October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CudaTel and Freeswitch
by Alex Reid 05 Oct '12

05 Oct '12

I wanted to try an experiament so I have rooted my CudaTel and have shell access. Is there a way that I can create my own dialplans and IVR in freeswitch? I have edited the dialplan.xml file I found, but it doesn't do anything. Anyone with any experiance or idea? Maybe it's just not able to work like that with the cudatel. Thank you.

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Using Lua logging from mod_lua
by JP 05 Oct '12

05 Oct '12

Hi, I am a newbie developer of FreeSWITCH. I have Lua scripts that FreeSWITCH is executing perfectly. But I need to log things from my Lua scripts that I don't want mixed with FreeSWITCH's logs. I looked at ( https://github.com/Neopallium/lualogging) to do this. But how do I make mod_lua in FreeSWITCH link with lualogging. Is there any other solution to my problem? Thanks in advance.

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Re: [Freeswitch-users] uuid_bridge, uuid_broadcast is not working
by Peter Olsson 05 Oct '12

05 Oct '12

Before anything else, update your version to a more current one. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Shahzad Bhatti Skickat: den 5 oktober 2012 14:27 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] uuid_bridge, uuid_broadcast is not working Hi, I am creating a script in Lua to originate 2 calls, bridge them and then play a file using uuid_bridge and uuid_brodcast but unable to get the result and call hangup. my code is as follows: cli = "1111"; caller = "1000"; dest = "1001"; profile= "internal"; param = "{ignore_early_media=flase,origination_caller_id_number="..cli.."}"; api = freeswitch.API(); domain = api:executeString("eval ${domain}"); file = "/usr/local/freeswitch/sounds/test_unset.wav"; session1 = freeswitch.Session(param.."sofia/"..profile.."/".. caller .."%"..domain); session2 = freeswitch.Session(param.."sofia/"..profile.."/".. dest .."%"..domain); uuid1 = tostring(session1:get_uuid()); uuid2 = tostring(session2:get_uuid()); api:executeString("uuid_bridge "..uuid1.." "..uuid2); api:executeString("uuid_broadcast "..uuid1.." "..file.." aleg"); and the fs_cli log is as : 2012-10-05 22:06:57.386448 [NOTICE] sofia.c:6318 Channel [sofia/internal/1001] has been answered 2012-10-05 22:06:57.406452 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001) Running State Change CS_SOFT_EXECUTE 2012-10-05 22:06:57.406452 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1001) State SOFT_EXECUTE 2012-10-05 22:06:57.406452 [DEBUG] mod_sofia.c:588 SOFIA SOFT_EXECUTE 2012-10-05 22:06:57.406452 [DEBUG] switch_core_state_machine.c:241 sofia/internal/1001 Standard SOFT_EXECUTE 2012-10-05 22:06:57.406452 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1001) State SOFT_EXECUTE going to sleep 2012-10-05 22:06:57.406452 [INFO] switch_channel.c:2711 sofia/internal/1001 Flipping CID from "" <1111> to "Outbound Call" <1001%192.168.0.19> 2012-10-05 22:06:57.406452 [INFO] switch_channel.c:2711 sofia/internal/1000 Flipping CID from "" <1111> to "Outbound Call" <1000%192.168.0.19> 2012-10-05 22:06:57.406452 [DEBUG] switch_ivr_bridge.c:1619 (sofia/internal/1000) State Change CS_SOFT_EXECUTE -> CS_HIBERNATE 2012-10-05 22:06:57.406452 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000) Running State Change CS_HIBERNATE 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1000 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/1000) State HIBERNATE 2012-10-05 22:06:57.426425 [DEBUG] mod_sofia.c:222 sofia/internal/1000 SOFIA HIBERNATE 2012-10-05 22:06:57.426425 [DEBUG] switch_ivr_bridge.c:749 (sofia/internal/1000) State Change CS_HIBERNATE -> CS_RESET 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1000 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/1000) State HIBERNATE going to sleep 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000) Running State Change CS_RESET 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/1000) State RESET 2012-10-05 22:06:57.426425 [DEBUG] mod_sofia.c:166 sofia/internal/1000 SOFIA RESET 2012-10-05 22:06:57.426425 [DEBUG] switch_ivr_bridge.c:734 sofia/internal/1000 CUSTOM RESET 2012-10-05 22:06:57.426425 [DEBUG] switch_ivr_bridge.c:741 (sofia/internal/1000) State Change CS_RESET -> CS_SOFT_EXECUTE 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1000 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/1000) State RESET going to sleep 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000) Running State Change CS_SOFT_EXECUTE 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1000) State SOFT_EXECUTE 2012-10-05 22:06:57.426425 [DEBUG] mod_sofia.c:588 SOFIA SOFT_EXECUTE 2012-10-05 22:06:57.426425 [DEBUG] switch_ivr_bridge.c:759 sofia/internal/1000 CUSTOM SOFT_EXECUTE 2012-10-05 22:06:57.426425 [DEBUG] switch_channel.c:2849 (sofia/internal/1001) Callstate Change ACTIVE -> HANGUP 2012-10-05 22:06:57.426425 [NOTICE] switch_ivr_originate.c:1056 Hangup sofia/internal/1001 [CS_SOFT_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2012-10-05 22:06:57.426425 [DEBUG] switch_channel.c:2872 Send signal sofia/internal/1001 [KILL] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001) Running State Change CS_HANGUP 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1001) State HANGUP 2012-10-05 22:06:57.426425 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1001 hanging up, cause: DESTINATION_OUT_OF_ORDER 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1001 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_channel.c:2849 (sofia/internal/1000) Callstate Change ACTIVE -> HANGUP 2012-10-05 22:06:57.426425 [NOTICE] switch_ivr_bridge.c:817 Hangup sofia/internal/1000 [CS_SOFT_EXECUTE] [ORIGINATOR_CANCEL] 2012-10-05 22:06:57.426425 [DEBUG] switch_channel.c:2872 Send signal sofia/internal/1000 [KILL] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1000 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/1000) State SOFT_EXECUTE going to sleep 2012-10-05 22:06:57.426425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000) Running State Change CS_HANGUP 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:816 Send signal sofia/internal/1001 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] switch_core_session.c:816 Send signal sofia/internal/1000 [BREAK] 2012-10-05 22:06:57.426425 [DEBUG] mod_sofia.c:517 Sending BYE to sofia/internal/1001 2012-10-05 22:06:57.446365 [DEBUG] switch_cpp.cpp:1007 sofia/internal/1001 destroy/unlink session from object 2012-10-05 22:06:57.446365 [DEBUG] switch_cpp.cpp:1007 sofia/internal/1000 destroy/unlink session from object 2012-10-05 22:06:57.446365 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000) State HANGUP 2012-10-05 22:06:57.446365 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1001 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2012-10-05 22:06:57.446365 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1001) State HANGUP going to sleep 2012-10-05 22:06:57.446365 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1001) State Change CS_HANGUP -> CS_REPORTING 2012-10-05 22:06:57.446365 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/1001 [BREAK] 2012-10-05 22:06:57.446365 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001) Running State Change CS_REPORTING and the complete fs_cli log is here : http://pastebin.freeswitch.org/19992 I AM USING CentOS 6.2, and FreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000) kindly help me and your response is highly appreciated. Regards Shahzad Bhatti. !DSPAM:506ecf7032761330817986!

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mod_spandsp hylafax dialplan problem
by Tamer Higazi 05 Oct '12

05 Oct '12

Hi people! I wanted to set up hylafax with mod_spandsp. as starting FS, I see FS0 and FS1 2012-10-05 09:42:04.523208 [INFO] mod_spandsp_modem.c:371 Modem [/dev/FS1]->[/dev/pts/1] Ready 2012-10-05 09:42:04.523208 [INFO] mod_spandsp_modem.c:371 Modem [/dev/FS0]->[/dev/pts/0] I did the guide, registering the number of modems, also with "minicom" I tested if the devices are up and running. It works! But I still don't understand how to set up the dialplan. What is mean with x = slot and y = dialed number for incoming faxes..... <action application="bridge" data="modem/a/a"/> the other problem I have, after setting up with "faxsetup" the modems, and started hylafax and freeswitch, I tried to send a file from the console with sendfax. running: sendfax -n -vv -d 1xxxxxxxx /tmp/testX.ps which gives me absolutely no output on the freeswitch console?! What has happened wrong???? Trying localhost [2] (127.0.0.1) at port 4559... Connected to localhost. 220 localhost server (HylaFAX (tm) Version 5.5.1) ready. -> USER root 230 User root logged in. match against (..., 512) rule: offset 0 string = "%!" -- success (result postscript, rule "") Apply DisplayNumber rules to "10613273585121" --> return result "10613273585121" -> FORM PS 200 Format set to PS. -> TYPE I 200 Type set to Image. SEND compressed data, 36409 bytes -> EPRT |1|127.0.0.1|55571| 200 EPRT command successful. -> MODE Z 200 Mode set to ZIP. -> STOT 150 FILE: /tmp/doc62.ps (Opening new data connection). SEND 14845 bytes transmitted (2.5x compression) 226 Transfer complete (FILE: /tmp/doc62.ps). -> JNEW 200 New job created: jobid: 67 groupid: 67. -> JPARM FROMUSER "root" 213 FROMUSER set to "root". -> JPARM LASTTIME 000259 213 LASTTIME set to 000259. -> JPARM MAXDIALS 12 213 MAXDIALS set to 12. -> JPARM MAXTRIES 3 213 MAXTRIES set to 3. -> JPARM SCHEDPRI 127 213 SCHEDPRI set to 127. -> JPARM DIALSTRING "10613273585121" 213 DIALSTRING set to "10613273585121". -> JPARM NOTIFYADDR "root@localhost" 213 NOTIFYADDR set to "root@localhost". -> JPARM VRES 98 213 VRES set to 98. -> JPARM PAGEWIDTH 215 213 PAGEWIDTH set to 215. -> JPARM PAGELENGTH 279 213 PAGELENGTH set to 279. -> JPARM NOTIFY "none" 213 NOTIFY set to "none". -> JPARM PAGECHOP "default" 213 PAGECHOP set to "default". -> JPARM CHOPTHRESHOLD 3 213 CHOPTHRESHOLD set to 3. -> JPARM DOCUMENT /tmp/doc62.ps 200 Added document /tmp/doc62.ps as docq/doc62.ps.67. -> JSUBM 200 Job 67 submitted. request id is 67 (group id 67) for host localhost (1 file)

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Community Members on Austin Area
by João Mesquita 05 Oct '12

05 Oct '12

Anyone on the Austin,TX area that is willing to meet? Are there many of you around? Regards, João Mesquita

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shared line
by Darcy Primrose 04 Oct '12

04 Oct '12

Does any one know if it is possible to make a line private on the shared line appearance. When a line is in active mode and another set has the shared line key pressed, you join the converstation, this is not normal mode of operation for a key system. Any assistance or guidance would be appreciated. Darcy Primrose

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DTMF delay when using FreeSWITCH
by Emrah 04 Oct '12

04 Oct '12

Hi guys, I am comparing this with an Asterisk and FreeSWITCH installation, using the same route, same codecs, same carrier, same phones and same servers… :P I experience a delay when pressing DTMFs on the line that uses FreeSWITCH. I am estimating the delay to be around 500 ms. What are the settings I can fine tune to avoid this? All the best, Emrah

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New Node.js ESL Library
by Chad Engler 04 Oct '12

04 Oct '12

Hello, I couldn't find a Node.js library for FreeSWITCH Event Sockets that really followed the reference: http://wiki.freeswitch.org/wiki/Event_Socket_Library#Reference; so I created on to implement it as closely as possible. Feedback is appreciated: https://github.com/englercj/node-esl Thanks, Chad Engler Web Programmer PATLive 1.800.775.7790 x746 Chad.Engler(a)patlive.com <mailto:Chad.Engler@patlive.com> Hosted Communications | Friendly Service www.patlive.com <http://www.patlive.com/>

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Limit maximum number of outgoing calls of a sip user
by mbo 04 Oct '12

04 Oct '12

How can I limit the number of outgoing sip calls for a sip user. I'm using the following configuration: <include> <user id="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="0000"/> </params> <variables> <variable name="max_calls" value="2"/> </variables> </user> </include> However the variable max_calls is set to 2, I can do 5 outbound calls from a registered sip client. Is this parameter only for inbound calls? Is there another parameter for outbound calls? Thanks Markus

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Google talk no RTP connection
by Tor Petterson 04 Oct '12

04 Oct '12

Hi I am having a problem with establishing voice calls to Google talk. >From the logs I can see that Freeswitch tries to tell google talk the IP address and port to talk to but Google returns an error. I have opened all RTP ports in my firewall and as far as I can see Google never tries to access the RTP port >From Freeswitch to Google: <iq xmlns="jabber:client" from="AAA(a)AAA.AAA/talk5FCFAFC0" to="BBB(a)BBB.BBB/Talk.v1040F85EB97" type="set" id="301"> <jin:jingle xmlns:jin="urn:xmpp:jingle:1" action="transport-info" sid="6694447203"> <jin:content name="audio" creator="initiator"> <p:transport xmlns:p="http://www.google.com/transport/p2p"> <candidate name="rtp" address="94.X.X.X" port="25150" username="URR6q0MaqN1fhc6I" password="xxnqT56PyyjkOnGz" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </p:transport> </jin:content> <jin:content name="video" creator="initiator"> <p:transport xmlns:p="http://www.google.com/transport/p2p"> <candidate name="video_rtp" address="94.X.X.X" port="25152" username="TEMYCYid3JW95dDQ" password="Iok84tXaLWUhr2Ib" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </p:transport> </jin:content> </jin:jingle> </iq> <iq xmlns="jabber:client" from="AAA(a)AAA.AAA/talk5FCFAFC0" to="BBB(a)BBB.BBB/Talk.v1040F85EB97" type="set" id="302"> <ses:session xmlns:ses="http://www.google.com/session" type="candidates" id="6694447203" initiator="AAA(a)AAA.AAA/talk5FCFAFC0"> <ses:candidate name="rtp" address="94.X.X.X" port="25150" username="URR6q0MaqN1fhc6I" password="xxnqT56PyyjkOnGz" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> <ses:candidate name="rtcp" address="94.X.X.X" port="25151" username="4ovle7muLeASgwrD" password="UFUDAQiYfs8TGFiA" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> <ses:candidate name="video_rtp" address="94.X.X.X" port="25152" username="TEMYCYid3JW95dDQ" password="Iok84tXaLWUhr2Ib" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> <ses:candidate name="video_rtcp" address="94.X.X.X" port="25153" username="xlZZa823HO2DSwjq" password="KtpFMcFoYqvgj4hG" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> </ses:session> </iq> >From Google to Freeswitch: <iq type="error" to="AAA(a)AAA.AAA/talk5FCFAFC0" id="301" from="BBB(a)BBB.BBB/Talk.v1040F85EB97"> <jin:jingle action="transport-info" sid="6694447203" xmlns:jin="urn:xmpp:jingle:1"> <jin:content name="audio" creator="initiator"> <p:transport xmlns:p="http://www.google.com/transport/p2p"> <candidate name="rtp" address="94.X.X.X" port="25150" username="URR6q0MaqN1fhc6I" password="xxnqT56PyyjkOnGz" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </p:transport> </jin:content> <jin:content name="video" creator="initiator"> <p:transport xmlns:p="http://www.google.com/transport/p2p"> <candidate name="video_rtp" address="94.X.X.X" port="25152" username="TEMYCYid3JW95dDQ" password="Iok84tXaLWUhr2Ib" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </p:transport> </jin:content> </jin:jingle> <error code="501" type="cancel"></error> </iq> <iq to="AAA(a)AAA.AAA/talk5FCFAFC0" id="302" type="error" from="BBB(a)BBB.BBB/Talk.v1040F85EB97"> <ses:session type="candidates" id="6694447203" initiator="AAA(a)AAA.AAA/talk5FCFAFC0" xmlns:ses="http://www.google.com/session"> <ses:candidate name="rtp" address="94.X.X.X" port="25150" username="URR6q0MaqN1fhc6I" password="xxnqT56PyyjkOnGz" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> <ses:candidate name="rtcp" address="94.X.X.X" port="25151" username="4ovle7muLeASgwrD" password="UFUDAQiYfs8TGFiA" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> <ses:candidate name="video_rtp" address="94.X.X.X" port="25152" username="TEMYCYid3JW95dDQ" password="Iok84tXaLWUhr2Ib" preference="1.0" protocol="udp" type="local" network="0" generation="0"></ses:candidate> </ses:session> </iq>

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