- FreeSWITCH-users - FreeSWITCH Mailing Lists

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.

6 10

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.

6 10

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

2 1

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

5 12

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

3 4

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

1 0

Console Dialing in Freeswitch
by Baskar 11 Dec '08

11 Dec '08

*Hi,* *It is possible to dial outbound through console dialing. Yes means me How?* *Another question whether there is any api command for console dialing.* * Thank you with regards, N.Baskar *

6 18

conference module question - prompts
by jonathan augenstine 11 Dec '08

11 Dec '08

I am trying to modify the behavior of the playing of prompts when someone enters the conference. When I enable the conf-welcome prompt and a new participant enters the conference, the prompt is played to the conference and everyone hears the welcome. Is there a any way to modify the configuration so that the welcome prompt is only heard by the participant entering the conference? Thank you. Jonathan jaugenstine(a)gmail.com

2 1

Audio routing problem between FS and Voxeo
by mszlazak＠aol.com 10 Dec '08

10 Dec '08

I’m trying to route calls from X-lite <--> FS (Nov. 6 2008 svn) <--> Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR. All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this. I’ve set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml: <extension name="Doctors Office"> <condition field="destination_number" expression="^2007$"> <action application="info"/> <action application="bridge" data="sofia/internal/sip:Doctorsoffice@127.0.0.1:5068"/> </condition> </extension> I hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.” Any ideas? Thanks. Mark.

3 8

Hardware requests.
by Brian West 10 Dec '08

10 Dec '08

Does anyone have a Polycom 320 they would like to donate to the project? I need one to trouble shoot a problem. Please contact me off list. Thanks, Brian West

1 0