April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

play_and_detect_speech errors
by Anita Hall 25 Apr '12

25 Apr '12

Hi I was using play_and_detect_speech with Nuance Speech Server 5 and it generally works great! Much thanks to Chris Rienzo for this cool App :) There are 2 cases where I need help to understand what is happening. 1. In general my SIP connection between mod_unimrcp and Nuance works fine with all the calls answered. But in 3 repeated cases, the SIP connection was not established as a result of which the call got hungup. Is there any graceful way in play_and_detect_speech (like some channel variable set) which can be used to find if the SIP call has been answered by Nuance or not. Like in this case, I got 2012-04-20 13:20:47.955516 [DEBUG] apt_consumer_task.c:97 () Wait for Messages [MRCP Client] 7430 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582 () Receive SIP Event [nua_i_state] Status 0 INVITE sent 7431 2012-04-20 13:20:47.955516 [NOTICE] mrcp_sofiasip_client_agent.c:522 () SIP Call State 0x7f4f04188270 [calling] 7432 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582 () Receive SIP Event [nua_r_invite] Status 503 Service Unavailable 7433 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582 () Receive SIP Event [nua_i_state] Status 503 Service Unavailable 7434 2012-04-20 13:20:47.955516 [NOTICE] mrcp_sofiasip_client_agent.c:522 () SIP Call State 0x7f4f04188270 [terminated] 7435 2012-04-20 13:20:47.955516 [DEBUG] apt_task.c:278 () Process Message [MRCP Client] [1;4] 7436 2012-04-20 13:20:47.955516 [INFO] mrcp_client_session.c:535 (ASR-0) Raise App Response 0x7f4f04188270 <new> [2] FAILURE [2] 7437 2012-04-20 13:20:47.955516 [ERR] mod_unimrcp.c:1860 (ASR-0) RECOGNIZER channel error! 7438 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1866 Terminating MRCP session 7439 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1541 (ASR-0) CLOSED ==> ERROR 2. After performing a recognition with a low confidence score, the call got disconnected but FreeSWITCH could not detect the call disconnection and repeatedly gave the following debug message in the log till it flooded hundreds of log files. switch_core_session.c:2102 FreeTDM/1:2/4948608 Channel is hungup and application play_and_detect_speech does not have the zombie_exec flag. During this "show calls" on fs_cli will crash fs_cli I had to shutdown freeswitch. regards, Anita

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One way Audio
by Andrew Paul 25 Apr '12

25 Apr '12

Hai, I have a freeswitch setup in that one asterisk pbx registered. For this setup am facing problem with lower version of yealink phone and grandstreams . Whenever 200 ok came in freeswitch changing IP phone ip to asterisk PBX ip.It is showing messages in FS cli switch_rtp.c "*Auto changing IP PHONE IP to ASTERISK PBX IP *" . So no audio is passing to PBX from from IP phone. Still IP phone able to get the incoming calls. Both PBX and IP are in same network. Hopping the earliest reply . Thanks And Regards Andrew Paul

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Re: [Freeswitch-users] park bridged call legs
by Peter Olsson 25 Apr '12

25 Apr '12

Try this, I'm not sure if it works though.. :) uuid_transfer <uuid> -both park inline /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Lappe, Adam Skickat: den 25 april 2012 15:03 Till: 'FreeSWITCH Users Help' Ämne: Re: [Freeswitch-users] park bridged call legs I guess I use the wrong syntax. uuid_transfer <uuid> -both park isn't working. Any advice? Thanks, Adam ________________________________ Von: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces@lists.freeswitch.org]> Im Auftrag von Peter Olsson Gesendet: Mittwoch, 25. April 2012 11:43 An: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Betreff: Re: [Freeswitch-users] park bridged call legs uuid_transfer should work for this. It can handle the flag "-both", which can transfer both legs somewhere. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces@lists.freeswitch.org]> För Lappe, Adam Skickat: den 25 april 2012 11:22 Till: freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: [Freeswitch-users] park bridged call legs Hi all, is there a way to park two call legs after I bridged them? When I use: uuid_bridge <uuid_1> <uuid_2> and then uuid_park <uuid_1> Freeswitch kills <uuid_2> with NORMAL CLEARING. I tried to use park_after_bridge but this is not useful imo since it parks every normal call when any1 hangs up. Any ideas? Thanks, Adam !DSPAM:4f97f4cc32761721410779!

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Reminder: FreeSWITCH Conference Call Today
by Michael Collins 25 Apr '12

25 Apr '12

Just reminding everyone of our call today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25 We will be discussing the execution of dialplan apps and APIs using the "execute_on" and "api_on" channel variables. We'll also have a lively discussion about why shouldn't do billing stuff inline and why people do it anyway. :) See you at 1PM Eastern, 10AM Pacific -Michael

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Re: [Freeswitch-users] park bridged call legs
by Peter Olsson 25 Apr '12

25 Apr '12

uuid_transfer should work for this. It can handle the flag "-both", which can transfer both legs somewhere. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Lappe, Adam Skickat: den 25 april 2012 11:22 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] park bridged call legs Hi all, is there a way to park two call legs after I bridged them? When I use: uuid_bridge <uuid_1> <uuid_2> and then uuid_park <uuid_1> Freeswitch kills <uuid_2> with NORMAL CLEARING. I tried to use park_after_bridge but this is not useful imo since it parks every normal call when any1 hangs up. Any ideas? Thanks, Adam !DSPAM:4f97c0f632765009093151!

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autoanswer by re-INVITE
by Евгений 25 Apr '12

25 Apr '12

Hi! I can send a new call with autoanswer by <action application="export" data="sip_auto_answer=true"/> It works well. But I need some different feature I want to "re-INVITE" existing call with autoanswer option, how can I do this? Or may be somebody knows, how can I make Cisco SPA508G to answer the call with any command from FreeSWITCH. It will be very usefull with headset to answer calls by pressing some buttons with keybord or mouse on PC. From Russia with Love :) EvgineyС уважением Евгений.

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park bridged call legs
by Lappe, Adam 25 Apr '12

25 Apr '12

Hi all, is there a way to park two call legs after I bridged them? When I use: uuid_bridge <uuid_1> <uuid_2> and then uuid_park <uuid_1> Freeswitch kills <uuid_2> with NORMAL CLEARING. I tried to use park_after_bridge but this is not useful imo since it parks every normal call when any1 hangs up. Any ideas? Thanks, Adam

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HTTPS and mod_http_cache
by RaviRaj Mulasa 25 Apr '12

25 Apr '12

Hi I am trying to prefetch a file to play using mod_http_cache. The URL @ which the file is located uses HTTPS instead of HTTP HTTPS freeswitch@myplayground> http_prefetch https://myfiles/file_1.wav USAGE: <url> HTTP freeswitch@myplayground> http_prefetch http://myfiles/file_1.wav +OK 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:396 Locked cache 2012-04-24 08:25:48.108625 [INFO] mod_http_cache.c:476 Cache MISS: size = 0 (0 MB), hit ratio = 0/1 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:541 Adding http://myfiles/file_1.wav(/tmp/http_cache/c4/9828e2-8e21-11e1-9e33-017c98d3… to cache index 0 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:407 Unlocked cache 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:728 opening /tmp/http_cache/ Please let me know how can I HTTP_PREFETCH HTTP_GET using a URL that uses HTTPS. Thanks RaviRaj.

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Bypass Media not work
by Cucu Florian 24 Apr '12

24 Apr '12

Hello, I tried to test the behavior of "bypass media" in FS with the following scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001) On each Xlite I set one single codec (that is currently unsupported by FS) as it said in wiki page: *"This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSWITCH "* As the FS tutorial said *"Before executing the bridge action you must set the "bypass_media" flag to true. "* so I put the following line in my dial plan. <extension name="extension-intercom"> <condition field="destination_number" expression="1001"> * <action application="set" data="bypass_media=true"/>* <action application="bridge" data="user/1001"/> </condition> </extension> <extension name="extension-intercom"> <condition field="destination_number" expression="1000"> *<action application="set" data="bypass_media=true"/>* <action application="bridge" data="user/1000"/> </condition> </extension> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was ok. But when I put only codecs that FS do not support(ex. iLBC,BroadVoice32) the call failed. What am I missing? I made this scenario because I need to route via FS some calls with G729r8 codec and I need no codec negotiation. Ty

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How to enable Hardware DTMF detection (Dahdi/Digium card)
by Haythem Horbit 24 Apr '12

24 Apr '12

Hi, We experience some issues to capture SIA ContactID messages from an alarm system, with Freeswitch, the problem is that some DTMFs are missed randomly, so that we can't have a correct message we use a Digium card 'AEX2400' with DAHDI drivers, the alarm system is connected to an FXO port. We see in the freeswitch log that software DTMF detection is enabled, may be the issue comes from this fact... So our question is: how can we enable hardware DTMF detection in DADHI configuration with freeswitch, and how to disable software DTMF detection in freeswitch -- *Haythem HORBIT Responsable développement NGI Maghreb* 53, rue 8601 - 2035 Charguia I Tunis - TUNISIE Tél. : +216 71 808 565 Fax : +216 71 808 395 Gsm : +216 28 801 737 www.ngi-maghreb.com

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