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Update on Brian West's condition
by Ken Rice 17 Feb '12

17 Feb '12

As you already know Brian West is back in the hospital. The previous issue with the cyst has returned this time the Doctors have decided they need to go in and excise the cyst to relieve pressure on a nerve that is causing the heart problems. Here's a quick Update from Brian himself : "OK they are going in next tuesday (the 21st we can argue what NEXT tuesday vs THIS tuesday means later seems to be a HUGE debate). They may let me go home or they may not between now and then. They are NOT opening my chest they are going in thru my right side between the ribs. I'll have 5-10+ days of recovery after the surgery." See http://jira.freeswitch.org/browse/BKW-1 for more info on how you can help out and semi regular updates from Brian's hospital room Ken

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Bug?: recordFile with input callback always ends recording
by Michael Lutz 17 Feb '12

17 Feb '12

Hi All, I am using the following peace of code, It should be able to set a termination key with the cStopRecording variable, and inside the input callback it should terminate if the dtmf key entered is valid and should return "break". However as soon as it gets inside the input callback function it always terminates the recording? Based on the wiki example this should work. By debugging I can see it never gets to the"return "break"". So the evaluation is correct. Can someone please tell me what I am doing wrong or is it a bug? Regards, Mike. ---------------------------------------------------- cStopRecordingKey = "#" cFile = "blah.wav" session:setInputCallback("Record_onInput", ""); session:recordFile(cFile, 120, 10, 10); session:streamFile(cFile); function Record_onInput(s, type, obj) if cRecordingStopkey == "-" then cRecordingStopkey = "" end -- None if cRecordingStopkey == "+" then cRecordingStopkey = "0123456789*#" end -- Any if type == "dtmf" then freeswitch.consoleLog("NOTICE", "Got recording input: ".. obj['digit']); freeswitch.consoleLog("NOTICE", "Stopkey : ".. cRecordingStopkey); if (string.find(cRecordingStopkey, obj['digit'])~=nil) then -- It should only terminate when the key is in the list! return "break"; end end end ----------------------------------------------------

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mod_snmp warning messages
by Spencer Thomason 17 Feb '12

17 Feb '12

Hello, We are using mod_snmp to provide data to a graphing utility for switch stats. Recently we switched from Centos 5.6 to Debian Squeeze. I'm now seeing several WARNING messages similar to the following when mod_snmp is loaded or restarted. Everything seems to be working as it should however, the warnings were a little concerning. An example: 2012-02-06 16:14:06.822536 [WARNING] mod_snmp.c:70 Cannot adopt OID in LM-SENSORS-MIB: lmSensorsMIB ::= { lmSensors 1 } Any ideas? and is it something to worry about? Thanks, Spencer

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Ringtone on multiple destinations
by Christian Benke 17 Feb '12

17 Feb '12

Hello! I have a question about ringback-tone when calling multiple destinations. When i call a single directory-user, i get a proper ringback-tone - but when i add a second user to the bridge-command, there's no more ringback-tone. This is not about early-media - the phones only send 180 Ringing and the ringtone is provided by FS. Shouldn't FS send a ringback-tone as soon as the first phone answers with 180? Or is this behaviour just as intended? I've attached sip-traces of both situations plus the respective extension. Best regards Christian

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Internal profile not showing
by Gopalakrishnan N 17 Feb '12

17 Feb '12

Hi, I have installed Freeswitch from git in my Ubuntu 11.10. There are some default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. But when I try to register one extension to IP Phone or Softphone the account is not registered and responding as Authentication failuere. 403 Forbidden error, even though I have changed the password in 1000.xml file. Also in FS CLI I am not able to see the output for "sofia status profile internal". Please advice. Regards.

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Try to discover DND state
by Weigel, Stefan 17 Feb '12

17 Feb '12

Hi all, I'm asking myself if it's possible to get a phone state. In our case I want to know which phone (Polycom SoundPoint IP 560) is on 'do-not-disturb'. Is there a possibility to find out from FreeSWITCH site (events, notify, etc.) or should I program the soft keys on the phone. Any hint in the right direction would be really helpful. Thanks a lot! Stefan

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cdr_csv.conf.xml
by Miha Zoubek 17 Feb '12

17 Feb '12

Hi, I can see this insert in .xml file: <template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template> For outgoing calls I am getting right cdr-s. Problems the is bothering me is, how can I change caller_id_name and caller_id_number when 302 redirect happens beacuse caller_id_name and caller_id_number are wrong (inbound caller_id_name and caller_id_number , this values should be from phone, that has 302 set). Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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P-Asserted-Identity does not respect _undef_
by Steven Lam, KeenSystems B.V. 17 Feb '12

17 Feb '12

Hi list! When setting the caller id name to undefined like this: <action application="set" data="effective_caller_id_name=_undef_"/> the P-Asserted-Identity header in an outgoing invite looks like this: P-Asserted-Identity: "_undef_" <sip:123456@192.168.9.22> Other headers, for instance the From: header, do respect this. The From header in the same outgoing invite looks like this: From: <sip:123456@192.168.9.22>;tag=NcB07tpNK673p Is there something I'm missing here or is this a bug? Steven

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stop sip_info
by Piyush Sharma 17 Feb '12

17 Feb '12

I wrote a JavaScript for IVR, I used session.execute("stop_dtmf"); now dtmf is not detected. but if the phone generates the sip_info or rfc2833, it does not seem to work. Would someone tell me how can i stop them along with dtmf detection. -- Regards, Piyush Sharma, Coral Telecom Ltd.

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how to get user's friendlist on mod_skypopen
by 熊子奇 17 Feb '12

17 Feb '12

hi guys, im writing a esl python program on freeswitch skypopen, and i have subscribed MESSAGE EVENT and added filter 'proto skype', so i can send and recieve msg to/from remote skype clients. the question is : when i send bgapi "skypopen $interface #username SEARCH FRIENDS" from my program, a #username friend list can be found on freeswitch fs_cli, but no result in my program(actually, the result is: {'data': {}, 'Reply_Text': '+OK Job-UUID: 6c5905b2-590a-11e1-bac0-a5b26c9279b5', 'Job_UUID': '6c5905b2-590a-11e1-bac0-a5b26c9279b5', 'Content_Type': 'command/reply'}, that means freeswitch give a reponse to my request, but the real data i want is not returned.). any suggestions to fix this problem? are there some other EVENTs should i subscribe? thanks, regards.

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