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help with record_session
by John Rutherford 03 Oct '08

03 Oct '08

Hi, I'm having trouble getting record_session to work. I added record_session to my dialplan and now all calls are rejected with 488. The error logged says "sofia_glue.c :1476 sofia_glue_tech_set_codec() No audio codec available". I have tried several different file extensions including wav, gsm, au, etc., and I get this same error with all of them. Has anyone run into this before? I installed from the openSUSE 10.3 rpm. Thanks! John

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Issue (maybe a Bug?) in Receiving DTMF
by Klaus Teller 02 Oct '08

02 Oct '08

Hi, I have a voice application running in a voice browser and sending DTMF tones. On the other hand, I have a self-written Java tool running through Freeswitch to load-test the voice application. To interact, both applications essential do the following: sending DTMFs, reading DTMS, playing audios. Now, when a lot of calls are initiated (48 calls with at most 12 concurrent), it seems that Freeswitch misses some digits. I looked at the wireshark trace and it shows that the sequence that was sent by the voice application is 253988 while Freeswitch only received 23988. In each call, the application under test will typically play 10 such DTMF sequences; multiplying this by the number of calls, it means that the problem happened once in 480 times. Or in other terms since each dtmf sequence has 6 digits, this problem happened for one digit out of 2880. Yet, we can consistently reproduce this. I have the Freeswitch logs and the Wireschark dump (50Megs) if somebody would like to take a look. Is there anything i might be doing wrong? Thanks, Klaus. -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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Who has a Polycom SoundStation IP 6000
by Brian West 02 Oct '08

02 Oct '08

Just wanting to know who might have one of these wonderful phones and would be willing to test a few things with FreeSWITCH and this phone.... Or ship me some for testing... either contact me off least please. /b

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No ring-back tone.
by henkoegema 01 Oct '08

01 Oct '08

When I dial with my mobile phone to my DID line, I don't get a ring-back tone. :confused: The extension where the DID line is answered is like this: <extension name="henk"> <condition field="destination_number" expression="^henk$"> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_from_${caller_id_number}.wav"/> <action application="bridge" data="sofia/internal/2002@$${domain}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="hangup"/> </condition> </extension> I tried to add <action application="set" data="transfer_ringback=${us-ring}"/> after (and before) the 'bridge' application, but that doesn't help. Henk -- View this message in context: http://www.nabble.com/No-ring-back-tone.-tp19767182p19767182.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Record_session volume
by Cesar Cepeda 30 Sep '08

30 Sep '08

Hi, I'm using "record_session" to record a one-legged call, that is, a call which is established between a FS extension that plays audio files and a person. The call is recorded OK, but the volume of the person talking is several times lower compared to that of FS playback. Is there a way to manage the gain of each channel during a record_session? Is this something that should be happening? Cesar Cepeda.

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Update procedure
by jflowers 30 Sep '08

30 Sep '08

I have a several weeks old freeswitch-snapshot installed on FreeBSD 6.2 that I would like to update and keep current with svn but I am unsure of the correct procedure. I checked out freeswitch trunk, ran bootstrap.sh, ./configure and make but am having second thoughts about running make install over the freeswitch-snapshot install directory (/usr/local/freeswitch). Is that the best way to do it? Once I have done whatever I need to do to get to the svn version, can I just do svn update, ./configure make && make install? Can I use make speedy-current? Reason I ask is that this is an older slow machine with limited disk space. Takes a while. -- View this message in context: http://www.nabble.com/Update-procedure-tp19751007p19751007.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Which version to run in "production"
by Martin Joseph 30 Sep '08

30 Sep '08

HelloFreeSWITCHers, I am just curious as to which version of FS you all run on your in use servers? It seems like the SVN of trunk is being recommended to fix a lot of issues, but are people actually using trunk on customers servers? I am running 1.01 and haven't had issues with it, but I am curious about this. Thanks for your input, Marty PS FS is great so thanks to all for your contributions.

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk
by Jon Bruel 30 Sep '08

30 Sep '08

OK, you can access it directly now, and yes it would be fine if we can find out what's wrong once for all. Phone is: +45 45 16 1001. I'm on CET. Regarding the calling application: It is Asterisk, setting up one leg to MOH and the other to FS, so audio is certainly going through. Regarding the responsiveness of FS when the load is high: it does respond, but the is a delay of around 1 second. The load measurement is done by "top". /Jon

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cant access mailbox if userid is non numeric
by xbipin 30 Sep '08

30 Sep '08

under windows, i faced the following problem with FS. if i change the user_id to a non numeric value then the user can register to the FS but can access his mail box by calling 1005. as soon as i change the user_id to 1005 then dialing that i cana ccess my mailbox, y is this so? y cant the user have his name as user_id rather a number? <include> <user id="demo" mailbox="1005"> <params> </params> <variables> <variable name="accountcode" value="1005"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Ankur"/> <variable name="effective_caller_id_number" value="1005"/> </variables> </user> </include> -- View this message in context: http://www.nabble.com/cant-access-mailbox-if-userid-is-non-numeric-tp197242… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Default config changes
by Brian West 30 Sep '08

30 Sep '08

FreeSWITCHers, I have been working to make the default configuration multi-domain aware. So please check it out. I have made a few more changes that were recommended by the community and added some documentation into the config and some warnings that shouldn't be ignored! ;) The 9999 extensions no longer plays the bong tone. It plays a file saying "this call has been secured" and it only does so if TLS and SRTP are active. Before it would do it only on SRTP which wasn't really secure in my opinion since you just sent the crypto key in plain text only milliseconds before. Please check it out and let me know if you find any bugs or have input for additions. /b

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