October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Stopping recording and playing wav files and explicit one way audio?
by Peter P GMX 09 Oct '08

09 Oct '08

I have another 3 Questions. I know I had already 2 before within the last 15 minutes, but I need to qualify whether we can build this special app with freeswitch or not. The questions are: 1.) When I do a uuid_playback I want to be able to stop this playback #1 immediately. My Idea is that I may send an empty playback file #2 next e.g. with a length of e.g. 0.1 seconds. Does file #2 stop playback of file#1? What happens, if I transfer the file via uuid_transfer to another extension while a file is playing. Is the file then still being played to the incoming leg or does it stop plaing (my whish)? 2.) I wan to manually stop recording in mod_voicemail. Via uuid_transfer I may transfer the call away from mod_voicemail to a next available agent. The question is: Is the recording still stored and available although I unbridged the recording just in the middle? I would think so as I would expect the same behaviour with a normal hangup, but can anyone confirm? 3) Is it possible to bridge a call in a way that UA1 can hear UA2 and UA2 cannot hear UA1 (one way audio)? Thank you so much for an answer. I promise I'll update the wiki with your answers. Pete

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 09 Oct '08

09 Oct '08

I have been asked by Mark Crane and others to contribute more positively by joining the community, and I would be glad to do so. As with anyone I have my own agenda being to create a virtual PBX suite - using FS or what ever I find most suitable. At this stage, I have added a load testing page to the wiki, a modest first step to my contribution. A few words about my findings regarding load tests: On the surface the FS compares with the Asterisk: +/- 20% in terms of CPU load. The test may not credit the FS for possible better resilience. Knowing Asterisk I also know its instability under load, and with the architecture of the FS, this may be resolved by design, and I'm looking forward to verifying that. So I'll - in between the other tasks - continue with load tests which attempt to simulate real life situations, with complex dial plans and with erroneous internets. Ideas are welcome! /Jon

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Error loading mod_event_multicast
by Jon Bruel 09 Oct '08

09 Oct '08

I have compiled FS with mod_event_multicast uncommented in the source/modules.conf file, but I get the following error when I start FS: [ERR] mod_event_multicast.c:220 mod_event_multicast_load() Multicast Error [CRIT] switch_loadable_module.c:767 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_event_multicast.so **Module load routine returned an error** I run Linux Fedora 9, 64 bit. Any ideas how to resolve it? /Jon

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Voicemail Event
by Nicholas Amorim 09 Oct '08

09 Oct '08

I'm building a web interface with Python/Django. Freeswitch will run on a separate server and fetches the information using xml_curl. That's working fine. What I want to do is: I want that for every voicemail received, freeswitch uploads it to another server, using some kind of event... In other words, voicemail received, FS sends it through a url or something like that. Is that possible ? Can you guys point me in any direction ? I look at the mod_voicemail documentation and see no light ;/ thanks in advance, nicholas. -- View this message in context: http://www.nabble.com/Voicemail-Event-tp19884284p19884284.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Bridge 2 calls, catchup hangup and connect to 3rd party
by Birgit Arkesteijn 09 Oct '08

09 Oct '08

Hi, I'm running FreeSWITCH 498:8901 on x86_64 GNU/Linux. I'm trying to get person 1 to speak to person 2, then after person 2 hangs up, speak to person 3, etc, etc. So: person1 -> person2 person1 -> person3 ... person1 -> personN In between connecting person1 to the next person, I need to do some admin and speak phrases to person1 and personX independently. Our software originates the calls, so I've chosen for RPC and JS. I succeed orginating a call to person1 and person2 and bridging them. However, I struggle to find a way to 'catch' the hangup and make person1 connect to person3. I tried using session.setHangupHook, but the function is never called. Any idea how I can catch the hangup of person2 so it doesn't hangup on person1, so I can connect person1 to person3? This is what I've got so far: function on_hangup2(hup_session, how) { console_log("info", "on_hangup2(): " + how + " HOOK" + " name: " + hup_session.name + " state: " + hup_session.state + " cause: " + hup_session.cause + "\n"); exit(); } function on_hangup1(hup_session, how) { console_log("info", "on_hangup1(): " + how + " HOOK" + " name: " + hup_session.name + " state: " + hup_session.state + " cause: " + hup_session.cause + "\n"); cSession.speak("cepstral", "Diane", "Wait we are trying to find the next person"); var m2Session = new Session("sofia/gateway/westhawk/0661"); cSession.setHangupHook(on_hangup2); m2Session.setHangupHook(on_hangup2); bridge(cSession, m2Session); } var cSession = new Session("{ignore_early_media=true,call_timeout=60}sofia/gateway/westhawk/0662"); while (cSession.ready()) { // The call has been answered cSession.speak("cepstral", "Diane", "Wait we are trying to find the next person"); var m1Session = new Session("sofia/gateway/westhawk/0663"); cSession.setHangupHook(on_hangup1); m1Session.setHangupHook(on_hangup1); bridge(cSession, m1Session); } (end js). I tried cSession.bridge("sofia/gateway/westhawk/0663|sofia/gateway/westhawk/0661"); but that didn't work either. Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Ringing event on outbound legs
by Peter P GMX 09 Oct '08

09 Oct '08

Hello, is there any way to determine whether the outbound leg is ringing? For example when (in SIP terms) the called UA is sending back a Ringing message via SIP, can I somehow recognize this event? Best Regards Peter

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Playing Ring tones without answering
by Peter P GMX 09 Oct '08

09 Oct '08

Hello, is it possible to play special ring tones or a wav file before answering the call? I do not really believe so, but maybe there is a chance? Best regards Peter

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Openzap error
by Luis M. Zuccolo 09 Oct '08

09 Oct '08

Hi, I've and openvox card with one fxo on port 1 connected to a line and one fxs card on port 3 connected to a phone. Up to svn 9890 its works ok but when upgrade to this version I get an DESTINATION_OUT_OF_ORDER error in calls bridged to a fxs. Thanks in advance 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:144 switch_core_standard_on_execute() OpenZAP/1:1/1 Execute bridge(openzap/3:1/1) 2008-10-08 12:11:02 [ERR] mod_openzap.c:907 channel_outgoing_channel() Invalid dial string 2008-10-08 12:11:02 [ERR] switch_ivr_originate.c:964 switch_ivr_originate() Cannot create outgoing channel of type [openzap] cause: [DESTINATION_OUT_OF_ORDER] 2008-10-08 12:11:02 [DEBUG] switch_ivr_originate.c:1454 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-10-08 12:11:02 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-10-08 12:11:02 [NOTICE] mod_dptools.c:1875 audio_bridge_function() Hangup OpenZAP/1:1/1 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-10-08 12:11:02 [DEBUG] switch_channel.c:1434 switch_channel_perform_hangup() Send signal OpenZAP/1:1/1 [KILL] 2008-10-08 12:11:02 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal OpenZAP/1:1/1 [BREAK] 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (OpenZAP/1:1/1) State EXECUTE going to sleep 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() OpenZAP/1:1/1 Running State Change CS_HANGUP 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (OpenZAP/1:1/1) State HANGUP 2008-10-08 12:11:02 [DEBUG] mod_openzap.c:428 channel_on_hangup() Changing state on 1:1 from IDLE to HANGUP 2008-10-08 12:11:02 [DEBUG] mod_openzap.c:471 channel_on_hangup() OpenZAP/1:1/1 CHANNEL HANGUP 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() OpenZAP/1:1/1 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2008-10-08 12:11:02 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (OpenZAP/1:1/1) State HANGUP going to sleep 2008-10-08 12:11:02 [DEBUG] switch_core_session.c:815 switch_core_session_thread() Session 3 (OpenZAP/1:1/1) Locked, Waiting on external entities 2008-10-08 12:11:02 [NOTICE] switch_core_session.c:833 switch_core_session_thread() Session 3 (OpenZAP/1:1/1) Ended 2008-10-08 12:11:02 [NOTICE] switch_core_session.c:835 switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP] openzap.conf [span zt] name => OpenZAP number => 1 fxo-channel => 1 [span zt] name => OpenZAP number => 2 fxo-channel => 2 [span zt] name => OpenZAP number => 3 fxs-channel => 3 <openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="9"/> </settings> <analog_spans> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-analog-option" value="3-way"/> <param name="moh" value="$${base_dir}/sounds/Dardedisco.wav"/> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </analog_spans> </configuration> added to dialplan/default.xml <extension name="incoming-fx0-channel-1"> <condition field="source" expression="mod_openzap"/> <condition field="destination_number" expression=="^(1)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="openzap/3/1"/> </condition> </extension> __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! ¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar

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New Stuff
by Anthony Minessale 09 Oct '08

09 Oct '08

For those who like to wiki, there is some new stuff. 1) Per leg call timeouts in originate strings: [leg_progress_timeout=5,leg_timeout=10]sofia/default/1000(a)some.box 2) session_heartbeats (ask in irc or see commits) 3) mod_managed (mono) 4) mod_loopback (kinda like chan_loco in asterisk) 5) bound sip profiles (share presence and mwi and users across multiple profiles) Anyone think of any more? -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Voicemail - mod_native_sound
by Gabriel Kuri 08 Oct '08

08 Oct '08

I'm trying to get voicemail to work with G.729a clients in pass-through mode, by simply using G.729a everywhere and not even dealing with transcoding, even from/to the carriers. I posted back in September asking about the sound file formats and how to playback native G.729a sounds files for the voicemail menus, to which Brian kindly responded, indicating that I need to remove the .wav extensions from sounds.xml and convert the sound files into their native format with the appropriate extension (ie vm-press.G729), and mod_native_sound would take care of the rest. That works great! For reference, here's that original thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/00654… The problem I'm running into at this point has to do with calling into voicemail and leaving a message, recording your greeting, recording your name, etc. Voicemail is attempting to encode the audio stream into what appears to be PCM format, which fails since it is unable to decode the G.729 stream. Is there any way to have mod_voicemail store the recording, greeting, etc. it in it's native format without any transcoding (ie use mod_native_sound) to avoid this issue with transcoding? Someone posted a similar question on jira, but it doesn't look as though it had been answered ... http://jira.freeswitch.org/browse/MODENDP-46 Thanks for the help, Gabriel Kuri

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