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listening on multiple events in mod_managed
by babak yakhchali 10 May '10

10 May '10

Hi How is it possible to listen for maore than one type of events but not all of them something like EventConsumer con = new EventConsumer("CUSTOM CHANNEL_DESTROY CHANNEL_HANGUP", null); thanks

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signal_bond and uuid
by Madovsky 10 May '10

10 May '10

if for example I d like to send dtmf to the legB in api mod_commands should I use signal_bond as uuid of the channel ? Thanks F

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bridge_hangup_cause not for internal sip calls?
by Helmut Kuper 10 May '10

10 May '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, in my dialplan for calls from internal phones I transfer each call after hangup to a dialplan which handles the hangup causes (e.g. for announcements). Unfortunately there are at least two different channel variables used to signal the hangup cause back to FS depending on what type of target/interface was dialed. When I dial an busy internal sip phone, the hangup cause can be found in "originate_disposition" When I dial an busy external freetdm target, the hangup cause can be found in "bridge_hangup_cause" and "originate_disposition" is set to "SUCCESS". Is the a consolidated single channel variable for all types of interfaces? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFL59Mx4tZeNddg3dwRAkwDAJ0VjInzSHT9lpc0rhSVLXTBhsPLpQCgi1VL EzizysQI373J/c8GCHAetzg= =ImNz -----END PGP SIGNATURE-----

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Sorting through a list of conditions
by Ken Fulmer 10 May '10

10 May '10

We'd like to match on either Caller ID or ANI for outbound faxes and lock them to PCMU. Here's what we've been using: <extension name="FAX-LCL-7D"> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> <action application="export" data="nolocal:absolute_codec_string=PCMU"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$0(a)X.X.X.X"/> <action application="set" data="instant_ringback=true"/> <action application="set" data="ringback=${us-ring}"/> </condition> </extension> If we use a list and break=on-false, searching stops after the first non-match. If we use break=on-true, it stops after the first match, and we get no digits to send upstream. If we use break=never, it applies the digit match to all numbers, regardless of Caller ID. I realize this is all normal behavior. Is there a way to sort through a list of Caller ID's, match one, and then match the outbound digits? We're trying to avoid having to create a dial-plan entry for every Caller ID. Here's what it *might* look like with multiple internal fax lines: <extension name="FAX-LCL-7D"> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="caller_id_number" expression="^XXXXXXXXXX$"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> <action application="export" data="nolocal:absolute_codec_string=PCMU"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$0(a)X.X.X.X"/> <action application="set" data="instant_ringback=true"/> <action application="set" data="ringback=${us-ring}"/> </condition> </extension> Thanks, Ken Fulmer

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call recording with Snom and SIP-INFO not working
by Helmut Kuper 10 May '10

10 May '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I found that call recording using Snoms which sending a SIP-INFO with "Record: on" leads to a disconnect. My sip-Profile has arecord-template defined: <param name="record-template" value="recordings/${caller_id_number}.${destination_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> AFAIK this is the default one. My dialplan hast the following bind-meta-app: <action application="export" data="RECORD_STEREO=true"/> <action application="bind_meta_app" data="9 ab s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> $base_dir is "/opt/app/voip/ippbx" "recordings"-directory exists. When I press the record button on my Snom, it sends a SIP-INFO containing a "Record: on" entry. FS detects this and disconnects the call. FS's logfile shows this: [...] EXECUTE sofia/internal/2850(a)85.16.246.6 bind_meta_app(9 ab s record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav) 2010-05-10 13:24:37.257965 [INFO] switch_ivr_async.c:2332 Bound A-Leg: 9 record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav 2010-05-10 13:24:37.257965 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 9 record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav [...] Sending SIP-INFO from Snom now 2010-05-10 13:24:40.157036 [ERR] mod_sndfile.c:195 Error Opening File [/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/recordings/2850.4916.2010-05-10-13-24-40.wav] [System error : Datei oder Verzeichnis nicht gefunden.] 2010-05-10 13:24:40.157036 [ERR] switch_ivr_async.c:997 Error opening /opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/recordings/2850.4916.2010-05-10-13-24-40.wav 2010-05-10 13:24:40.157036 [NOTICE] switch_ivr_async.c:998 Hangup sofia/internal/2850(a)85.16.246.6 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] [...] So it's quite clear what happens here: Somehow FS creates a wrong file path ("/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/" instead of "/opt/app/voip/ippbx/recordings/"] When I do the same by using "*9" on my Snom, everything works fine. I'm using "FreeSWITCH Version 1.0.head (17097:17188M)" Any ideas what I'm doing wrong? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFL5/ES4tZeNddg3dwRAgYGAJ9NWJP08vUfNpM3FtmVxmvSGyRVjACfQcIX zxCC4CGXo7j1lsHaUOLX3NM= =3D1h -----END PGP SIGNATURE-----

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Encoding issue with mod_xml_cdr (apparently)
by Isaac Jurado 10 May '10

10 May '10

Hi, We are setting up a web service for mod_xml_cdr but we have problems with non-ascii characters encoded in UTF-8. The POST request (url-encoded) stops its data when it finds the first non-ascii, like the following: -----8<----- <?xml version="1.0"?> <cdr> <channel_data> <state>CS_REPORTING</state> <direction>inbound</direction> <state_number>11</state_number> <flags>0=1;18=1;36=1;38=1;51=1</flags> <caps>1=1;2=1;3=1</caps> </channel_data> <variables> <uuid>faa6c7e9-9b4a-4ac0-8ebb-971b2e2f8eaa</uuid> <sip_local_network_addr>192.168.1.99</sip_local_network_addr> <sip_network_ip>192.168.1.99</sip_network_ip> <sip_network_port>1024</sip_network_port> <sip_received_ip>192.168.1.99</sip_received_ip> <sip_received_port>1024</sip_received_port> <sip_via_protocol>udp</sip_via_protocol> <sip_authorized>true</sip_authorized> <sip_number_alias>33</sip_number_alias> <sip_auth_username>33</sip_auth_username> <sip_auth_realm>econcept.es</sip_auth_realm> <number_alias>33</number_alias> <user_name>33</user_name> <domain_name>econcept.es</domain_name> <accountcode>33</accountcode> <user_context>econcept.es</user_context> <caller_id_name>Peque%% ----->8----- This is what the web service finds under the 'cdr' POST key. The last two percent characters seem to precede the spanish 'ñ'. The caller_id_name is supplied at register time by issuing an UTF-8 encoded XML to feed mod_xml_curl. I've tried disabling the url-encoding like so (xml_cdr.conf.xml): <param name="encode" value="false"/> But then, the web service does not appear to be receiving anything. Any ideas? Cheers. -- Isaac Jurado Internet Busines Solutions eConcept http://www.econcept.es

4 3

group_confirm_file
by guru singh 10 May '10

10 May '10

Hi, I am using group_confirm_file to play a message to the called party... I am not being able to sync it properly, i want it to start playing as soon as the phone is answred... right now the called party hears it from a random point depending upon how soon the call was answered i guess. How do I manage this? here is my config file http://pastebin.freeswitch.org/12943 thanks gs

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Re: [Freeswitch-users] mod_commands
by Madovsky 10 May '10

10 May '10

found : expand uuid_send_dtmf ${uuid_getvar(4345345 signal_bond)} 2@400 no pain anymore :) From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, May 09, 2010 10:53 PM Subject: Re: [Freeswitch-users] mod_commands yes ----- Original Message ----- From: Brian West To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, May 09, 2010 9:05 PM Subject: Re: [Freeswitch-users] mod_commands Are you doing this at the cli? /b On May 9, 2010, at 8:01 PM, Madovsky wrote: Hi, I d like to do this : uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1@100 but don't find the correct syntax... Thanks for your help F _______ ---------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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mod_commands
by Madovsky 10 May '10

10 May '10

Hi, I d like to do this : uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1@100 but don't find the correct syntax... Thanks for your help F

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Anyone in UK want a free UTStarcom F1000G wifi phone?
by Bruce Hopkins 09 May '10

09 May '10

It's as new, in its original packaging - but I'm not going to need it and it's too obscure to sell through ebay. So if you want it, it's yours. I'm moving house and need to cut down on the amount of unused stuff here - so I'll even pay the UK postage to avoid throwing it away. Any taker? Seriously ! Best regards Bruce

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