September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Documentation Effort: Lost the Post
by Michael Gende 26 Sep '09

26 Sep '09

I had posted that I was interested in writing some documentation for FS or improving upon what I've found. I'd come up with a "for dummies" guide to get my crew started on FS and drew from various sources. Probably not worth much for engineers who are working on FS module code, but it might help someone wanting to get started using FS (getting off that other VoIP system). There was a recent couple of posts about this in last week or so, about some kind of effort to coordinate documentation creation and modification. I've deleted that thread, absentmindedly. Was that Brian? Michael? None of the above? Mike G.

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Transfer hangs.
by Harry Vangberg 25 Sep '09

25 Sep '09

Hello My setup is this (I've simplified everything, because a lot of my logic isn't necesarry for showcasing this): A calls in, transfer is bound as meta app, B is bridged. When the meta app is processed, the call is transfered to a new extension, which rebridges A. But! After triggering the meta app, it hangs 20 seconds, until transfering to the new extension, unless the B-leg hangs up manually. It should be noted that I've set dtmf-type=sip-info, as I would like to bypass media–if there's a better solution to get DTMF events while bypassing media, please say so, as I know the SIP INFO solution is kinda havoced. This is my dialplan: <include> <context name="public"> <extension name="ff-ivr"> <condition field="destination_number" expression="^(\d+)$"> <action application="answer" /> <action application="bind_meta_app" data="1 b a transfer::ff-transfer XML public" /> <action application="bridge" data="sofia/gateway/gw1.fonet.dk/46934488" /> </condition> </extension> <extension name="ff-transfer"> <condition field="destination_number" expression="^ff-transfer$"> <action application="bridge" data="sofia/gateway/gw1.fonet.dk/31354228" /> </condition> </extension> ... </context> </include> A full trace of a session with A calling in, B answering, B triggering meta app, waiting for transfer, and finally bridge to C is attached. This is using freeswitch-trunk@14962

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No ring tone while recording incoming call. Please help.
by Svetik VOIP 25 Sep '09

25 Sep '09

Brian, Thank yo very much for your reply. I have tried to add transfer_ringback action, but it did not solve my problem. Destination phone is ringing, but the person who is calling does not hear ringing tone in hte handset. Is there anything in the logfile that can help you to identify the problem? Closest I can see is: 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/ 4163641113(a)67.205.74.164 receive message [TRANSCODING_NECESSARY] 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ sip:main@192.168.0.121:5060 entering state [proceeding][180] 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/ sip:main@192.168.0.121:5060! 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/ 4163641113(a)67.205.74.164 entering state [terminated][487] 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/ 4163641113(a)67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] Thank you, Igor >set ringback before record_session and also set transfer_ringback >because record_session causes an pre-answer. > >/b > >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: > >> Hi, >> >> I have trouble recording incoming calls with FreeSwitch. >> >> I have followed the instruction from Misc. Dialplan Tools record >> session >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) >> It works well for outgoing calls, but I have the problem with >> incoming calls. >> >> The person who is calling does not hear ring tone, he hears just the >> silence until >> I pick up the phone. Everything else is working, we can talk, >> conversation is recorded. >> >> Here is a copy of my dialplan for incoming calls >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml >> >> <include> >> <extension name="voipms">  >> <condition field="destination_number" >> expression="XXXXXXXXXX">  >> <action application="set" data="RECORD_TITLE=Recording $ >> {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H: >> %M)}"/> >> <action application="set" data="RECORD_COPYRIGHT=(c) >> 2009"/> >> <action application="set" >> data="RECORD_SOFTWARE=FreeSwitch"/> >> <action application="set" >> data="RECORD_ARTIST=FreeSwitch"/> >> <action application="set" >> data="RECORD_COMMENT=FreeSwitch"/> >> <action application="set" data="RECORD_DATE=${strftime >> (%Y-%m-%d %H:%M)}"/> >> <action application="set" data="RECORD_STEREO=true"/> >> <action application="set" data="RECORD_ANSWER_REQ=true"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="record_session" data="$${base_dir}/ >> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ >> {destination_number}_${caller_id_number}.wav"/> >> <action application="bridge" data="user/user1@$ >> {domain_name}"/> >> </condition> >> </include> > >

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Bind to more than one ethernet interface
by Yehavi Bourvine 25 Sep '09

25 Sep '09

Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi:

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FreeSWITCH Weekly Conference Starting, Please Call In!
by Michael Collins 25 Sep '09

25 Sep '09

Come on in! sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> or via the good old PSTN at +1-213-799-1400

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mod_cdr_csv missing timestamps in A-LEG
by Tihomir Culjaga 25 Sep '09

25 Sep '09

hello, i'm on latest trunk and for some reason i cannot get timestamps dumped in my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a and b legs dumped. cdr_csv.conf.xml: <configuration name="cdr_csv.conf" description="CDR CSV Format"> <settings>   <param name="default-template" value="example"/>   <param name="rotate-on-hup" value="true"/>  <param name="legs" value="ab"/> </settings> <templates> <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${s tart_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode} ");</template> <template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answ er_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${ write_codec}"</template> <template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_ stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${wr ite_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template> <template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answ er_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${ write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template> <template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_ name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}", "${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template> </templates> </configuration> call flow is the following: CALLER => FS => CALLED FS answers the call from CALLER, plays an announcement and bridges towards CALLED. I get different behavior when the call is released by Caller and by Called. Released by Caller: the CDR is ok having all timestamps OK CDR: Outbound LEG => "016659280","016659280","0914392122","public","2009-09-24 12:02:48","2009-09-24 12:02:54","2009-09-24 12:03:01","13","7","NORMAL_CLEARING","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA" Inbound LEG => "016659280","016659280","05000403","public","2009-09-24 12:02:27","2009-09-24 12:02:41","2009-09-24 12:03:01","34","20","NORMAL_CLEARING","5d530192-a8f1-11de-962a-e328afdb9d8d","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA" Released by Called: the CDR is NOT OK as timestamps are missing NOT OK CDR: Inbound LEG => "016659280","016659280","0914392122","public","2009-09-24 12:05:20","2009-09-24 12:05:30","2009-09-24 12:05:39","19","9","NORMAL_CLEARING","c479411a-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA" Outbound LEG =>"016659280","016659280","015000403","public",*"","","",* "0","0","NORMAL_CLEARING","b82f2046-a8f1-11de-962a-e328afdb9d8d","c479411a-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA" What can be wrong?

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ASR using mod_unimrcp, vlingo
by Jim Page 25 Sep '09

25 Sep '09

Hi All Has anyone had any experience doing ASR with mod_unimrcp in javascript? In particular, how do you deal with grammars? A simple piece of demo code would be massively appreciated - the documentation on mod_unimrcp ASR javascript bindings is TBD, which I assume means 'to be documented' ... unless it means 'to be developed' ... Also is anyone aware of a vlingo integration for freeswitch? All the best Jim

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How do I get freeswitch variable which has DTMF values.
by velusamy velu 25 Sep '09

25 Sep '09

Dear All, I am doing IVR by using perl ESL libraries. I have used ESL::IVR module. I get the DTMF by using playAndGetDigits subroutine which is defined in ESL::IVR.pm. The DTMF digit stored in freeswitch "digit" variable. To get the freeswitch variable I used getVar subroutine which is defined in ESL::IVR.pm file. When I print that digits, Perl program prints empty value while playing the menu itself. If I need to get the DTMF value I need to wait the perl program. My question is why the Perl program prints the empty value before executing the getVar function? How can get the DTMF value by using getVar function without waiting in the Perl program? Please any one help me in this problem.... Thanks, Velusamy.

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A little Q931 tool
by Helmut Kuper 25 Sep '09

25 Sep '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps and puts them in a .pcap file which is directly readable and decodeable by Wireshark/tshark. The big plus here is, that you are able to get an isdn Q931 trace even of a call in the past (as long as you have FS in DEBUG loglevel). regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKu10F4tZeNddg3dwRAjV3AKCltpyAhdqB0usC4Z2AFReRNUj/5ACfTmyH y6O0PXB/IFzXlSFpPQN13JA= =OWEy -----END PGP SIGNATURE-----

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How to play more than one voice file in play_and_get_digits function
by velusamy velu 25 Sep '09

25 Sep '09

Dear All, I am in the process of doing IVR development on FreeSWITCH. I am having doubt in the play_and_get_digits application. I am using Perl language for handling IVR. How can I play more than one sound file in play_get_digits application? For an example, $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav , /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); Is the above statement is right? Can I use more than one file in it? OR Can I use the play_get_digits as following? $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/please.av /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); In the above statements how can I assure that the first application or second application is executed? When the digit is get while playing the first application the second application should not be played. How Can I do that? Is this my understanding wrong? Correct me If I am wrong? Please help me?

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