September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

select batchfile after call
by NOx-WHV 08 Sep '09

08 Sep '09

Hi, does anybody have a tip how to start a batchfile after hanging up. After ext. 1000 calls 1001 and hang up, i need a request to call: /../../FS/batchfile 1000 if 1001 calls 1000 i need: /../../FS/batchfile 1001 and so on... Thanks for help -- View this message in context: http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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mod_managed ILoadNotificationPlugin
by Josh Rivers 08 Sep '09

08 Sep '09

What is the purpose if the ILoadNotificationPlugin? I thought it could be used to start off background code, but code run from that point seems to be terminated when the method returns. Does it only exist to check dependencies?

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ESL C questions
by Remko Kloosterman 08 Sep '09

08 Sep '09

Hi there, I wonder, is ESL documentation available for C or does someone have something in draft? I'm trying to write an outbound socket application for some generic IVR features. I didn't find exactly that on the wiki except TODO J. The perl/ruby/javascript pages help a bit and the libs/esl source code provides examples that seem useful for trial and error, but I'd rather understand a bit more first. Right now I have a socket server that forks a process, answers a call, generates beep and plays voice. How can I retrieve digits? Place an outbound call and bridge both legs or retrieve a cause if the call failed? Send/receive SIP INFO? Disconnect the call with some cause code? And all that (and some more) in C. Any help or pointers is appreciated. Thanks, Remko

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Passing custom Dial Plans in perl
by Ahmed Munir 08 Sep '09

08 Sep '09

Hi, I've set some variables in dialplan XML and I want to call these variables in perl, i.e. dialplan.xml <action application="set" data="DIALEXCUTED=NO"/> Also after executing in perl the value of this variable pass to dialplan.xml Kindly tell me how to do this. -- Regards, Ahmed Munir

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freeswitch - q: originate calls from database
by tom 08 Sep '09

08 Sep '09

hi, is this scenario doable? let the system call people , they talk do an ivr, and can dependend on their selection end up in a real call-queue. 1) how would i tell FS to call xyz-people from the databse? thx

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No dial-string available error
by Juan Backson 07 Sep '09

07 Sep '09

Hi, I am getting no dial-string available error when using xml_odbc module to bridge a call. How can I resolve this problem? EXECUTE sofia/internal/180001(a)192.168.1.130 bridge(user/180001) 2009-09-05 16:31:29.853456 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key_value]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key_name]=[name] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [tag_name]=[domain] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [section]=[directory] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Name]=[REQUEST_PARAMS] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Core-UUID]=[f649cdfd-3715-4d18-94d9-417aa7e26873] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-Hostname]=[server] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-IPv4]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-IPv6]=[::1] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-Local]=[2009-09-05 16:31:29] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-GMT]=[Sat, 05 Sep 2009 08:31:29 GMT] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-Timestamp]=[1252139489853456] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-File]=[mod_dptools.c] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-Function]=[user_outgoing_channel] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-Line-Number]=[2365] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [as_channel]=[true] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key]=[id] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [user]=[180001] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [domain]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:319 DEBUG GOING TO RENDER TEMPLATE [default] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:214 DEBUG Performing Query: SELECT enabled, sip_password, account_id FROM agent WHERE sip_user = 180001 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:214 DEBUG Performing Query: SELECT sip_password FROM agent WHERE sip_user = 180001 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:417 Debug dump of generated XML: <document type="freeswitch/xml"> <section name="directory"> <domain name="192.168.1.134"> <user id="180001"> <params> <param name="password" value="111111"></param> </params> <variables> <variable name="account_id" value="18"></variable> </variables> </user> </domain> </section> </document> 2009-09-05 16:31:29.856442 [CONSOLE] mod_xml_odbc.c:457 Generated XML is in [/tmp/16d2f01a-4332-43ff-9535-5615a192b40e.tmp.xml] 2009-09-05 16:31:29.856442 [ERR] mod_dptools.c:2430 No dial-string available, please check your user directory. 2009-09-05 16:31:29.856442 [ERR] switch_ivr_originate.c:1527 Cannot create outgoing channel of type [user] cause: [MANDATORY_IE_MISSING]

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Re: [Freeswitch-users] Checking Busy Status
by Ahmed Munir 07 Sep '09

07 Sep '09

Hi, Thanks for quick reply. I want to know how can I apply USER_BUSY in perl? Like for hangup I'm calling it from function Freeswitch::CoreSession i.e. $session->hangup(); Do I have to call it as listed below; $session->USER_BUSY(); or there other way around in perl? Kindly do let me know. > ---------- Forwarded message ---------- > From: Mathieu Rene <mrene_lists(a)avgs.ca> > To: freeswitch-users(a)lists.freeswitch.org > Date: Sun, 6 Sep 2009 22:01:58 -0700 > Subject: Re: [Freeswitch-users] Checking Busy Status > The hangup cause will be USER_BUSY. > > You can hop on #freeswitch if you need some explanations. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene(a)avgs.ca > > > > > On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: > > Hi, >> >> How can I check the busy status in FS? I've searched all the wiki pages >> i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but >> couldn't find checking busy status. >> >> I've written a perl script but couldn't complete it because theres no any >> function or class regrding busy status. Kindly let me know how can I check >> the busy status in mod_perl and also in dialplan tools as well. >> >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> _______________________________________________ >> 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 >> > > > > > _______________________________________________ > 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 > > -- Regards, Ahmed Munir

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Set disable-transcoding in dialplan
by rod 07 Sep '09

07 Sep '09

Hi all, I'm wondering if I can do something like this: - in my internal profile, I have this because of some PEER using G729: - <param name="disable-transcoding" value="true"/> But for a specific PEER, I'd like to activate transcoding: - for this PEER, only G711 is used - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND So in my dialplan, I tried before bridging: - <action application="set" data="disable-transcoding=false"/> - <action application="start_dtmf_generate" data="true"/> But I still see RFC2833 events between my FS and PEER and the DTMF are not working. So 2 questions: - does application "start_dtmf_generate" requires transcoding - if yes, can I set the variable disable-transcoding in my dialplan regards, rod

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Checking Busy Status
by Ahmed Munir 07 Sep '09

07 Sep '09

Hi, How can I check the busy status in FS? I've searched all the wiki pages i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but couldn't find checking busy status. I've written a perl script but couldn't complete it because theres no any function or class regrding busy status. Kindly let me know how can I check the busy status in mod_perl and also in dialplan tools as well. -- Regards, Ahmed Munir

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Call Transfer Question
by DJB 07 Sep '09

07 Sep '09

I am not sure if my question did not get post correctly earlier. I wonder whether anyone can give me any recommendations. Here is the call flow: I call from the PSTN (A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re-invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. Thank you.

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