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mod_cepstral for Windows build -> Change language
by Christian Knoblauch 30 Oct '10

30 Oct '10

Hi, Is there a method to "dynamically" change the language for cepstral TTS at runtime ? Thanks & Rewgards Christian

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Problems making a receiveing calls with mod_jingling
by Rob Hutton 30 Oct '10

30 Oct '10

I am trying to get mod_dingaling working, and I am sure I am doing something stupid, but I can't figure out what it is... Relevant parts of call log: Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Regex (PASS) [dingaling_1_pattern_1] destination_number(94045550100) =~ /^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action set(prepend=1) Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_1/1001(a)192.168.1.200 Action bridge(dingaling/GoogleTalk/+${prepend}4045550100(a)voice.google.com) 2010-10-13 12:15:41.427591 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/1001(a)192.168.1.200) State Change CS_ROUTING -> CS_EXECUTE ... EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-spymap/1001/87e31e76-56ff-47f2-b210-a316c3191a44) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-last_dial/1001/94045550100) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 hash(insert/192.168.1.200-last_dial/global/87e31e76-56ff-47f2-b210-a316c3191a44) EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(RFC2822_DATE=Wed, 13 Oct 2010 12:15:41 -0400) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [RFC2822_DATE]=[Wed, 13 Oct 2010 12:15:41 -0400] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(prepend=1) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [prepend]=[1] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 set(hangup_after_bridge=true) 2010-10-13 12:15:41.431854 [DEBUG] mod_dptools.c:1024 sofia/sipinterface_1/1001(a)192.168.1.200 SET [hangup_after_bridge]=[true] EXECUTE sofia/sipinterface_1/1001(a)192.168.1.200 bridge(dingaling/GoogleTalk/+14045550100(a)voice.google.com) 2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:1727 Unknown Profile! 2010-10-13 12:15:41.432855 [DEBUG] mod_dingaling.c:704 Terminate called from line 1728 state=CS_NEW 2010-10-13 12:15:41.432855 [NOTICE] mod_dingaling.c:715 Close Channel N/A [CS_NEW] 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:430 () Running State Change CS_DESTROY 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY 2010-10-13 12:15:41.432855 [DEBUG] switch_core_state_machine.c:440 (N/A) State DESTROY going to sleep 2010-10-13 12:15:41.432855 [ERR] switch_ivr_originate.c:2605 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [DEBUG] switch_ivr_originate.c:3413 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [INFO] mod_dptools.c:2575 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2445 (sofia/sipinterface_1/1001(a)192.168.1.200) Callstate Change RINGING -> HANGUP 2010-10-13 12:15:41.432855 [NOTICE] mod_dptools.c:2638 Hangup sofia/sipinterface_1/1001(a)192.168.1.200 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2010-10-13 12:15:41.432855 [DEBUG] switch_channel.c:2461 Send signal sofia/sipinterface_1/1001(a)192.168.1.200 [KILL] I think the interesting message is "Unknown Profile!" Here is the dialplan that is being matched: <extension name="dingaling_1_pattern_1" continue="true"> <condition field="destination_number" expression="^91{0,1}([2-9][0-8][0-9][2-9][0-9]{6})$" bluebox="pattern_1"> <action application="set" bluebox="prepend" data="prepend=1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="dingaling/GoogleTalk/+${prepend}$1(a)voice.google.com"/> </condition> </extension> Here is the profile: <profile name="dingaling_1"> <param name="name" value="GoogleTalk"/> <param name="login" value="justlikeef(a)gmail.com/gtalk"/> <param name="password" value="!Cl0r0x1.."/> <param name="server" value="talk.google.com"/> <param name="dialplan" value="XML"/> <param name="message" value=""/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="auto-login" value="true"/> <param name="auto-reply" value=""/> <param name="sasl" value="plain"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> <param name="context" value="context_1"/> <param name="exten" value="1001"/> </profile> Complete config and log in pastebin... -- Thanks, Rob

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Problem with T.38 fax receive using ESL
by peely 30 Oct '10

30 Oct '10

Hi, I'm having problems receiving faxes through the Event Socket API using T.38 and SpanDSP. If I use the example dialplan entry: <extension name="NTS TESTfax"> <condition field="destination_number" expression="^\+(testfax)"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="/tmp/faxes/FAX-${uuid}.tif"/> <action application="hangup"/> </condition> </extension> Then everything is OK and a fax is received nicely whereas if I use an ESL command on an answered call like: SendMsg 5716720c-e291-11df-939c-75e5ad3b2de7 call-command: execute execute-app-name: rxfax execute-app-arg: {fax_enable_t38_request=true,fax_enable_t38=true,fax_header=testheader,fax_ident=testident}/tmp/faxes/myfax.tif Then there's lots of T.38 mucking about but the fax terminates with a communications error. I've tried sleeping for 2 seconds, broadcasting silence_stream://2000 etc but to no avail. Has anyone successfully got T.38 fax receive working through the ESL, and if so, what chain of commands do you execute? I'm using a GIT trunk circa Oct 15th BTW. Many thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-T-38-fax-receive… Sent from the freeswitch-users mailing list archive at Nabble.com.

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wiki question - want to add Mac osX info
by Mario G 30 Oct '10

30 Oct '10

I am about to start work my wiki contribution which specifically target my installation on Mac osX and other related information to Mac osX. I saw there is no category for "installation". So then does this info go under "Examples"? Mario

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Weekly Binaries for Windows are available
by Jeff Lenk 29 Oct '10

29 Oct '10

These evaluation binaries are built with VS2010(release mode) and are packaged in a MSI for easy distribution and installation. They are available in both X86 and X64 versions and install a base version of FreeSWITCH including all default modules along with 8Khz sounds. mod_managed can be used also but you will need to install the DotNet 4 Framework - http://www.microsoft.com/downloads/en/details.aspx?FamilyID=9cfb2d51-5ff4-4… Then enable the module in modules.conf. NOTICE!! Make sure if you install a newer version of the MSI you create backup copies of your /conf folder so you dont lose any configuration settings. New versions of the MSI will be available every week at approx 2am(GMT-6) Sunday morning barring any unforeseen problems. http://files.freeswitch.org/windows/installer/ The MSI will install v9 of the MSCRT libraries to your computer but otherwise makes no modifications other than placing files in your program files\FreeSWITCH folder and adding a "FreeSWITCH" menu entry in your start menu. This is a work in progress and will have frequent modifications if needed to correct any problems. You can also download and build the source code for yourself here. http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code Please enjoy and give feedback! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Weekly-Binaries-for-Windows-a… Sent from the freeswitch-users mailing list archive at Nabble.com.

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progressinband=yes in freeswitch?
by Peter P GMX 29 Oct '10

29 Oct '10

Hello, we have some delays with media setup with Aastra SIP over DECT phones. Aastra recommends using "progressinband=yes" for Asterisk so that the media to the phone is already setup during ringing phase. Are there any parameter in Freeswitch to support that? If yes: As we have registered more than one phone (Aastra SIP/DECT + Snom) to one extension, do we expect any problems in that case when media is already setup to both phones during ringing time? Best regards Peter

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microsecond timestamps
by François Delawarde 29 Oct '10

29 Oct '10

I've just seen that there are some channel variables (Caller-Channel-Created-Time, ...) having timestamps in micro-seconds. It is quite cool to see such resolution, but I was just wondering if there was a practical use for it. Does anyone actually uses such fine granularity in their CDRs or elsewhere? Do carrier environments have such requirements (and why?)? François.

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transfer related question
by Meftah Tayeb 29 Oct '10

29 Oct '10

hi, if i do blind or atended transfer i like my phone to pass through a special dialplan extension befaure thay refer to the other phone any work arround this? because i wanna falback to my phone if the transfer fail using *1 in dp could do it, but using the phone transfer button no idea thank you -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313

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FS incoming calls Queue ACD implement
by Johnson 29 Oct '10

29 Oct '10

Hi FS Group, I am a new vanilla. Just want to realize Queue ACD function in Freeswitch. Is any one know how to configure or create the script to process incoming calls put into defined Queues without using mod_FIFO. Any additional prompting is very appreciated. Johnson -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-incoming-calls-Queue-ACD-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Same caller id comes in freeswitch
by abubacker 29 Oct '10

29 Oct '10

Dear all , I faced an issue which comes rarely , when the freeswitch process the calls in a queue it would give the same customer caller id to the agents $ fifo list <bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:51" bridge_start_epoch="1288330431"> <caller uuid="161fc7e8-e31e-11df-9a13-f1c4ac37d22d" caller_id_name="" caller_id_number="1"></caller> <consumer> <uuid>1b4c7090-e31e-11df-9a28-f1c4ac37d22d</uuid> <outgoing_uuid>c21108c2f348bea840a8bab0dbc3458a</outgoing_uuid> </consumer> </bridge> <bridge fifo_name="test_fifo" bridge_start="2010-10-29 11:03:58" bridge_start_epoch="1288330438"> <caller uuid="164f03b4-e31e-11df-9a14-f1c4ac37d22d" caller_id_name="" caller_id_number="2"></caller> <consumer> <uuid>1c7e4614-e31e-11df-9a2c-f1c4ac37d22d</uuid> <outgoing_uuid>906dab93f19c508ba53ef7628ee7ce3b</outgoing_uuid> </consumer> </bridge> above output says that the caller_id_number 1 and 2 are connected with the agent 1000 and 1001 but in the twinkle it says both 1 and 1 are connected with the agent 1000 and agent 1001 why it is showing bad caller id Note: Don't know whether attachment is allowed ( sorry if it has been prohibited ) -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer: http://www.bksystems.co.in/email-policy

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