October 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

2 4

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

1 0

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

1 0

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.

2 4