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Maintaining call detail record
by Faisal Maqsoodi 13 Dec '08

13 Dec '08

How can i interface fs with mysql in order to maintain calls record like caller id and time n date of call etc. I ve worked on xml cdr but it contains too much info, more than i need and in a format which is not easily understandable. I also tried perl coding mentioned on the link at the bottom of the page http://wiki.freeswitch.org/wiki/Mod_cdr_csv, but so many error msgs r displayed during its execution. Is there any easy method for that. Plz help me. faisal

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Confirmation checking on VAD is needed.
by mszlazak＠aol.com 13 Dec '08

13 Dec '08

I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value="none". ???  ???  ???  ???  ???  ???  ???  ???  ???  ???  However, is there a way to confirm this in the FS console?

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missing 3 seconds of audio on bridge calls
by Angel Carpintero 13 Dec '08

13 Dec '08

Hi guys, I've a strange issue with FS , version svn -r10584 , when FS bridges a call first 3 seconds of audio are missing , looks that only happens on PSTN calls and using ringback or transfer_ringback. This only happens in calls from PSTN , not from VOIP. Some scenarios i tried to isolate this issue : - Issue PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN - Good setting bypass_media before run bridge but i need rtp in FS path PSTN --> FS ( brigde ) ringback/transfer_ringback -> PSTN - Good PSTN --> FS ( brigde ) WITHOUT ringback/transfer_ringback -> PSTN - Always good VOIP --> FS ( brigde ) -> PSTN Dialplan has nothing wrong ( i guess ): <extension name="Transfers"> <condition field="destination_number" expression="^1??XXXXXXXXXX$"> <action application="answer"/> <action application="speak" data="cepstral|Allison-8kHz|blah"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="playback_terminators=#"/> <action application="set" data="ringback=$${us-ring}"/> <action application="set" data="transfer_ringback= $${hold_music}"/> <action application="set" data="effective_caller_id_name= ${caller_id_name}"/> <action application="set" data="effective_caller_id_number= ${caller_id_number}"/> <action application="set" data="originate_timeout=30"/> <action application="set" data="call_timeout=30"/> <action application="bridge" data="sofia/default/18008226235@PSTN_GW"/> <action application="speak" data="cepstral|Allison-8kHz|Transfer finished"/> <action application="hangup"/> </condition> </extension> Any ideas ? Attached log of FS ( F8 from console ). Thanks in advance ! -- Angel Carpintero ack ( at ) telefonica ( dot ) net Key fingerprint = 3FD3 9C90 149E 7824 CECD 6BCF AC2C CA61 6EF1 B90D

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Cepstral SDK
by Pedro . 12 Dec '08

12 Dec '08

Hi, I'm trying to integrate Cepstral TTS I read in the wiki that I need Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I get the trial version of this SDK?. Thanks.

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CDR logs - adding a custom field
by Alexandru Nedelcu 12 Dec '08

12 Dec '08

In Asterisk I was able to set a custom CDR field by doing something like: Set(CDR(userfield)=${SOMETHING}) I need to set a custom field in FreeSwitch, and preferably I want to have control over its value from Javascript. Can someone tell me how? :) Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode

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Configuring FreeSwitch
by Alexandru Nedelcu 12 Dec '08

12 Dec '08

Hi, I'm a newbie trying to configure freeswitch. Our needs are simple: for starters we need to call external phones from freeswitch and play a wav file. Now, we have the voice provider's IP (no user/password authentication), and we have the codec used (called "ulaw" in Asterisk, which I think is called PCMU?). And this is basically all that we specify in Asterisk. I compiled Freeswitch with the default settings, and I was wondering if you guys can give me some hits, point me in the right direction. Thanks, -- Alexandru Nedelcu http://alexn.org

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Freeswitch logging
by Alexandru Nedelcu 12 Dec '08

12 Dec '08

Hi, I see that mod_cdr is marked as being non-functional on the wiki. I'm working on a dialer and I need a way to log information about calls. What module should I use? Thanks,

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call transfer question
by jonathan augenstine 12 Dec '08

12 Dec '08

I have a call scenario that involves transferring the call and dropping out of the SIP/RTP stream. I need to accept the SIP call, play a prompt, and retrieve a pin code. After a database lookup, I need to transfer the call to another FS server and drop out of the SIP path. I have done this with the RTP media stream previously. I am not sure what I need to do to drop out of the SIP path. Is this possible on FS? Jonathan

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conference_auto_outcall_announce
by Carole O. 12 Dec '08

12 Dec '08

Hello, I would like to call automatically a group of user and make them listening an announcement before anybody talks. I have already added an extension in the dialplan that worked. Actually, I use the extension 0911 in the default dialplan of freeswitch which I have changed a little. This works fine. The problem is to make the announcement. 1- I have created a profile for this conference and add: < param name="enter-sound" value="conference/my_file.wav"/> It did not work. I have tested this, it seems to work only if someone enters a conference which has been previously created. For instance after the 1st participant has entered the conference room. 2- Then, I have seen there is the function "conference_auto_outcall_announce" but I did not manage to make it work. I have added in my extension: <action application="set" data="conference_auto_outcall_announce=play:path_file_to_play/file.wav"/> I have got the following error: [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav] [System error : no such file or directory] I have tried to change the path but something else is wrong. 3- I have tried with the API command: conference <confname> play <file_path> It works from the console but not from the dialpan. If someone could tell me where I am wrong or has another idea it would be very helpfull. Thanks, Carole -- View this message in context: http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20955216.… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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fifo.conf.xml usage
by Jon Bruel 12 Dec '08

12 Dec '08

I'm happy to see that you can add consumers to queues using the fifo.conf.xml configuration file. I have made some tests and I hope it may lead to a more universal way of setting up queues for small organisations than the one I have described in the wiki, and which includes (too) many javascripts. I have some questions to clarify my understanding. Using the fifo.conf.xml, I find: - That the consumers continue to ring after the caller has abandoned the queue. Is there a way to avoid this? Further: - Is there a way to control the caller_id_name/number presented to the consumer? - Is there a way to control the ringing tone in the consumers like the one which can be used in the dialplan? - Can the fifo.conf.xml refer to an ODBC connection in order to get the members from a database? Finally, thanks for all the good work everybody in the FS community has put into FS, I truly believe in the possibilities of this product. Checking the hits on Google certainly indicates you moving into the right direction. /Jon

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