December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Sounds for pending 1.0.2/Hardware
by Brian West 12 Dec '08

12 Dec '08

FreeSWITCHers, I'm looking for donations for the next batch of sound files we need to have done for the up coming 1.0.2 release. I have had others pitch in some money in the past and I thank everyone for doing so. I hope everyone can come together and help me raise about $200 to pay for this batch of prompts. I also would like to thank Bandwidth.com and Teliax for their support of the FreeSWITCH project. Are you ready for 1.0.2? Go download SVN Trunk and beat it up for us... If you wish to donate please paypal brian(a)freeswitch.org that'll help out! Happy Holidays, Brian West FreeSWITCH.org PS: If you know of any sound files we need let me know.

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segmentation fault by auto record
by Carole O. 11 Dec '08

11 Dec '08

Hello, I have intalled Freeswitch from opensuse.org as a rpm. I have opensuse 10.3. I did not make any big configuration, I have just changed a little the default dialplan and adapted some other files like conference.conf.xml. I have created a new profile in conference.conf.xml and add the command to order the automatic record of the conferences: <profile name="profilespeaker"> </profile> I have the following extension in my dialplan that uses this profile: <extension name="call_speakers"> <condition field="destination_number" expression="^0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=call_speakers" /> <action application="set" data="conference_auto_outcall_id_number=0911" /> <action application="set" data="conference_auto_outcall_timeout=60" /> <action application="set" data="conference_auto_outcall_flags=mute" /> <action application="set" data="api_hangup_hook=conference conf_speaker kick all" />  <action application="conference_set_auto_outcall" data="user/1021@$${domain}" /> <action application="conference_set_auto_outcall" data="user/1022@$${domain}" />  <action application="conference" data="conf_speaker@profilespeaker" /> </condition> </extension> 1021 and 1022 are IP loud speakers. I call them with 0911, they answer, I can talk and everything works well. However, when I hang up a segmentation fault appears and freeswitch shutdowns. In recordings I can find a file which corresponds to the recorded call but this is empty. Nothing has been recorded. If I comment the line with the recording command then it works without problem except the recording... I have joined two files: one contains the errors that appears when I run freeswitch and the other what happens if I call the extension 0911. http://www.nabble.com/file/p20935513/running_freeswitch.txt running_freeswitch.txt http://www.nabble.com/file/p20935513/call_extension.txt call_extension.txt If someone has an idea, it would be very helpful Thanks!! Carole -- View this message in context: http://www.nabble.com/segmentation-fault-by-auto-record-tp20935513p20935513… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Restricting SIP methods in Allow header
by Erick Johnson 11 Dec '08

11 Dec '08

Is it possible it configure a sip-profile so that the UA reports a restricted set of methods in the SIP allow header? For instance, I would like to remove all methods except ACK, BYE, CANCEL, NOTIFY, and PRACK Thanks Erick -- Erick Johnson

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No audio after transfer
by John Rutherford 11 Dec '08

11 Dec '08

Sorry to repost, but I haven't heard anything back on this in a little while. I checked out the trunk last week. I'm on revision 10597. Thanks, John From: John Rutherford Sent: Monday, December 08, 2008 4:36 PM To: freeswitch-users(a)lists.freeswitch.org Subject: No audio after transfer I'm trying to get an attended transfer work with freeSWITCH, but it's not quite working. I have Microsoft Speech Server on one side and Televantage on the other. MSS is originating a call, which freeSWITCH is bridging to Televantage. That calls connects just fine. Then, MSS sends a re-INVITE to Televantage to put the call on hold. This works too. Then, MSS originates another call to freeSWITCH, which is again bridged to Televantage. This works fine too. Then, MSS sends a REFER to freeSWITCH to do the transfer. The transfer should be complete, but there is no audio between the two calls-just silence. I have looked at pcaps and the freeSWITCH logs, but I'm not seeing anything obviously wrong. After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. Any help would be greatly appreciated. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Thanks! John

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Having trouble with capturing key presses from Javascript
by Alexandru Nedelcu 11 Dec '08

11 Dec '08

Hi, I've tried a simple example from here: http://wiki.freeswitch.org/wiki/Session_streamFile You can view my code here: http://paste.scsys.co.uk/paste The problem is, streamFile doesn't fire the event on keypress. Is there anything wrong with my code? Thank you, -- Alexandru Nedelcu Software Developer, Sinapticode

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Freeswitch Dialer configuration
by Alexandru Nedelcu 11 Dec '08

11 Dec '08

Hi, I'm trying to setup a simple dialer with Freeswitch. What I have right now is the following dialplan: <extension name="wavs"> <condition field="destination_number" expression="^2009$"> <action application="playback" data="/var/lib/axonite/factura.wav"/> </condition> </extension> Right now I'm testing the setup with the following command: originate sofia/external/123123123123(a)provider.com 2009 How can I configure it (or where to find examples) for the following: 1) the message should start when the phone is answered (right now it starts when the phone starts ringing I think) 2) I need keys interaction ... like when the user presses 1, the message should repeat itself, and when the user presses 2 another message should play Thank you, -- Alexandru Nedelcu Software Developer, Sinapticode

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