September 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

MEDIA_TIMEOUT
by roy j 29 Sep '15

29 Sep '15

Hi, all We have some unclear situation. Call flow is below. 1. A --INVITE(SDP)--> FS C 2. A FS --INVITE(SDP)--> C 3. A FS <--183(SDP)-- C 4. A <--183(SDP) FS C 5. A FS <--RTP-- C 6. A <--RTP-- FS C 7. A <--480-- FS C 8. A FS --CANCEL--> C FreeSwitch (FS) sends to 'A' SIP 480 (Reason header: SIP;cause=604;text="MEDIA_TIMEOUT"). There is no RTP from 'A' while progress. Time between 4. and 7. - 30 seconds. Is this expected behaviour or we should update? Current version - FreeSWITCH Version 1.5.6b+git~20131029T184027Z~88a13f1fc2~32bit (git 88a13f1 2013-10-29 18:40:27Z 32bit)

1 0

Getting 503 Service Unavailable while enable record_session
by bhavik patel 29 Sep '15

29 Sep '15

Hi Guys, I am using record_session applications on one of my live server, which have 100 concurrent calls are running. When i enable using dialplan then calls are dropping and getting most of call "503 Service Unavilable", so it means record_session generating load on freeswitch. I am using FreeSWITCH Version 1.4.21+git~20150901T202622Z~a223dd0236~64bit (git a223dd0 2015-09-01 20:26:22Z 64bit). Can any one please suggest me is it bug of Freeswitch ? -- Thanks, Bhavik Patel

12 24

Porting old custom FS module (in C) to latest stable
by Imran Moinuddin 29 Sep '15

29 Sep '15

Hi all, Not sure if this is the right forum to be asking this - but we've inherited a customized Freeswitch module that was developed atop an older version (1.04), and we've been eagerly looking for ways to port this module (naturally in C) to the latest version so that we can get some of our servers off of 1.04 for good. I was curious to know if there's anyone on this list that might be able to help us in performing this code port to the latest stable ? (we'd naturally pay for this assistance) Looking forward to any recommendations. Thanks! Imran

2 1

The FreeSWITCH 1.6.2 release is here!
by Ken Rice 28 Sep '15

28 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1KG5gQs The FreeSWITCH 1.6.2 release is here! The FreeSWITCH 1.6.2 release is here! This is the release of the Video/MCU branch! Release files are located here: Tarball: http://ift.tt/1KG5h6I Packaging: http://ift.tt/1U8FMQN (Please Note the updated location for 1.6 packages) And we’re dropping support in packaging for anything older than Debian 8.0 and anything older than Centos 7.0 due to a number of dependency issues on older platforms. New features that were added: FS-8094 [verto_communicator] Added googEchoCancellation, googNoiseSuppression, and googHighpassFilter settings to the UI enabled by default FS-8107 [switch_ivr_menu] Adding CUSTOM esl events menu::enter and menu::exit when a call enter and exit an ivr menu FS-6833 Allow Freeswitch to initiate late offer calls FS-8075 [mod_hiredis] Add conf file to spec file too and updates for limit release case FS-8053 Handle a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call Improvements in build system, cross platform support, and packaging: FS-7966 [Windows] Build now uses visual studio 2015 and builds successfully, but is still missing video functionality FS-8124 [verto_communicator] Adding option –debug to grunt build, dist app will be generated without minified code. FS-7168 [Debian] Update packages so that FS core libraries are setup as runtime dependencies FS-7697 Chown the /etc/freeswitch/tls directory so that the freeswitch user can have read/write for TLS certificate generation FS-7966 [Windows] Explore use of nuget for wix build dependency FS-7967 [build] SmartOS compatibility FS-8072 [Debian] Fixed a missed space The following bugs were squashed: FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality FS-7988 [filebug.pl] Moved -askall to -terse so old -askall behavior is default and old default is now -terse FS-8029 [jitterbuffer] Fixed robotic sound when using jitterbuffer FS-8103 [mod_rayo] Fixed handling of malformed requests FS-8108 [mod_lua] Removed legacy mod_lua, the regular mod_lua works with system lua now FS-8082 [mod_rayo] Fixed a segfault FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message FS-8116 [verto] Fixed device enumeration hanging on init FS-8118 [verto] Fixed calls not properly rejecting video when video is offered but only audio is accepted FS-7994 [verto_communicator] Using factory for group calls in history FS-8117 [verto_communicator] Calling verto.iceServers upon useSTUN changing on ModalSettings, correctly check for STUN setting in localStorage, and fixed ignoring useSTUN settings FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change FS-8088 [verto_communicator] Fixed members list being destroyed after switching conferences and ending the first conference FS-8130 Port video buffer to also support audio and remove original STFU jitter buffer, add some more resilience to video packet loss, add codec control mechanism for both call-specific debug and codec/call specific parameters, make opus function better in packet loss and latent situations, use new codec control prams to make jitter buffer lookahead FEC optionally enabled or disabled mid-call, and add a parameter to allow jitter buffer lookahead to be enabled. FS-8131 [mod_voicemail] Fixed issues with allowing an empty password change and then locking out the user FS-8136 [mod_h26x] Do not load passthru video codecs by default FS-8140 [mod_sofia] Fixed a user_name typo in sofia_handle_sip_i_invite FS-8142 Fixed a thread cache thread-safety and caching FS-8075 [mod_hiredis] Fix for failover when you pull power on redis, while redis clients under load test FS-8144 [mod_opus] Readability and code formatting cleanup FS-8126 [switch_core] Fixed the pruning of a media bug causing all media bugs on a session to be pruned FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction. FS-8147 [mod_erlang_event] Fixed the process spawning segfault FS-8149 [mod_xml_cdr] Fixed curl dependency in makefile FS-1772 [mod_voicemail] Fixed the reset of voicemail greeting to default to allow entering 0 to restore the default greeting

2 3

The FreeSWITCH 1.4.23 release is here!
by Ken Rice 28 Sep '15

28 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1KG5gzG The FreeSWITCH 1.4.23 release is here! The FreeSWITCH 1.4.23 release is here! This is a routine maintenance release and the resources are located here: Tarballs: http://ift.tt/1KG5gzI Packaging: http://ift.tt/1U8FMQN New features that were added: FS-7752 [mod_rayo] Increase maximum number of elements from 30 to 1024 to allow adhearsion to create large grammars to navigate IVR menus. Improvements in build system, cross platform support, and packaging: FS-8055 [build] Add confdir variable to freeswitch.pc The following bugs were fixed: FS-7135 [mod_sofia] Fixed response to re-invite with duplicate sdp (such as we get from session refresh) when soa is disabled to include an sdp. Fixed t.38 fax failure on session refresh FS-7903 [proxy_media] Fix Codec PROXY Exists but not at the desired implementation. 0hz 0ms 1ch error when using proxy media. FS-8056 [mod_voicemail] Fixed a segfault on vm_inject, regression from FS-7968 FS-7968 [mod_voicemail] Fixed verbose events FS-8110 [mod_rayo] Prompt IQ error reply was being deleted after being sent for delivery. This is incorrect since message delivery thread will clean up the message FS-8082 [mod_rayo] Do not remove items from hash while iterating FS-8103 [mod_rayo] Handle where finishes unexpectedly before start event is received FS-8143 [mod_rayo] Fixed a crash caused by client disconnecting from mod_rayo while a message is being delivered to that client. This is caused by the XMPP context’s JID -> XMPP stream mapping not being cleaned up on XMPP stream destruction. FS-8127 [mod_conference] Fixed an audio issue caused by the codec not updating often enough when detecting rate change FS-8099 [mod_lua] Restored LUA dialplan ACTIONS functionality FS-8142 Fixed thread cache thread-safety and caching FS-8185 [core] Allow xml preprocessor to expand variables where the resulting value is much longer than the original size FS-8167 [mod_lua] Fixed a segfault caused by using api:execute or session:execute and not quoting the first argument like api:execute(log, “Second argument”) instead of api:execute(“log”, “Second argument”) FS-8169 Fixed uuid_displace on stereo channels can lead to memory corruption causing a crash FS-8114 Fixed opus and telephone event payload types colliding on REFER

1 0

FreeSWITCH Week in Review (Master Branch) September 19th-25th
by Ken Rice 28 Sep '15

28 Sep '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1iWQ48x FreeSWITCH Week in Review (Master Branch) September 19th-25th Hello, again. This past week in the FreeSWITCH master branch we had 57 commits! Some of our features for this week are: added reset button to default settings, auto-provision/config support, No Camera option, and splash screen feature in the verto communicator. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8095 [verto_communicator] Added reset button to default settings FS-8095 [verto_communicator] Do not reset name and email upon logout FS-8102 [verto_communicator] Added auto-provision/config support FS-8205 [verto_communicator] Added splash screen feature FS-8193 [verto communicator] Added No Camera option FS-8183 [verto_communicator] Add google API logins, moved google ClientID to config file where it belongs, and tweaks to the CSS FS-8089 [verto_communicator] Opening members list when start conference FS-8204 [verto] Added stereo audio support for opus on FireFox and add sprop-stereo FS-8042 FS-8182 [mod_sofia] Added ping time (in ms) to sip_registrations table, displays as part of the show commands that show registration details, add force_ping=true user var to force options ping on individual registered endpoints FS-8130 Added support for timestamp based counting for jitter buffer in audio mode Improvements in build system, cross platform support, and packaging: FS-8230 FSRTC – calling localStream.stop when it’s available or localStream.active = false to make it work on newer chrome canary FS-7697 [Debian] Moved chown of /etc/freeswitch/tls to postinst FS-7909 [Debian] Removed explicit set of WorkingDirectory FS-7130 [Debian] Use systemd RuntimeDirectory for /run/freeswitch The following bugs were squashed: FS-8196 [verto_communicator] Fixed sidebar layout when user is moderator and added nowrap to white-space FS-8200 [verto_communicator] Ending screenshare if user hangup call and screenshare is now for all instead of only mods FS-7980 [verto_communicator] Fixed video padding-top FS-8224 [verto_communicator] Allowed CallerID to be set from login settings FS-8213 [verto] Added support to skip permission checks FS-8202 [verto] Stopped peer when ending screen share FS-8210 [verto] Fix for module being unloaded while it is in use FS-8190 [mod_event_socket] When using nixevent, freeswitch stops sending us certain custom event that were NOT part of the nixevent command FS-7673 [mod_v8] ODBC NULL value incorrectly evaluated FS-8031 [RTP] Fix for Firefox giving up because we replied to their stun FS-8130 [mod_conference] Add jb_video_low_bitrate for a bit rate to ask for when the jitter buffer is taxed FS-8211 [mod_conference] Fixed video recordings of layouts with overlap having flickering video FS-7911 [mod_conference] Fixed a memory leak FS-8130 Fixed a regression in bridged channels with jitterbuffers FS-8215 Fixed the accuracy of MacOSX nanosleep FS-8216 [mod_av] Fixed for occasional lip sync problems when recording And, this past week in the FreeSWITCH 1.4 branch we had 8 new commits merged in from master. And the FreeSWITCH 1.4.23 release is here!Go check it out! New features that were added: FS-8042 FS-8182 [mod_sofia] Added ping time (in ms) to sip_registrations table, displays as part of the show commands that show registration details, add force_ping=true user var to force options ping on individual registered endpoints Improvements in build system, cross platform support, and packaging: FS-7130 [Debian] Use systemd RuntimeDirectory for /run/freeswitch FS-7909 [Debian] Removed explicit set of WorkingDirectory The following bugs were squashed: FS-8169 Fixed uuid_displace on stereo channels can lead to memory corruption causing a crash FS-8167 [mod_lua] Fixed a segfault caused by using api:execute or session:execute and not quoting the first argument like api:execute(log, “Second argument”) instead of api:execute(“log”, “Second argument”) FS-8185 [core] Allow xml preprocessor to expand variables where the resulting value is much longer than the original size FS-7911 [mod_conference] Fixed a memory leak

1 0

SIP SIMPLE with mod_sms
by Michael Nielsen 28 Sep '15

28 Sep '15

I would like to implement SIP SIMPLE into my FS configuration. For use with Bria client. I've looked into mod_sms and it seems to be the way go. If so, are there a standard example for a dial plan for sending messages between clients? And instead "voicemail" for calls are there a queue functions build in, for if clients are offline?

1 0

make gsmopen Error
by Shabbir abbasi 28 Sep '15

28 Sep '15

Centos 6.7 final Linux localhost.localdomain 3.10.40-1.el6.elrepo.i686 i have downloaded FREESWITCH MASTER branch yesterday, cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build make DEBUG=0 GPIB=0 g++ -c -o release/ctb_lib_portscan.o -DNDEBUG -D_THREAD_SAFE -pthread -O0 -I../include -MTrelease/ctb_lib_portscan.o -MF`echo release/ctb_lib_portscan.o | sed -e 's,\.o$,.d,'` -MD -MP ../src/portscan.cpp In file included from ../include/ctb-0.16/linux/serport.h:12, from ../include/ctb-0.16/serport.h:15, from ../include/ctb-0.16/ctb.h:12, from ../src/portscan.cpp:1: ../include/ctb-0.16/serportx.h: In member function ‘char* ctb::SerialPort_DCS::GetSettings()’: ../include/ctb-0.16/serportx.h:178: error: ‘_snprintf’ was not declared in this scope make: *** [release/ctb_lib_portscan.o] Error 1 any solution ?

3 3

Unable to make calls with gsmopen
by Pete Kay 27 Sep '15

27 Sep '15

Hi I am having problem trying to make outgoing calls using gsmopen. Does anyone know what is wrong? 2015-09-27 15:16:25.057678 [DEBUG] gsmopen_protocol.cpp:2222 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2222 ][gsm01 ][-1,24, 3] sending: AT 2015-09-27 15:16:25.057678 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (A) 2015-09-27 15:16:25.077696 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (T) 2015-09-27 15:16:25.077696 [DEBUG] gsmopen_protocol.cpp:2105 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2105 ][gsm01 ][-1,24, 3] sent (carriage return) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:748 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 748 ][gsm01 ][-1,24, 3] Read line 0: |OK| la_counter=1 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:1684 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 1684 ][gsm01 ][-1,24, 3] got OK 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2309 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2309 ][gsm01 ][-1,24, 3] sending: AT+CBC, expecting: OK 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (A) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (T) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (+) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (C) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (B) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (C) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:2105 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2105 ][gsm01 ][-1,24, 3] sent (carriage return) 2015-09-27 15:16:25.097686 [DEBUG] gsmopen_protocol.cpp:628 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 628 ][gsm01 ][-1,24, 3] TIMEOUT=500 2015-09-27 15:16:25.117703 [DEBUG] gsmopen_protocol.cpp:748 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 748 ][gsm01 ][-1,24, 3] Read line 0: |ERROR| la_counter=1 2015-09-27 15:16:25.117703 [DEBUG] gsmopen_protocol.cpp:1690 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 1690 ][gsm01 ][-1,24, 3] got ERROR 2015-09-27 15:16:25.117703 [WARNING] gsmopen_protocol.cpp:3159 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][WARNINGA 3159 ][gsm01 ][-1,24, 3] AT+CBC didn't get OK from the phone. Continuing. 2015-09-27 15:16:25.117703 [DEBUG] gsmopen_protocol.cpp:2309 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2309 ][gsm01 ][-1,24, 3] sending: AT+CSQ, expecting: OK 2015-09-27 15:16:25.117703 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (A) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (T) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (+) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (C) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (S) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (Q) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:2105 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2105 ][gsm01 ][-1,24, 3] sent (carriage return) 2015-09-27 15:16:25.137704 [DEBUG] gsmopen_protocol.cpp:628 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 628 ][gsm01 ][-1,24, 3] TIMEOUT=500 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:748 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 748 ][gsm01 ][-1,24, 3] Read line 0: |ERROR| la_counter=1 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:1690 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 1690 ][gsm01 ][-1,24, 3] got ERROR 2015-09-27 15:16:25.157687 [WARNING] gsmopen_protocol.cpp:3167 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][WARNINGA 3167 ][gsm01 ][-1,24, 3] AT+CSQ didn't get OK from the phone. Continuing. 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2222 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2222 ][gsm01 ][-1,24, 3] sending: AT+MMGL="HEADER ONLY" 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (A) 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (T) 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (+) 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (M) 2015-09-27 15:16:25.157687 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (M) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (G) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (L) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (=) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (") 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (H) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (E) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (A) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (D) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (E) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (R) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... ( ) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (O) 2015-09-27 15:16:25.177691 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (N) 2015-09-27 15:16:25.197940 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (L) 2015-09-27 15:16:25.197940 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (Y) 2015-09-27 15:16:25.197940 [DEBUG] gsmopen_protocol.cpp:2060 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2060 ][gsm01 ][-1,24, 3] sent data... (") 2015-09-27 15:16:25.197940 [DEBUG] gsmopen_protocol.cpp:2105 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 2105 ][gsm01 ][-1,24, 3] sent (carriage return) 2015-09-27 15:16:25.217702 [DEBUG] gsmopen_protocol.cpp:748 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][DEBUG_GSMOPEN 748 ][gsm01 ][-1,24, 3] Read line 0: |COMMAND NOT SUPPORT| la_counter=1 2015-09-27 15:16:25.317682 [WARNING] gsmopen_protocol.cpp:3186 rev 1.4.18+git~20150312T185523Z~4eed221b69~32bit [(nil)|37 ][WARNINGA 3186 ][gsm01 ][-1,24, 3] AT+MMGL="HEADER ONLY" didn't get OK from the phone. If your phone is not Motorola, please contact the gsmopen developers. Else, if your phone IS a Motorola, probably a long msg was incoming and ther first part was read and then deleted. The second part is now orphan. If you got this warning repeatedly, and you cannot correctly receive SMSs from this interface, please manually clean all messages (and the residual parts of them) from the cellphone/SIM. Continuing. freeswitch@raspberrypi> freeswitch@raspberrypi> gsm list gsm console is NOT yet assigned F ID Name Operator IMEI IB (F/T) OB (F/T) State CallFlw UUID = == ========== ================ =============== ========= ========= ======= =============== ==== 1 gsm01 351553035745952 0/0 0/1 CALL_NOSRVC CALL_DIALING 24c33e2c-652a-11e5-bf8d-53b016c80f16 Total Interfaces: 1 IB Calls(Failed/Total): 0/0 OB Calls(Failed/Total): 0/1

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ICE SDP Params
by Serge Yuriev 27 Sep '15

27 Sep '15

Hello, Endpoints involved: Cisco UCM FreeSWITCH (latest 1.6 GIT) Call flow is rather simple: <param name="enable-3pcc" value="proxy”/> <param name="inbound-late-negotiation" value="true"/> proxy and bypass media commented out Receive <extension name="from_cucm"> <condition field="network_addr" expression="^10\.0\.64\.6[1-6]$"> <action application="pre_answer"/> <action application="transfer" data="${destination_number} XML internal"/> </condition> </extension> And in Internal plan route to carrier: <extension name="mts_tascom"> <condition field="destination_number" expression="^(17[0-5]\d\d)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/mts_tascom-1/$1" /> </condition> </extension> recv 1221 bytes from udp/[10.0.64.63]:6060 at 20:34:55.093189: ------------------------------------------------------------------------ INVITE sip:17000@10.0.64.100:6060 SIP/2.0 Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100> Date: Wed, 16 Sep 2015 17:34:55 GMT Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a(a)10.0.64.63 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CP7942G/9.3.1 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback,X-cisco-original-called Call-Info: <sip:10.0.64.63:6060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 1025134336-0000065536-0000139308-1061158922 P-Charging-Vector: icid-value="3D1A4F00000100000002202B3F40000A";icid-generated-at=msk-vo-s-ucm3;orig-ioi="IMS Inter Operator Identification" Session-Expires: 1800 P-Asserted-Identity: "IT, ????? ??????" <sip:12550@10.0.64.63> Remote-Party-ID: "IT, ????? ??????" <sip:12550@10.0.64.63>;party=calling;screen=yes;privacy=off Contact: <sip:12550@10.0.64.63:6060> Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ send 400 bytes to udp/[10.0.64.63]:6060 at 20:34:55.093460: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100> Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a(a)10.0.64.63 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Content-Length: 0 ------------------------------------------------------------------------ send 1676 bytes to udp/[10.0.64.63]:6060 at 20:34:55.095896: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a(a)10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 2100659333 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 901 v=0 o=FreeSWITCH 1442403045 1442403046 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS no6JiXqEFi18v7d8DbGarc1t7srxqe0E m=audio 21850 RTP/SAVPF 18 9 8 0 3 101 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fingerprint:sha-256 09:0E:1F:D3:0D:FD:F9:54:AA:FF:AD:7F:CC:0A:85:94:D6:16:38:9F:94:43:3F:78:AC:E8:FC:8B:2B:0C:83:CA a=setup:actpass a=rtcp-mux a=rtcp:21850 IN IP4 10.0.64.100 a=ssrc:1443087231 cname:sXdALt5gCXfXVjVo a=ssrc:1443087231 msid:no6JiXqEFi18v7d8DbGarc1t7srxqe0E a0 a=ssrc:1443087231 mslabel:no6JiXqEFi18v7d8DbGarc1t7srxqe0E a=ssrc:1443087231 label:no6JiXqEFi18v7d8DbGarc1t7srxqe0Ea0 a=ice-ufrag:ksoCugWrzdPPiGkB a=ice-pwd:un25HYb9LhXxqILA0Z2l6ODw a=candidate:9999950793 1 udp 659136 10.0.64.100 21850 typ host generation 0 a=ptime:20 ------------------------------------------------------------------------ recv 888 bytes from udp/[10.0.64.63]:6060 at 20:34:55.101106: ------------------------------------------------------------------------ PRACK sip:mod_sofia@10.0.64.100:6060 SIP/2.0 Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK3008370972459 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Date: Wed, 16 Sep 2015 17:34:55 GMT Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a(a)10.0.64.63 CSeq: 102 PRACK RAck: 2100659333 101 INVITE Allow-Events: presence, kpml Max-Forwards: 70 Content-Type: application/sdp Content-Length: 376 v=0 o=CiscoSystemsCCM-SIP 357909 1 IN IP4 10.0.64.63 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 21850 RTP/AVP 9 0 8 3 18 101 a=rtcp-mux a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 617 bytes to udp/[10.0.64.63]:6060 at 20:34:55.101291: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK3008370972459 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a(a)10.0.64.63 CSeq: 102 PRACK Contact: <sip:mod_sofia@10.0.64.100:6060> User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Supported: precondition, 100rel, timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ On 24 Sep 2015, at 19:45, Brian West <brian(a)freeswitch.org> wrote: > What is the call flow and devices involved? You shouldn't be getting the > webrtc SDP if your bridge line and settings are correct, can you > elaborate? Also that second part looks like a bug those should go on JIRA. -- Serge S. Yuriev

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