- FreeSWITCH-users - FreeSWITCH Mailing Lists

JIRA
by Brian West 10 Mar '16

10 Mar '16

FreeSWITCHers, FYI If I close an old JIRA you still care about, by all means REOPEN IT, be attentive and help drive it thru the process, I thank everyone for opening JIRAs on issues, but you'll need to provide assistance to push them over the finish line. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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mod_logfile change logging path on windows
by Alexander Haugg 10 Mar '16

10 Mar '16

Hi, what is to do to change the log folder to d:, or to a windows file share, or another computer? I try some string formats and file urls and complete my changes with reload mod_logfile, but my changes in the "logfile" property will ignored. Thanks a lot!

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MOD_VERTO Documentation
by Shlomi Schwartz 10 Mar '16

10 Mar '16

Hi all, Hello, I'm trying to implement mod_verto JSONRPC on IOS, is there any documentation regarding the dialogParams on verto.invite method? The only documentation I found was : https://freeswitch.org/confluence/display/FREESWITCH/mod_verto Thanks :) -- Shlomi Schwartz R&D Technical Leader T: +972-74-700-4511 <http://www.linkedin.com/company/164748> <http://twitter.com/liveperson> <http://www.facebook.com/LivePersonInc> We Create Meaningful Connections <http://info.liveperson.com/201602/liz-welch/liveperson-robert-locascio-on-d…> -- This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this on behalf of the addressee you must not use, copy, disclose or take action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply email and delete this message. Thank you.

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mod_amqp for Windows
by Gregor Nanger 10 Mar '16

10 Mar '16

Hi! Does anybody knows if module mod_amqp is available for windows if I build by myself? Best regards, Gregor

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Mod_sms and script to send to Twilio
by David Rodrigues 09 Mar '16

09 Mar '16

Thank you. I'm new to the mailing list. I had my mail client set for html. Perhaps that was the problem. Here is the Python script: #/usr/bin/env python import re import pprint import sys import urllib import subprocess from freeswitch import * ACCOUNTSID = 'my-account-sid' AUTHTOKEN = 'my-auth-token' PHONENUMBER = '+my-source-phone-number' def chat(message, args): to_phone = '+1' + re.sub('\D', '', urllib.unquote(message.getHeader('to_user'))).lstrip('+1') body = message.getBody() subprocess.call("curl -X POST 'https://api.twilio.com/2010-04-01/Accounts/'" + ACCOUNTSID + "/Messages.json --data-urlencode 'To=" + to_phone + "' --data-urlencode " + "'From=+" + PHONENUMBER + "' --data-urlencode 'Body=" + body + "' -u " + ACCOUNTSID + ":" + AUTHTOKEN, shell=True) Here is my chatplan XML file: <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="100"> <condition field="to" expression="^(.*)$"> <action application="python" data="sms"/> </condition> </extension> </context> </include> And here is the error (keep in mind that the script actually works fine. I receive the SMS as expected): 2016-03-05 15:17:54.040913 [DEBUG] mod_python.c:286 Finished calling python script 2016-03-05 15:17:54.040913 [WARNING] sofia_presence.c:221 Not sending to local box for (XXX)%CX%A0XXX-XXX(a)my.sipdomain.com <-- (Note that I substituted Xs for the actual phone digits just for this post) 2016-03-05 15:17:54.040913 [ERR] sofia_presence.c:272 Chat proto [sip] from ["d100" <sip:100@my.sipdomain.com>;tag=0b2cec63] to [(XXX)%CX%A0XXX-XXX(a)my.sipdomain.com] Test sms Nobody to send to: Profile internal -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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'make' Error: fs_cli-esl.o
by VM Knott 09 Mar '16

09 Mar '16

Hello. I'm attempting to install freeswitch-1.6.6 from (freeswitch-1.6.6.tar.gz) and when I attempt to perform the 'Make', I get the following error: --------------- CC fs_cli-esl.o CC fs_cli-esl_config.o CC fs_cli-esl_event.o CC fs_cli-esl_threadmutex.o CC fs_cli-fs_cli.o libs/esl/fs_cli.c: In function "main": libs/esl/fs_cli.c:1351:6: error: "variable use_history_file" set but not used [-Werror=unused-but-set-variable] int use_history_file = 0; ^ cc1: all warnings being treated as errors make[2]: *** [fs_cli-fs_cli.o] Error 1 make[2]: Leaving directory `/home/ec2-user/src/freeswitch-1.6.6' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/ec2-user/src/freeswitch-1.6.6' make: *** [all] Error 2 --------------- I'm hoping that I'm overlooking something obvious? Any thoughts, guidance, or suggestions would be greatly appreciated. VMK

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Thursday FreeSWITCH Bug Hunt
by Ítalo Rossi 09 Mar '16

09 Mar '16

FreeSWITCHers, Join us tomorrow 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? https://conference.freeswitch.org/vc/#/?autocall=888 What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

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mod_curl or mod_httapi
by John Nash 09 Mar '16

09 Mar '16

I need to call an http URL with some Post parameters from lua script and set some channel variables based on response data received. Which module will be best from performance point of view?

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Freeswitch behind NAT with 2 external providers: No audio
by Peter Steinbach 09 Mar '16

09 Mar '16

Hello, I have the following scenario * Freeswitch is behind NAT with external profile with ext_sip_ip and ext_rtp_ip * Echo from an external SIP provider test inbound works with audio * Calling outbound to the same external SIP provider works with audio * *Receiving a call from the external SIP provider and bridging it to another number at the same external SIP provider does not have audio* * Receiving a call from the external SIP provider and bridging it to a local Patton PRI works with audio I also set up another (modified NAT) profile * like external profile - but with a different ext_rtp_ip * calling outbound to via NAT profile to another SIP provider works with audio * *Receiving a call from the external SIP provider (external profile) and bridging it to another number ****at the another external SIP provider ***(nat***profile)** does not have audio* * Tcpdump btw. does not show an attempt of Freeswitch to issue any outbound RTP packet. I can see only SIP messages towards the outbound SIP provider* * So I think, there is a problem with briding a natted external call to another natted external call, both with ext-rtp-ip. It tried this on an older Freeswitch (1 year old) and with a more recent Freeswitch (1 month old), both behave the same. Same scenario with a non-natted Freeswitch works fine. Btw. Freeswitch logs do show RTCPpacket not written right after bridginbg the call. Question: Does anybody have a clue, what could cause the problem? Maybe there is any parameter in the profiles, that may help? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Two outbound ESL connections
by Dan Barua 09 Mar '16

09 Mar '16

Funny you mention this, I am working on something similar. I found this works but it feels like a bit of a hack: api uuid_park 932506bd-7208-446c-aa4e-d91848df0ff3 api sched_api +0.01 none uuid_transfer 932506bd-7208-446c-aa4e-d91848df0ff3 'socket:127.0.0.1:8085 async full' inline (Exit proxy socket immediately) I have tried the custom event route, but as soon as the proxy ESL connection exits then the call terminates. Hope that helps -Dan Hello there, > > I have a setup where I would like FreeSWITCH to make an outbound ESL > connection to a proxy-like service. The proxy-like service should then make > FreeSWITCH open another outbound ESL connection to a different service that > will handle the call. > > The proxy-like service should be as transparent as possible. Ideally, it > should be possible to 'skip' it, and point FreeSWITCH directly to the > service that handles the call. > > Currently, I have a dialplan that looks like this: > > -- > <action application="set" data="continue_on_fail=true"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="ringback=%(1000,4000,425)"/> > <action application="socket" data="127.0.0.1:3000 async full"/> > <action application="hangup" data="FACILITY_NOT_IMPLEMENTED"/> > -- > > The proxy-like service listenens on port 3000. It will order FreeSWITCH to > execute the following command: > > socket 127.0.0.1:3001 async full > > This works, somewhat. I would like the proxy-service to terminate the > connection as fast as possible. However, if I send the exit-command[1] > before the call is bridged to the hangup from the XML dialplan is executed. > If the call is bridged when I send the exit-command, it works fine. > > I would like the service on 3001 to emit a custom event, which the > proxy-like service listens for and terminates the ESL connection. Currently > I need to sleep a little while and then terminate the ESL connection. > > How do I keep the hangup from being executed, when there is an open > outbound ESL connection? > > I currently use FreeSWITCH Version > 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 > 18:59:51Z 64bit). > > I hope this makes sense. > > Regards, Asbjørn > >

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