March 2007 - FreeSWITCH-dev - FreeSWITCH Mailing Lists

Unintended call termination
by Per Møller 11 May '07

11 May '07

I am testing FS with a range of phones. My first problem is that when I make a call from a Snom 3xx phone (www.snom.com) the moment the call is answered - the call is terminated. If I instead make the call from one of my Linksys phones, the call works fine. Below is the output from the console from the moment the call is answered until the call was terminated. I call from locally attached phone 12 to locally attached phone 11 (in this case 2 Snom 360 phones). Maybe there is some indication to the problem here: 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:4447 event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1961 audio_bridge_thread() read: sofia/standard/12(a)192.168.11.222 Bad Frame.... Bubye! Somebody have any idea how to fix this, or perhaps some hints on how to debug this further? // Per ------------------------------------------------------------ 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/11 entering state [ready] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2567 sip_i_state() Remote SDP: v=0 o=root 217248721 217248722 IN IP4 192.168.11.187 s=call c=IN IP4 192.168.11.187 t=0 0 m=audio 29442 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2201 negotiate_sdp() Codec Compare [pcma:8]/[PCMA:8] 2007-01-23 07:19:12 [INFO] mod_sofia.c:1379 tech_set_codec() Set Codec sofia/standard/11 pcma/8000 20 ms 2007-01-23 07:19:12 [INFO] mod_sofia.c:1427 activate_rtp() RTP [sofia/standard/11] 192.168.11.222:16408->192.168.11.187:29442 codec: 8 ms: 20 2007-01-23 07:19:12 [DEBUG] switch_rtp.c:500 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-01-23 07:19:12 [NOTICE] mod_sofia.c:2806 sip_i_state() Channel [sofia/standard/11] has been answered 2007-01-23 07:19:12 [INFO] mod_sofia.c:1427 activate_rtp() RTP [sofia/standard/12(a)192.168.11.222] 192.168.11.222:16410->192.168.11.144:25064 codec: 8 ms: 20 2007-01-23 07:19:12 [DEBUG] switch_rtp.c:500 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1528 sofia_answer_channel() Local SDP sofia/standard/12(a)192.168.11.222: v=0 o=FreeSWITCH 164101169533152819133 164101169533152819133 IN IP4 192.168.11.222 s=FreeSWITCH c=IN IP4 192.168.11.222 t=0 0 a=sendrecv m=audio 16410 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2007-01-23 07:19:12 [NOTICE] switch_ivr.c:2870 switch_ivr_originate() Channel [sofia/standard/12(a)192.168.11.222] has been answered 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:2894 switch_ivr_originate() Originate Resulted in Success: [sofia/standard/11] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1890 sofia_receive_message() De-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/12(a)192.168.11.222 entering state [completed] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1890 sofia_receive_message() De-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:3389 switch_ivr_multi_threaded_bridge() sofia/standard/11 State Change CS_HOLD -> CS_LOOPBACK 2007-01-23 07:19:12 [DEBUG] switch_core.c:3115 switch_core_session_run() (sofia/standard/11) State LOOPBACK 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1302 sofia_on_loopback() SOFIA LOOPBACK 2007-01-23 07:19:12 [NOTICE] mod_sofia.c:4467 event_callback() Hangup sofia/standard/12(a)192.168.11.222 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2007-01-23 07:19:12 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/standard/12(a)192.168.11.222 [1] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:2562 sip_i_state() Channel sofia/standard/12(a)192.168.11.222 entering state [ready] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:4447 event_callback() event [nua_i_error] status [500][Responding to a Non-Existing Request] session: sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1961 audio_bridge_thread() read: sofia/standard/12(a)192.168.11.222 Bad Frame.... Bubye! 2007-01-23 07:19:12 [INFO] switch_ivr.c:1968 audio_bridge_thread() Kill sofia/standard/11 [3] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1896 sofia_receive_message() Re-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1975 audio_bridge_thread() BRIDGE THREAD DONE [sofia/standard/12(a)192.168.11.222] 2007-01-23 07:19:12 [NOTICE] switch_ivr.c:2001 audio_bridge_on_loopback() Hangup sofia/standard/11 [CS_LOOPBACK] [NORMAL_CLEARING] 2007-01-23 07:19:12 [INFO] switch_channel.c:969 switch_channel_perform_hangup() Kill sofia/standard/11 [1] 2007-01-23 07:19:12 [DEBUG] switch_core.c:2970 switch_core_session_run() (sofia/standard/11) State HANGUP 2007-01-23 07:19:12 [INFO] switch_ivr.c:1968 audio_bridge_thread() Kill sofia/standard/12(a)192.168.11.222 [3] 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1896 sofia_receive_message() Re-activate timed RTP! 2007-01-23 07:19:12 [DEBUG] switch_ivr.c:1975 audio_bridge_thread() BRIDGE THREAD DONE [sofia/standard/11] 2007-01-23 07:19:12 [DEBUG] switch_core.c:2970 switch_core_session_run() (sofia/standard/12(a)192.168.11.222) State HANGUP 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1251 sofia_on_hangup() Channel sofia/standard/11 hanging up, cause: NORMAL_CLEARING 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1267 sofia_on_hangup() Sending BYE to sofia/standard/11 2007-01-23 07:19:12 [DEBUG] switch_core.c:2674 switch_core_standard_on_hangup() Standard HANGUP sofia/standard/11, cause: NORMAL_CLEARING 2007-01-23 07:19:12 [DEBUG] switch_core.c:3413 switch_core_session_thread() Session 14 (sofia/standard/11) Locked, Waiting on external entities 2007-01-23 07:19:12 [INFO] switch_core.c:3418 switch_core_session_thread() Session 14 (sofia/standard/11) Ended 2007-01-23 07:19:12 [NOTICE] switch_core.c:3294 switch_core_session_destroy() Close Channel sofia/standard/11 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1251 sofia_on_hangup() Channel sofia/standard/12(a)192.168.11.222 hanging up, cause: DESTINATION_OUT_OF_ORDER 2007-01-23 07:19:12 [DEBUG] mod_sofia.c:1267 sofia_on_hangup() Sending BYE to sofia/standard/12(a)192.168.11.222 2007-01-23 07:19:12 [DEBUG] switch_core.c:2674 switch_core_standard_on_hangup() Standard HANGUP sofia/standard/12(a)192.168.11.222,cause: DESTINATION_OUT_OF_ORDER 2007-01-23 07:19:12 [DEBUG] switch_core.c:3413 switch_core_session_thread() Session 13 (sofia/standard/12(a)192.168.11.222) Locked,Waiting on external entities 2007-01-23 07:19:12 [INFO] switch_core.c:3418 switch_core_session_thread() Session 13 (sofia/standard/12(a)192.168.11.222) Ended 2007-01-23 07:19:12 [NOTICE] switch_core.c:3294 switch_core_session_destroy() Close Channel sofia/standard/12(a)192.168.11.222

3 2

Build problem with xmlrpc_config.h
by Jason Steele 31 Mar '07

31 Mar '07

Hi, I've just downloaded (yesterday) FreeSWITCH for the first time, loaded it into VS 2005 and right-clicked Build. I got the error "PRJ0019 A tool returned an error code from "performing pre-build event."" in file xmlrpc. Followed by lots of errors indicating that xmlrpc_config.h is missing. This is my first post. I'm really hoping to make use of FreeSWITCH for an IVR telephony application we have which is written (currently) in VB.NET. In your opinion is FreeSWITCH (and mod_mono) stable enough yet to accomplish this? Many thanks, Jason Steele

2 2

FreeSWITCH Core Dump
by Peter Nixon 30 Mar '07

30 Mar '07

Hi Guys I managed to track down the core dump I was seeing. It occurs whenever I use dingaling with mod_cdr loaded. If mod_cdr is not loaded it doesn't crash. I included a backtrace below (without symbols) and will followup with one with symbols included shortly.. Cheers Peter # gdb /opt/freeswitch/bin/freeswitch core.7413 GNU gdb 6.4 Copyright 2005 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i586-suse-linux"...Using host libthread_db library "/lib/libthread_db.so.1". Core was generated by `/opt/freeswitch/bin/freeswitch'. Program terminated with signal 11, Segmentation fault. warning: Can't read pathname for load map: Input/output error. Reading symbols from /opt/freeswitch/lib/libfreeswitch.so.1...Reading symbols from /usr/lib/debug/opt/freeswitch/lib/libfreeswitch.so.1.0.0.debug...done. done. Loaded symbols for /opt/freeswitch/lib/libfreeswitch.so.1 Reading symbols from /usr/lib/libsqlite3.so.0...done. Loaded symbols for /usr/lib/libsqlite3.so.0 Reading symbols from /lib/libm.so.6...done. Loaded symbols for /lib/libm.so.6 Reading symbols from /lib/librt.so.1...done. Loaded symbols for /lib/librt.so.1 Reading symbols from /lib/libdl.so.2...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/libcrypt.so.1...done. Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libpthread.so.0...done. Loaded symbols for /lib/libpthread.so.0 Reading symbols from /lib/libc.so.6...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /opt/freeswitch/mod/mod_console.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_console.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_console.so Reading symbols from /opt/freeswitch/mod/mod_syslog.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_syslog.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_syslog.so Reading symbols from /opt/freeswitch/mod/mod_enum.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_enum.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_enum.so Reading symbols from /opt/freeswitch/mod/mod_xml_rpc.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_xml_rpc.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_xml_rpc.so Reading symbols from /opt/freeswitch/mod/mod_cdr.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_cdr.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_cdr.so Reading symbols from /usr/lib/libstdc++.so.6...done. Loaded symbols for /usr/lib/libstdc++.so.6 Reading symbols from /lib/libgcc_s.so.1...done. Loaded symbols for /lib/libgcc_s.so.1 Reading symbols from /opt/freeswitch/mod/mod_dingaling.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_dingaling.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_dingaling.so Reading symbols from /usr/lib/libgnutls.so.12...done. Loaded symbols for /usr/lib/libgnutls.so.12 Reading symbols from /lib/libz.so.1...done. Loaded symbols for /lib/libz.so.1 Reading symbols from /usr/lib/libgcrypt.so.11...done. Loaded symbols for /usr/lib/libgcrypt.so.11 Reading symbols from /lib/libnsl.so.1...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /usr/lib/libgpg-error.so.0...done. Loaded symbols for /usr/lib/libgpg-error.so.0 Reading symbols from /opt/freeswitch/mod/mod_sofia.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_sofia.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_sofia.so Reading symbols from /usr/lib/libssl.so.0.9.8...done. Loaded symbols for /usr/lib/libssl.so.0.9.8 Reading symbols from /usr/lib/libcrypto.so.0.9.8...done. Loaded symbols for /usr/lib/libcrypto.so.0.9.8 Reading symbols from /lib/libnss_files.so.2...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /lib/libnss_dns.so.2...done. Loaded symbols for /lib/libnss_dns.so.2 Reading symbols from /lib/libresolv.so.2...done. Loaded symbols for /lib/libresolv.so.2 Reading symbols from /opt/freeswitch/mod/mod_bridgecall.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_bridgecall.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_bridgecall.so Reading symbols from /opt/freeswitch/mod/mod_commands.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_commands.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_commands.so Reading symbols from /opt/freeswitch/mod/mod_conference.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_conference.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_conference.so Reading symbols from /opt/freeswitch/mod/mod_dptools.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_dptools.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_dptools.so Reading symbols from /opt/freeswitch/mod/mod_echo.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_echo.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_echo.so Reading symbols from /opt/freeswitch/mod/mod_playback.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_playback.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_playback.so Reading symbols from /opt/freeswitch/mod/mod_dialplan_xml.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_dialplan_xml.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_dialplan_xml.so Reading symbols from /opt/freeswitch/mod/mod_g711.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_g711.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_g711.so Reading symbols from /opt/freeswitch/mod/mod_gsm.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_gsm.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_gsm.so Reading symbols from /opt/freeswitch/mod/mod_ilbc.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_ilbc.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_ilbc.so Reading symbols from /opt/freeswitch/mod/mod_l16.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_l16.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_l16.so Reading symbols from /opt/freeswitch/mod/mod_speex.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_speex.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_speex.so Reading symbols from /opt/freeswitch/mod/mod_sndfile.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_sndfile.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_sndfile.so Reading symbols from /opt/freeswitch/mod/mod_native_file.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_native_file.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_native_file.so Reading symbols from /opt/freeswitch/mod/mod_softtimer.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_softtimer.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_softtimer.so Reading symbols from /opt/freeswitch/mod/mod_say_en.so...Reading symbols from /usr/lib/debug/opt/freeswitch/mod/mod_say_en.so.debug...done. done. Loaded symbols for /opt/freeswitch/mod/mod_say_en.so #0 0xb7ef5d99 in switch_channel_get_caller_extension () from /opt/freeswitch/lib/libfreeswitch.so.1 (gdb) bt full #0 0xb7ef5d99 in switch_channel_get_caller_extension () from /opt/freeswitch/lib/libfreeswitch.so.1 No symbol table info available. #1 0xb52d76a4 in CDRContainer::add_cdr (this=0x8085538, session=0x80ea910) at cdrcontainer.cpp:217 newchannel = (switch_mod_cdr_newchannel_t *) 0x80adca8 __func__ = "add_cdr" __PRETTY_FUNCTION__ = "void CDRContainer::add_cdr(switch_core_session_t*)" #2 0xb52d7505 in my_on_hangup (session=0x80ea910) at mod_cdr.cpp:119 No locals. #3 0xb7effa4e in switch_core_session_run () from /opt/freeswitch/lib/libfreeswitch.so.1 No symbol table info available. #4 0xb7f00b01 in switch_core_session_thread () from /opt/freeswitch/lib/libfreeswitch.so.1 No symbol table info available. #5 0xb7f370b8 in dummy_worker () from /opt/freeswitch/lib/libfreeswitch.so.1 No symbol table info available. #6 0xb7e0234b in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0xb7d9a65e in clone () from /lib/libc.so.6 No symbol table info available. -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc

6 11

Help with dial plan
by Daniel Corbe 28 Mar '07

28 Mar '07

http://rafb.net/p/2mEe4P69.html I was wondering if its possible to get FreeSWITCH to act in the following scenario: 1) Route a call to an IVR to play a message file... The IVR will 200 OK the call, play its prompt, then send a BYE 2) After this, the call should be tried through a list of carriers. If 1 carrier 200 OKs, stop processing. I've implemented a dial plan which I thought would behave in this way; however the second <extension> block is NEVER tried on the call. Please see the above link for details. Thanks. -Daniel

3 3

FreeSWITCH SVN Trunk compile problems on Mac OS X
by Daniel Corbe 24 Mar '07

24 Mar '07

/usr/bin/libtool: can't map file: /usr/local/src/voice/freeswitch.trunk/libs/apr/.libs ((os/kern) invalid argument)/usr/bin/libtool: can't map file: /usr/local/src/voice/freeswitch.trunk/libs/apr-util/xml/expat/lib/.libs ((os/kern) invalid argument) make[2]: *** [libfreeswitch.la] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2

2 3

Re: [Freeswitch-dev] RingBack with FreeSwitch
by Anthony Minessale 20 Mar '07

20 Mar '07

The most efficient choice would be a .wav with raw PCM or pure raw pcm captured into a file (.raw, .r8 or .r16) that matches the samplerate of your typical calls e.g. 8kz Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jurijs Cerepanovs <jury(a)ht-telecom.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, March 20, 2007 4:52:09 AM Subject: Re: [Freeswitch-dev] RingBack with FreeSwitch Jurijs Cerepanovs wrote: so ringback is OK - big thx. Which will be the best wav format for it? > Michael Jerris wrote: > Yes - this can be true ;) > Thx. > >> There is already some custom ringback tone functionality (see >> http://wiki.freeswitch.org/wiki/Special:Search?search=ringback) Does >> this meet your needs? >> >> Mike >> >> >> >>> -----Original Message----- >>> From: freeswitch-dev-bounces(a)lists.freeswitch.org >>> >>> >> [mailto:freeswitch-dev- >> >> >>> bounces(a)lists.freeswitch.org] On Behalf Of Jurijs Cerepanovs >>> Sent: Monday, March 19, 2007 6:41 PM >>> To: freeswitch-dev(a)lists.freeswitch.org >>> Subject: [Freeswitch-dev] RingBack with FreeSwitch >>> >>> Good day for all. >>> >>> Question is that - can we create module that will intercept incoming >>> call - set it to waiting state and play any file with >>> session.streamFile(...) while in background initiates new session? >>> >>> _______________________________________________ >>> Freeswitch-dev mailing list >>> Freeswitch-dev(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-dev mailing list >> Freeswitch-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> >> > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started. http://mobile.yahoo.com/mail

3 2

RingBack with FreeSwitch
by Jurijs Cerepanovs 20 Mar '07

20 Mar '07

Good day for all. Question is that - can we create module that will intercept incoming call - set it to waiting state and play any file with session.streamFile(...) while in background initiates new session?

3 4

Re: [Freeswitch-dev] XML-RPC
by Anthony Minessale 19 Mar '07

19 Mar '07

Hi, It's necessary to pay attention to the modularity of the system and overall scoping concerns when making this type of decision. The XMLRPC is a loadable module loaded by the core at runtime. "show" is an API command registered by mod_commands.so Because of this, it is not possible to extend the knowledge of XMLRPC to the 'show' command because there is no guarantee either module will be loaded. So, it's the API interface that is designed with a [string-in / string-out] prototype and the XMLRPC is only providing an interface to the API interface of freeswitch. This independance is mandatory and a fundamental of my design principle and anything else must be abandoned despite how tempting because the API interface is called from other methods such as a direct http connection, an event socket or from a scripting language such as javascript. However, It is completely possible to add support to the show command to parse a special input param and deliver that string-out as XML for instance: "show channels as xml" "show channels as csv" Each API command must provide it's own complete interpretation of the input and decision about what to output and since the entire system only has one "show" command it is not very difficult to implement this and it would be happily accepted as a patch to mod_commands.c Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jonas Gauffin <jonas.gauffin(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Tuesday, March 6, 2007 4:08:42 AM Subject: Re: [Freeswitch-dev] XML-RPC Correction: <array> <data> <value><string>0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06</string></value> <value><dateTime.iso8601>20070306T10:42:01</dateTime.iso8601></value> <value><string>PortAudio/1234</string></value> </data> </array> You can also use the <struct> tag to be able to assign names to each value. More info: http://www.xmlrpc.com/spec/ On 3/6/07, Jonas Gauffin <jonas.gauffin(a)gmail.com> wrote: > Hello > > I've played around with the XML-RPC implementation and I have some > suggestions. The output of all commands are now simple text strings. > That makes the result parsing quite hard for applications that would > like to use the interface (and it's really telnet commands wrapped > into a xmlrpc body). > > For instance, take the "show channels" result. It currently looks like > this (have removed the xml rpc body): > > > uuid,created,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,read_codec,read_rate,write_codec,write_rate > 0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06 > 10:42:01,PortAudio/1234,CS_EXECUTE,FreeSwitch,5555551212,,1234,bridge,sofia… > 37138427-cdb7-324c-b200-7123c063d473,2007-03-06 > 10:42:02,sofia/mydomain.com/1234@conference.freeswitch.org,CS_LOOPBACK,Free… > > 2 total. > > That should have been wrapped into xml instead: > > <channels count="2"> > <channel> > <uuid>0983de41-2562-fa44-8c0c-a6e5747b2c17,2007-03-06</uuid> > <created>2007-03-06 10:42:01</created> > <name>PortAudio/1234</name> > [... and all other information ...] > </channel> > </channels> > > Same goes for all other commands implemented in XML-RPC. > > Best regards, > Jonas > _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Get your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

4 7

RTP stuff
by Anthony Minessale 10 Mar '07

10 Mar '07

ok the commits are in for the rtp. When something withholds audio from the channel (ie purposely not write) CNG frames (as now advertised in sdp will flow every 5th packet interval with the proper timestamp). timestamp values increment regardless of if a channel writes data so a gap will now be properly reflected with AAAC00000CAAAA if A means audio and C means cng frames and 0 means nothing. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Trixter aka Bret McDanel <trixter(a)0xdecafbad.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, February 23, 2007 1:11:02 PM Subject: Re: [Freeswitch-dev] SNMP Discussion. On 2/23/07, janvb(a)caselaboratories.com <janvb(a)caselaboratories.com > wrote: First and doremost alarms. Any alarm raised by any module/application. And I would like a config option in FreeSWITCH where I could format how Alarms where printed out. Meaning that if I in the stack raise a LOS/LOF I want to see it in SNMP in a confugurable format. There should also be a filtering mechanisme where I in a module can raise an alarm with a alarm no etc, and in config decide what severity and processing this alarm will receive (Ignore, Log, SNMP, email etc). The issue is that many maintenance departments have big display boards where alarms with major or critical severity is shown. Some of this might already be in here? Jan Its my understanding that you would be given some data, baed on the OID you could tell what its related to. I dont understand how it would be configurable. Also, I think clarification is in order here. There are two basic methods for doing SNMP, traps where FreeSWITCH would send to some SNMP monitoring agent and gets, where a client would poll FreeSWITCH for information. It is my understanding that traps are not currently supported, although my futher understanding is that its desired they would be. To get alarms in realtime you would likely want traps to be sent, so that you can see them as they happen. Polling via a SNMP get would work but you dont always want to poll too frequently as it does cause some load however minor. As for logging, it would be on the SNMP module to decide what gets logged. There is an eventing system already within the core of FreeSWITCH that would enable a SNMP module to monitor certain things. This would be similar to how the CDR modules work, except on other events. It would be upto the other modules to push events they think are desirable, such as a red alarm on a pri line in mod_wanpipe. The SNMP module would (presumably by configuration file) see that event, decide its worthy to be sent out, and send it. It should not be configured for each module to say what to do with it, it should instead be on the event omnitor modules such as SNMP to decide if they want it. So, you could have a module that wants critical events, and would send an email, as well as an SNMP module that would fire a trap, and whatever else, and they can all see the events as they happen. This already exists, its a matter of getting the modules to react to the events that is missing. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast +44 28 9099 6461 US +1 516 687 5200 http://www.trxtel.com the VoIP provider that pays you! _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html

3 3

Re: [Freeswitch-dev] adding support for incoming streaming audio
by Anthony Minessale 10 Mar '07

10 Mar '07

The idea for the bugs is so you can stream audio to the channel while it's doing something else. As for the file format thing, I do think that can be adapted to handle streaming audio too so I am going to add a little framework code so you could express file playback in url form such as wav:///tmp/file.wav which is silly yet valid but gives way to: shoutcast://mystream.com/foo.stream Look for that in the next commit. a shoutcast type module could stand to benifit from a common streaming framework that registers as an application, a file format and a media bug. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: "mda(a)discerning.com" <mda(a)discerning.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, March 9, 2007 5:12:49 PM Subject: Re: [Freeswitch-dev] adding support for incoming streaming audio On Fri, 9 Mar 2007 11:35:15 -0800 (PST), "Anthony Minessale" <anthmct(a)yahoo.com> said: > The proper approach for this would be to use the media_bugs which are > used to record the session have have been recently enhanced to displace > the actual audio with audio you can replace. hmmm, I can see this for the case where someone might want to just sniff the frames (like recording) or if it wants to do arbitrary signal manipulation (such as noise cleanup or whatever). (perhaps a more evocative name would be better, like "filter" or "interceptor"?) but this case seems very similar in theory to what it would be to mux audio in a conference, or bridge to a hardware endpoint -- something the core should be able to do without me having to write all the by-frame signal muxing? > each streaming protocol can implement it's own mod > say "mod_shoutcast" and have an application or api command interface > hook to attach itself to the channel. but for example that isn't what mod_iax does, yet an IAX call can participate in a conference as a member, right? And so can dingaling. so why would mod_shoutcast be different? far be it to be to argue architecture with the architect :), but i'm still at a loss as to how/why this should be a "media bug" entity instead of a "file" entity or an "endpoint" entity. -mda > > media bugs is a concept where you can latch a struct and a callback onto > a channel. Your callback will be called continuously when audio is read > or > written to a channel and you can choose on setup if you want it to be > called > on read/write with muxed audio from both sides of a call or on write > giving > you a chance to replace the audio. > > http://www.freeswitch.org/docs/group__mb1.html > > If you look in switch_ivr.c in switch_ivr_record_file > http://fisheye.freeswitch.org/browse/FreeSWITCH/trunk/src/switch_ivr.c?r=44… > > You can see this is how it's possible to record from the background. > > If you look at mod_ivrtest.c (basically a scratch pad to test code) > http://fisheye.freeswitch.org/browse/FreeSWITCH/trunk/src/mod/applications/… > > the bugtest function shows how to set the channel up to run a bug > that gives you a chance to replace the frames as they are supposed to be > written to the channel so you could attach your shoutcast socket and > whenever > audio was to be written to the channel you could replace the frame with > one > from your stream so the caller would hear the stream instead of silence > or whatever the channel was really writing etc. > > > When a bug is active on a channel it forces trancoding even when it's not > necessary for a bridge so you can always get the audio as raw SLIN which > is > the only common audio format you can work with when combining media. > > > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Mark D. Anderson <mda(a)discerning.com> > To: freeswitch-dev(a)lists.freeswitch.org > Sent: Thursday, March 8, 2007 5:04:50 PM > Subject: [Freeswitch-dev] adding support for incoming streaming audio > > I'd like to have a streaming audio source (not just playing a local file) > for users while they are parked or waiting for a conference to start. > > In digging into FS it seems there are several ways this > might be approached. Of course this is based on only a superficial > understanding of the FS internals. > But I've come up with 6 (count them, 6) possibilities: > > 1) hack the "playback" application. > Right now, the lookup of driver in switch-core.c is by file suffix > (with no override in xml conf). > Currently the only real format module is mod_sndfile. > (The string constants for metadata like SWITCH_AUDIO_COL_STR_TITLE > are conveniently set to be identical to the enum used by libsndfile.) > > It could be done by hacking mod_sndfile, or instead by > making a new format driver, say "mod_url". > libsndfile does not support mp3 (and won't), and is oriented around local > files. > So a mod_url driver would be better, with support for some suffixes > like .pipe or .m3u or .mp3. It would need to do content-type sniffing > in some cases. > > One issue is that it seems that mod_playback is blocking til completion. > There is a timer in switch_ivr_play_file; it would be necessary to invert > control there to allow the processing to proceed without the file ending. > > Pros: most useful; wherever a local file is used, a remote could be used > instead. > Cons: non-trivial amount of hacking. > > 2) Use an external process to convert streaming audio into sip. > For example, using mjua or pjsua as sip clients, along with some > command-line streaming audio client such as streamripper. > > Pros: no change to FS required > Cons: lots of moving parts, and the SIP gateway utility will probably > require some hacking. > > 3) implement a "mod_shout" as an endpoint for shoutcast/icecast protocol > > Pros: fits in well with FS architecture > Cons: requires understanding FS internals concerning threading, > buffering, etc. > > 4) implement a "mod_rtsp" as an endpoint. > This would mean I'd have to be satisfied with supporting only > rtsp streaming audio, not shoutcast/icecast streaming audio. > > Pros: In theory there wouldn't be much to do, since there is already RTSP > support in the FS stack. > Cons: requires underestanding FS internals *and* RTSP streaming audio > > 5) hack mod_conference to do stream-in just like record-out, as a thread. > [I'm not sure why mod_conference does it that way, rather than at the > core > switch level (which *also* has session recording ability) -- isn't the > point > of a softswitch framework that it can allow > for arbitrary combinations of modules? Shouldn't there be a way to > supply an arbitrary channel as a member?] > > Pros: relatively easy to do by blindly copying the session record case. > Cons: specific to conferencing, and makes mod_conference even bulkier. > > 6) implement mod_park with specific support for streaming audio > [right now mod_park is just an empty stub.] > > Pros: standalone so easier to implement > Cons: limits its usefulness to just that "application" > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > > > > > ____________________________________________________________________________________ > Expecting? Get great news right away with email Auto-Check. > Try the Yahoo! Mail Beta. > http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

1 0