- FreeSWITCH-users - FreeSWITCH Mailing Lists

Change default connection port
by George Ionescu 25 Jul '13

25 Jul '13

Hi guys, I have 2 csipsimple terminals registered on a non-default SIP port to FS server. I've changed the default SIP port from 5060 to 8095. When I'm trying to send messages from device1 to device2 on device2 the message appears from device1@<domain.com<mailto:device1@%3cdomain.com>> and not from device1@<domain.com:8095<mailto:device1@%3cdomain.com:8095>>. In that moment I can't reply back because the contact was re-wrote somehow. If I'm un-checking the "Allow contact re-write" option from csipsimple the registration won't work anymore. The entire configuration is built on Bluebox + Freeswitch (latest versions) . Thanks George

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Prerequisites of enabling freeswitch 1.2.7 to use gsm modem
by Ashish Mishra 25 Jul '13

25 Jul '13

Hello. Kindly help me with the prerequisites of enabling freeswitch to use gsm modem...i have already compiled and installed freeswitch on my ubuntu 12.04 LTS machine... Thank you... On Jul 25, 2013 4:42 AM, "Michael Collins" <msc(a)freeswitch.org> wrote: Hi Raghu, I would put the setInputCallback before the sleep, that way if the avmd is detected while you're sleeping then it should work. -MC On Wed, Jul 24, 2013 at 1:30 PM, Raghu <srraghu45(a)gmail.com> wrote: > I am new to freeswitch. I am trying to send a message to voicemail using > avmd for voicemail detection, here is the code which i am using with > originate command: > > from freeswitch import * > > # WARNING: known bugs with hangup hooks, use with extreme caution > def hangup_hook(session, what): > > consoleLog("info","hangup hook %s!!\n\n" % what) > return > > def input_callback(session, what, obj): > > if (what == "dtmf"): > consoleLog("info", what + " from callback " + obj.digit + "\n") > return > elif (what == "event" and session.getVariable('avmd_detect') == > "TRUE"): > consoleLog("info", "Voicemail Detected\n") > return "true" > > def handler(session, args): > > session.answer() > session.execute("avmd", "start") > session.execute("sleep", "25000") > callback = session.setInputCallback(input_callback) > consoleLog("info","callback returned %s!!\n\n" % callback) > while session.getVariable('avmd_detect'): > session.execute("speak", "flite|kal|'Hi how are you'") > > session.execute("avmd", "stop") > session.setHangupHook(hangup_hook) > session.hangup() #hangup the call > > > I can see avmd getting started but it does detect any beep. Is there a > simpler way to check if avmd is working without using 'originate'? > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Setting User to User Call Control Information (UUI)
by Jan Riedinger 25 Jul '13

25 Jul '13

Is there an easy way to set User to User Call Control Information (UUI) by FreeSwitch according http://potaroo.net/ietf/ids/draft-ietf-cuss-sip-uui-10.txt? UUI is used in Germany for emergency calls. Thank you in advance Jan

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FreeSWITCH IP wrong when make outbound call
by Khue Nguyen Minh 25 Jul '13

25 Jul '13

Hi all, I have a problem when make outbound call from FS. I have two networks interface and two ip address on my server. Assume are IP1 and IP2. In FS configuration (internal.xml, external.xml, event_socket.conf.xml, etc...), I set all parameter relate ip address is IP1. But, when I make outbound call via ESL, ip address of FS is become IP2. Please guide me fix this problem. Thanks, Khue.

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Debian or Ubuntu
by Richard Mace 25 Jul '13

25 Jul '13

Hi, I am getting the feeling that the majority of people on the list are using Debian over Ubuntu for their freeSWITCH boxen, and whilst I understand it to a certain extenent (speed), I was just wondering, if this was the case, why people would choose Debian over Ubuntu (LTS) as the life of the distro will be significantly longer with Ubuntu? (5 years) Regards Richard

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Help Request: Vestec ASR Documentation
by Michael Collins 24 Jul '13

24 Jul '13

Dear FreeSWITCHers, I'm looking for two or three persons to assist with some Vestec ASR documentation. If you have Vestec ASR working and can help us document the steps needed to get a basic speech rec app up and running please contact me off list. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Hello
by dravion.smith＠gmx.net 24 Jul '13

24 Jul '13

I just want to say Hello !

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BW estimation for VoIP trunk
by Adelia C. 24 Jul '13

24 Jul '13

Need help estimating BW needs. This is by far the most helpful forum I use so I am turning to you, good people of Freeswitch. I posted elsewhere with no success. Is there a site or book you found helpful for your estimates? I found a few sites/formulas but no "tutorial" on how to get from the info I have (mainly MOU, max. concurrent sessions, codec, average call duration, usual day/week traffic pattern) to an actual BW number. Where do I start? Am I asking the wrong question, do I use wrong parameters? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/BW-estimation-for-VoIP-trunk-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Hangup causes
by Muhammad Naseer Bhatti 24 Jul '13

24 Jul '13

Got a little confusion here. FreeSWITCH log: http://pastebin.freeswitch.org/21226 SIP trace: http://pastebin.freeswitch.org/21227 Both were taken once after each other but other than the timing, everything else remains the same. This is a vanilla install for FreeSWITCH Version 1.5.5b+git~20130724T035836Z~3ae87091e1 (git 3ae8709 2013-07-24 03:58:36Z). Just dialing a non existent number (98871002) from a registered user 1001. The hangup cause returned by the switch shows # mod_sofia.c:463 Channel sofia/internal/1001@10.211.55.3:50601 hanging up, cause: NORMAL_CLEARING However the SIP capture shows SIP/2.0 480 Temporarily Unavailable. This 480 is also showing by the switch mod_sofia.c:597 Responding to INVITE with: 480. I know the mapping is done by FreeSWITCH. In this case why 480, returned Normal Temporary Failure while the switch shows NORMAL_CLEARING? Is this because Q.850 returned is 16 and FreeSWITCH mapping for 16 is NORMAL_CLEARING? Or is there something I am missing. -- Thanks, Muhammad Naseer Bhatti

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ClueCon Weekly Community Conference Call
by Michael Collins 24 Jul '13

24 Jul '13

Hello all! On https://wiki.freeswitch.org/wiki/FS_weekly_2013_07_24today's call we have a few news items and then we'll be taking a look at some simple C programming examples using channel variables. The goal is to show how easy it is to tinker with FreeSWITCH. Talk to you soon! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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