October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with transfer and Snom phones...
by Klaus Hochlehnert 13 Oct '09

13 Oct '09

Hi, I'm facing an issue and I don't know why this happens or what I can do to solve this. Here's the scenario: - FS call timeout set to 30 sec - Setting continue_on_fail=true and hangup_after_bridge=false - 1 Snom phone and an external SIP/ISDN gateway (Lancom) connected to FS established a call and talk to each other (dir. from GW to Phone) - The called phone transfers the call to another internal Snom using the transfer button on the phone - The third phone doesn't answer and after 30 sec the call gets hung up with status NORMAL_CLEARING Actually I want the call back to the initiator of the transfer if the 3rd phone doesn't answer. Can anyone tell me why this happens and how I can set it up to do what I want? Thanks, Klaus P.S.: I'm using FS trunk from last week and newest Snom Firmware

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Re corded file as voicemail.
by Nagalenoj 13 Oct '09

13 Oct '09

Is it possible to treat a recorded voice as voice mail? Assume that, I've recorded a conversation and I want this recorded file to be treated like voicemail. So, I could check it like voicemail!! -- View this message in context: http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25751158.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Question about fax tone detection
by homqua 13 Oct '09

13 Oct '09

Hi, I have implemented the solution for tone detection in wiki, and also answer the channel before detecting the tone: <condition field="destination_number" expression="^(055138419992)$"> <action application="answer"/> <action application="tone_detect" data="fax 1100 r +15000 transfer fax XML default"/> <extension name="fax"> <condition field="destination_number" expression="^fax$"> <action application="answer" /> <action application="sleep" data="1000"/> <action application="rxfax" data="/usr/local/freeswitch/storage/fax/${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.tiff"/> <action application="set" data="fax_mode=recv"/> <action application="hangup"/> </condition> </extension> But FS cannot recognize the tone, and therefore cannot move to fax extension. Below are the error in FS: 2009-10-12 10:57:16.702287 [NOTICE] switch_channel.c:602 New Channel sofia/external/anonymous(a)anonymous.invalid [c431f0a3-9231-4724-ba39-9e4ef7edfca2] 2009-10-12 10:57:16.703413 [INFO] mod_dialplan_xml.c:315 Processing Anonymous->055138419992 in context public 2009-10-12 10:57:16.719288 [NOTICE] switch_ivr.c:1349 Transfer sofia/external/anonymous(a)anonymous.invalid to XML[055138419992@default] 2009-10-12 10:57:16.719288 [INFO] mod_dialplan_xml.c:315 Processing Anonymous->055138419992 in context default 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:649 Channel [sofia/external/anonymous(a)anonymous.invalid] has been answered 2009-10-12 10:57:16.722289 [NOTICE] mod_dptools.c:1324 Enabling tone detection 'fax' '1100' 2009-10-12 10:57:16.723302 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/external/anonymous(a)anonymous.invalid [CS_EXECUTE] [NORMAL_CLEARING] 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/anonymous(a)anonymous.invalid) Ended 2009-10-12 10:57:16.740285 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/anonymous(a)anonymous.invalid [CS_DESTROY] And the trace for SIP messages: http://pastebin.com/m4e47e7d9 If anyone has any idea, tell me please. Thanks. -- View this message in context: http://n2.nabble.com/Question-about-fax-tone-detection-tp3807298p3807298.ht… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Problem with Luasql on x86_64
by DJB 13 Oct '09

13 Oct '09

Hello, I am trying to use luasql with freeswitch but having a hard time compile it with x86_64 Server running CentOS 5.3 Linux 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux [root@fsx1 luasql-2.1.1]# make export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic -I../compat/src -I/opt/local/include -I/usr/local/include -o src/sqlite3.so -shared src/luasql.o src/ls_sqlite3.o -L/opt/local/lib -lsqlite3 /usr/bin/ld: src/luasql.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC src/luasql.o: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [src/sqlite3.so] Error 1 Any suggestions. Thank you.

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torrents
by Frank Carmickle 12 Oct '09

12 Oct '09

Hello I would like to watch these files. Can someone seed please. Thank you. --FC

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Upgrading causes no answer
by Lars Zeb 12 Oct '09

12 Oct '09

http://pastebin.freeswitch.org/10612 I having been running v14996 OK for a while. I have upgraded a couple of times after, but every time, an inbound call is hung up on. The only thing that has changed is the upgrade. This morning I upgraded to v15098 and the problem persists. I believe it has to do with a lua script I use for inbound calls. Reading from the log, just after the script is launched, the following two lines appear: switch_cpp.cpp:1116 session not ready switch_cpp.cpp:925 destroy/unlink session from object Has something changed recently with lua processing? Is there something in the lua script which is causing the problem? I would appreciate any help. Thanks, Lars

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hangup_after_bridge=false not continueing after failed bridge
by Matthew Fong 12 Oct '09

12 Oct '09

I think think this might be a bug, but wanted to post here instead of Jira in-case I'm overlooking a configuration variable Dialplan <extension name="1920"> <condition field="destination_number" expression="^1920$"> <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/gateway/ debug.com/14159927717"/> <action application="transfer" data="1999"/> </condition> </extension> API Command originate sofia/internal/sip_1%192.168.1.10 1920 When the bridge to 14159927717 fails (NO_ANSWER) both calls are terminated instead of continuing on in the dial plan to exten 1999 (which in my dialplan parks the call). hangup_after_bridge however seems to work OK if someone picks up in the bridge. Is this the correct behavior? How else can I prevent the call from hanging up if a bridge fails? Thanks. I'm using 15135M --matt http://www.hellohunter.com - Predictive Dialer http://www.hellohunter.com/voice_broadcast.php - Voice Broadcasting

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SIT tones and SIP Trunk provider.
by Vinuth Madinur 12 Oct '09

12 Oct '09

Hi, Does Freeswitch detect all of these hangup cases mentioned here [ http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP Trunk provider? If not, should I put in tone_detect application in the dialplan for detecting the SITs? Won't freeswitch have to depend on the SIP status sent from SIP trunk to know the hangup status? So, I'm wondering if tone_detect will work at all? Please provide your advice. Thanks, Vinuth.

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Is assistivetech.net confusing the market place with their use of FreeSwitch product name?
by EdPimentl 12 Oct '09

12 Oct '09

http://www.assistivetech.net/search/productDisplay.php?product_id=18854 -E

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Can freeswitch forward an un-implemented SIP method?(like a SIP proxy server)
by Ryanny Lin 12 Oct '09

12 Oct '09

Hi, all I used sipp to test to forward an un-implemented SIP method, but I got 501 Not Implemented. Can freeswitch be a SIP proxy server? SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 10.10.59.161:5061 From: <sip:1000@10.10.59.69:5060> To: <sip:1001@10.10.59.69:5060>;tag=gBpay50SD2SUD Call-ID: 10-27119(a)10.10.59.161 CSeq: 3 TESTA User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 It looks like freeswitch always processes SIP messages. -- Sincerely regards, Ryanny Oct. 13

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