May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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xml_cdr http post differentiation between legs
by Adam Ford 03 Jun '10

03 Jun '10

Is there any way to tell whether it is the a-leg or b-leg when receiving the POST data from mod_xml_cdr? I wrote a parser in Python which parses the a_*.cdr.xml log files, and only grabs the info in the associated b-leg log files if it needs additional information(for example - in the case of a call that ends with mod_fifo). I want to adapt this to using the POST features of mod_xml_cdr so that it can be real time. Any suggestions are greatly appreciated, -Adam

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rtpmap question PCMU/PCMA
by Mark Campbell-Smith 02 Jun '10

02 Jun '10

Hi All, I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer. Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'? I'm using sipdroid on android and it rejects this with 'codec not supported' Thanks o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]

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Portaudio: Call For Input
by Mitch Capper 01 Jun '10

01 Jun '10

Recently I have started attempting to replace an existing sip client with freeswitch. In doing so I have started looking more towards portaudio. My goal is to add some robustness to portaudio to make it better suited to being used by sip clients. Some of the features I have currently worked to implement: *Event generation for call holding and resuming *Ability to switch input/output devices during calls with live audio going on (useful for say speakerphone support) *Ability to switch both the indev and outdev at once rather than separately (time savings) *Ability to keep the audio stream initialized rather than each time a call is made/ device is rang (time savings) *Only init codecs during load / config reload rather than on audio stream init (time savings) *Ability to call play (for sounds) with active calls going on Most of the improvements so far have been to make portaudio a bit faster at doing things and less restricted than previously coded. Most changes have been made through additional configuration variables so out of the box portaudio will not function any differently than currently. My request for input is on next steps and warnings. Right now portaudio keeps track of two streams, the call audio stream and the ring audio stream (and both are inited on demand only accept with my always active stream change). I was thinking of taking the always active streams to a slightly higher level: I am thinking about moving to have it keep track of an arbitrary number of streams, a linked list of streams it is keeping track of with no more than one stream per input/output pair. This would allow for initing a device prior to use to allow for near instant use of that stream. This would remove the small delay that still exists (under 1 second currently I would estimate) when say switching on the speaker phone. Or allow for very quick playing of audio on a specific device. Aside from the speed increases in stream switching this would allow for you to play audio on a device that isnt currently the primary audio device. While I do not plan to take it this far currently, it would actually allow for a much easier time of handling multiple calls on different devices at the same time. The other way to go is to look towards trying to speed up initing streams more, there are some yields in the code that look like safety things that may be possible to remove without negative affects. Also I looked at the last merge of portaudio from upstream into trunk (end of 2007) and the changes that were made back then. The good news is there were not actually a lot of base code changes and most of them have actually been merged into upstream now so updating to the latest may not be a very hard thing to do. This is in part to see if it helps with the audio quality issues that people of portaudio seem to report as it certainly is not something you want in your client. If anyone has any input into the current state of portaudio or purposed changes please let me know. In addition if anyone has input into getting better call quality out of PA that would be extremely advantageous, as other than updating to trunk I doubt my changes will result in much of a quality improvement. I am specifically also interested if anyone knows of any of the reasons some of the safeguards that are there are in place, some of which I don't see a technical reason for them to be there unless there is funnyness in libportaudio itself (which certainly could be and may be partially resolved with updating to the changes from the last 3 years). If anyone knows any stress testing etc that presented issues for PA previously that would be helpful. I am working with PA in windows, do have a linux box I will do some limited testing on, but overall if you just can test various changes let me know as that will be helpful too as I believe removing some of the safety steps may not present as an issue right away. I could also add a "faster" option to portaudio config that would result in the optimizations rather than just removing them completely. ~Mitch

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RE FER with Replaces
by blacked 31 May '10

31 May '10

hi everyone well about my last post now I'll be more specific did anyone know how can I make a transfer, a REFER with replaces to repleace FS in the loop of the call and let the two legs with out FS. hope someone can help me -- View this message in context: http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] alias folder
by Madovsky 31 May '10

31 May '10

found the problem apparently FS didn't restarted well. I removed the pid and started again and everything is fine. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, May 31, 2010 12:50 PM Subject: alias folder Hi, is it a problem to have an alias folder (ex default directory in conf/directory) ? because it seems that reloadxml doesn't work or maybe it's a permission problem... Thanks Franck

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alias folder
by Madovsky 31 May '10

31 May '10

Hi, is it a problem to have an alias folder (ex default directory in conf/directory) ? because it seems that reloadxml doesn't work or maybe it's a permission problem... Thanks Franck

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anti SPIT mechanisms
by lands＠freenet.de 31 May '10

31 May '10

Hi, today I got my first automated SPIT call over my ekiga account. So I looked for anti SPIT mechanisms in freeswitch´s wiki and found http://wiki.freeswitch.org/wiki/Mod_limit Well, it only limits spit calls. So I looked further and found that asterisk script: exten => 123456,1,Set(pin=${UNIQUEID:-2}) exten => 123456,n,Playback(your_password_is) exten => 123456,n,SayDigits(${pin}) exten => 123456,n,Authenticate(${pin}) exten => 123456,n,Goto(some_context_ring_ring,s,1) So, let the caller repeat a random number and has to prove that he/she is an human. Well, has anyone done something like this or better mechanisms within freeswitch? Chris

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FSComm linker issue
by Jingwei Yang 31 May '10

31 May '10

Hello, I encountered six linker errors when compiling the latest codes of FSComm in VS2008. I followed the steps from here: http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to have the same issue but got no answers: http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=f… . Please enlighten me how to deal with these errors. Thanks and best regards, -Jingwei 10>fshost.obj : error LNK2019: unresolved external symbol "public: __thiscall Channel::Channel(class QString)" (??0Channel@@QAE@VQString@@@Z) referenced in function "private: void __thiscall FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class QString)" (?eventChannelCreate@FSHost@@AAEXV?$QSharedPointer@Uswitch_event @@@@VQString@@@Z) 10>mainwindow.obj : error LNK2019: unresolved external symbol "public: __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)" (??0StateDebugDialog@@QAE@PAVQWidget@@@Z) referenced in function "private: void __thiscall MainWindow::debugEventsTriggered(void)" (?debugEventsTriggered@MainWindow@@AAEXXZ) 10>mainwindow.obj : error LNK2019: unresolved external symbol "public: __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@ @QAE@PAVQWidget@@@Z) referenced in function "private: void __thiscall MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered@MainWindow @@AAEXXZ) 10>prefdialog.obj : error LNK2019: unresolved external symbol "public: __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@ @QAE@PAVQWidget@@@Z) referenced in function "public: void __thiscall Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi@Ui_PrefDialog @@QAEXPAVQDialog@@@Z) 10>prefsofia.obj : error LNK2019: unresolved external symbol "public: void __thiscall CodecWidget::setCodecString(class QString)" (?setCodecString@CodecWidget@@QAEXVQString@@@Z) referenced in function "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig@PrefSofia @@QAEXXZ) 10>prefsofia.obj : error LNK2019: unresolved external symbol "public: class QString __thiscall CodecWidget::getCodecString(void)" (?getCodecString@CodecWidget@@QAE?AVQString@@XZ) referenced in function "public: void __thiscall PrefSofia::writeConfig(void)" (?writeConfig@PrefSofia@@QAEXXZ)

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