July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

error in compiling freeswitch 1.2.7
by Ashish Mishra 25 Jul '13

25 Jul '13

hello, i have downloaded freeswitch installed the prerequisites: gnumake automake autoconf libtool gcc wget libncurses bzip2 and also have successfully done the following steps: run the bootstrap.sh edit modules.conf run the configure script BUT WHEN TYPING THE COMMAND make && make install i am getting the following error ./.libs/libfreeswitch.so: undefined reference to `inflateReset' ./.libs/libfreeswitch.so: undefined reference to `inflateEnd' ./.libs/libfreeswitch.so: undefined reference to `inflateInit2_' ./.libs/libfreeswitch.so: undefined reference to `inflate' ./.libs/libfreeswitch.so: undefined reference to `zlibVersion' ./.libs/libfreeswitch.so: undefined reference to `inflateInit_' collect2: ld returned 1 exit status make[1]: *** [freeswitch] Error 1 make: *** [all] Error 2 kindly help me Ashish

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Remote media gateway
by Ashwin Rath 25 Jul '13

25 Jul '13

Is there a way to configure FreeSWITCH such that it places an external media gateway in the media path when it acts like an SBC ? -- Ashwin Kumar Rath

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Restarting sofia
by Alex Lake 25 Jul '13

25 Jul '13

Is there a way to quickly restart all sofia gateways such that acl.conf will be re-read?

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Freeswitch not responding to sip messages from call out leg
by Zvi Agmon 25 Jul '13

25 Jul '13

Hello, We are using FS to bridge incoming call to an outbound telephone through a gateway provider. The bridge is created OK, but we found that after 3 minutes FS is no longer responding to sip messages from the call out provider. Thus, in case the provider sends sip-Invite message, it gets no response from FS and disconnect the call out leg. Note that FS is also not aware of the Bye message so leg A is still on the line and when it hangs up FS send Bye message to leg B and get as response "Unknown Dialog". I set the siptrace on and see that no packets arriving sofia (when the calling leg hanged up I do see the packets). Also I set all sofia components log level to 7 - this is what was printed out after the call answered: 2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel [sofia/external/97249126793] has been answered 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291 sofia/external/97249126793 process sched_hangup +7199 DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45 alloted_timeout: sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45 alloted_timeout) 2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6 switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE(a)81.201.82.45) to run at 1373275728 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/external/97249126793 [BREAK] 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal sofia/internal/972522977131(a)voxbone.com [BREAK] nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK (46298196) nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593 (git 0a2e713 2013-03-07 05:40:51Z) On ubuntu release 12.04 Please help. Thanks Zvi Agmon Best regards Zvi Agmon Lexifone email: zvi(a)lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109

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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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