before answering the leg >> >> Regards >> Shahzad Bhatti >> >>

3 2

Outbound call not reaching destination - target on TLS
by Michal Borkowski - MORCOM 01 Feb '14

01 Feb '14

Hi, I manage to solve most of the issues with FreeSwitch. One main left only..... I have mobile phone which is using CSipSimple All on FreeSwitch set up and I can register the mobile with my extension. I can call all destinations from the mobile (This is using TLS with all security settings) but I can not call that mobile. I was testing various codec sets in CSipSimple - but nothing helped. There must be something more. I'm getting the error message that destination is incompatible. Maybe looking to the log files sb would know the source of issue? Sorry for long log.... ======= => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_state_machine.c:679 (sofia/internal/sip:michalb@62.140.132.18:4203) State HANGUP going to sleep => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:michalb@62.140.132.18:4203 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION => 2014-01-31 18:12:41.230255 [DEBUG] mod_sofia.c:506 Channel sofia/internal/sip:michalb@62.140.132.18:4203 hanging up, cause: INCOMPATIBLE_DESTINATION => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_state_machine.c:679 (sofia/internal/sip:michalb@62.140.132.18:4203) State HANGUP => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:michalb@62.140.132.18:4203) Running State Change CS_HANGUP => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:41.230255 [DEBUG] switch_channel.c:3184 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [KILL] => 2014-01-31 18:12:41.230255 [NOTICE] sofia.c:6629 Hangup sofia/internal/ sip:michalb@62.140.132.18:4203 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] => 2014-01-31 18:12:41.230255 [DEBUG] sofia.c:5785 Channel sofia/internal/ sip:michalb@62.140.132.18:4203 entering state [terminated][488] => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:41.230255 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/sip:michalb@62.140.132.18:4203) State CONSUME_MEDIA going to sleep => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/sip:michalb@62.140.132.18:4203) State CONSUME_MEDIA => 2014-01-31 18:12:38.600281 [DEBUG] sofia.c:5785 Channel sofia/internal/ sip:michalb@62.140.132.18:4203 entering state [calling][0] => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:michalb@62.140.132.18:4203) Running State Change CS_CONSUME_MEDIA => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/sip:michalb@62.140.132.18:4203) State ROUTING going to sleep => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:38.600281 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:michalb@62.140.132.18:4203) State Change CS_ROUTING -> CS_CONSUME_MEDIA => 2014-01-31 18:12:38.600281 [DEBUG] mod_sofia.c:150 sofia/internal/ sip:michalb@62.140.132.18:4203 SOFIA ROUTING => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/sip:michalb@62.140.132.18:4203) State ROUTING => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:michalb@62.140.132.18:4203) Running State Change CS_ROUTING => 2014-01-31 18:12:38.600281 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/sip:michalb@62.140.132.18:4203) State INIT going to sleep => 2014-01-31 18:12:38.590255 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:38.590255 [DEBUG] mod_sofia.c:127 (sofia/internal/ sip:michalb@62.140.132.18:4203) State Change CS_INIT -> CS_ROUTING => => a=sendrecv => a=ptime:20 => a=fmtp:101 0-16 => a=rtpmap:101 telephone-event/8000 => a=fmtp:100 bitrate=48000 => a=rtpmap:100 G7221/32000 => a=fmtp:99 mode=30 => a=rtpmap:99 iLBC/8000 => a=rtpmap:98 SPEEX/8000 => m=audio 28466 RTP/AVP 8 0 98 99 3 100 9 101 13 => t=0 0 => c=IN IP4 XXX.XXX.XXX.XXX => s=FreeSWITCH => o=FreeSWITCH 1391159892 1391159893 IN IP4 XXX.XXX.XXX.XXX => v=0 => Local SDP: => 2014-01-31 18:12:38.590255 [DEBUG] sofia_glue.c:2738 sofia/internal/ sip:michalb@62.140.132.18:4203 sending invite version: 1.2.18 git f6cd294 2014-01-29 16:39:03Z 32bit => 2014-01-31 18:12:38.590255 [DEBUG] sofia_glue.c:2710 sip:michalb@62.140.132.18:4203;transport=TLS;ob Setting proxy route to sofia/internal/sip:michalb@62.140.132.18:4203 2014-01-31 18:12:38.590255 [DEBUG] sofia_glue.c:1093 STUN Success [XXX.XXX.XXX.XXX]:[28466] 2014-01-31 18:12:38.260275 [INFO] switch_nat.c:590 NAT port mapping disabled 2014-01-31 18:12:38.260275 [INFO] switch_nat.c:590 NAT port mapping disabled => 2014-01-31 18:12:38.260275 [DEBUG] mod_sofia.c:87 sofia/internal/ sip:michalb@62.140.132.18:4203 SOFIA INIT => 2014-01-31 18:12:38.260275 [DEBUG] switch_core_state_machine.c:455 (sofia/internal/sip:michalb@62.140.132.18:4203) State INIT => 2014-01-31 18:12:38.260275 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:michalb@62.140.132.18:4203) Running State Change CS_INIT => 2014-01-31 18:12:38.260275 [DEBUG] switch_ivr_originate.c:2782 sofia/internal/sip:michalb@62.140.132.18:4203 Setting leg timeout to 30 => 2014-01-31 18:12:38.260275 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/sip:michalb@62.140.132.18:4203 [BREAK] => 2014-01-31 18:12:38.260275 [DEBUG] mod_sofia.c:5231 (sofia/internal/ sip:michalb@62.140.132.18:4203) State Change CS_NEW -> CS_INIT => 2014-01-31 18:12:38.260275 [NOTICE] switch_channel.c:1052 New Channel sofia/internal/sip:michalb@62.140.132.18:4203 [=>] 2014-01-31 18:12:38.260275 [DEBUG] switch_event.c:1661 Parsing variable [presence_id]=[michalb(a)192.168.1.30] 2014-01-31 18:12:38.260275 [DEBUG] switch_event.c:1661 Parsing variable [leg_timeout]=[30] 2014-01-31 18:12:38.260275 [DEBUG] switch_event.c:1661 Parsing variable [sip_invite_domain]=[192.168.1.30] => 2014-01-31 18:12:38.260275 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [dialed_extension]=[1001] to event => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [origination_callee_id_name]=[1001] to event => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [domain_name]=[192.168.1.30] to event => 2014-01-31 18:12:38.260275 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [dialed_extension]=[1001] to event => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [origination_callee_id_name]=[1001] to event => 2014-01-31 18:12:38.260275 [DEBUG] switch_channel.c:1198 sofia/internal/ lith(a)192.168.1.30 EXPORTING[export_vars] [domain_name]=[192.168.1.30] to event => EXECUTE sofia/internal/lith(a)192.168.1.30 bridge(user/michalb(a)192.168.1.30 ) => EXECUTE sofia/internal/lith(a)192.168.1.30hash(insert/192.168.1.30-last_dial//=>) => 2014-01-31 18:12:38.260275 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [called_party_call_group]=[UNDEF] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(called_party_call_group=) => EXECUTE sofia/internal/lith(a)192.168.1.30hash(insert/192.168.1.30-last_dial_ext/1001/=>) => EXECUTE sofia/internal/lith(a)192.168.1.30hash(insert/192.168.1.30-call_return/1001/lith) => 2014-01-31 18:12:38.250273 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [continue_on_fail]=[true] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(continue_on_fail=true) => 2014-01-31 18:12:38.250273 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [hangup_after_bridge]=[true] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(hangup_after_bridge=true) => 2014-01-31 18:12:38.250273 [INFO] switch_ivr_async.c:3701 Bound B-Leg: *4 execute_extension::att_xfer XML features => EXECUTE sofia/internal/lith(a)192.168.1.30 bind_meta_app(4 b s execute_extension::att_xfer XML features) => 2014-01-31 18:12:38.250273 [INFO] switch_ivr_async.c:3701 Bound B-Leg: *3 execute_extension::cf XML features => EXECUTE sofia/internal/lith(a)192.168.1.30 bind_meta_app(3 b s execute_extension::cf XML features) => 2014-01-31 18:12:38.250273 [INFO] switch_ivr_async.c:3701 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2014/Jan/31/=>.wav => 2014-01-31 18:12:38.250273 [INFO] switch_ivr_async.c:3693 Bound A-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2014/Jan/31/=>.wav => EXECUTE sofia/internal/lith(a)192.168.1.30 bind_meta_app(2 ab s record_session::/usr/local/freeswitch/recordings/archive/2014/Jan/31/=>.wav) => 2014-01-31 18:12:38.250273 [INFO] switch_ivr_async.c:3701 Bound B-Leg: *1 execute_extension::dx XML features => EXECUTE sofia/internal/lith(a)192.168.1.30 bind_meta_app(1 b s execute_extension::dx XML features) => 2014-01-31 18:12:38.250273 [INFO] mod_hash.c:202 Usage for 192.168.1.30_1001 is now 1/5 => 2014-01-31 18:12:38.250273 [INFO] switch_limit.c:126 incr called: 192.168.1.30_1001 max:5, interval:0 => EXECUTE sofia/internal/lith(a)192.168.1.30 limit(hash 192.168.1.30 1001 5 ) => 2014-01-31 18:12:38.250273 [DEBUG] switch_channel.c:1244 EXPORT (export_vars) [dialed_extension]=[1001] => EXECUTE sofia/internal/lith(a)192.168.1.30 export(dialed_extension=1001) => 2014-01-31 18:12:38.250273 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [dialed_extension]=[1001] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(dialed_extension=1001) => 2014-01-31 18:12:38.250273 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [RFC2822_DATE]=[Fri, 31 Jan 2014 18:12:38 +0100] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(RFC2822_DATE=Fri, 31 Jan 2014 18:12:38 +0100) => 2014-01-31 18:12:38.250273 [DEBUG] switch_channel.c:1244 EXPORT (export_vars) [origination_callee_id_name]=[1001] => EXECUTE sofia/internal/lith(a)192.168.1.30export(origination_callee_id_name=1001) => 2014-01-31 18:12:38.250273 [DEBUG] mod_dptools.c:1402 sofia/internal/ lith(a)192.168.1.30 SET [call_direction]=[local] => EXECUTE sofia/internal/lith(a)192.168.1.30 set(call_direction=local) => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_state_machine.c:209 sofia/internal/lith(a)192.168.1.30 Standard EXECUTE => 2014-01-31 18:12:38.250273 [DEBUG] mod_sofia.c:243 sofia/internal/ lith(a)192.168.1.30 SOFIA EXECUTE => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/lith(a)192.168.1.30) State EXECUTE => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/lith(a)192.168.1.30) Running State Change CS_EXECUTE => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/lith(a)192.168.1.30) State ROUTING going to sleep => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/lith(a)192.168.1.30 [BREAK] => 2014-01-31 18:12:38.250273 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/lith(a)192.168.1.30) State Change CS_ROUTING -> CS_EXECUTE ====== And - the remote clients are using TLS/SRTP/ZRTP (only on mobile phones) Any idea why the mobile can not be reached? Thanks in advance.... Michal

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