August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re- BT SIP Testing require FreeSWITCH main developer's statement of compliance
by Andrew Keil 03 Aug '15

03 Aug '15

I guess this is targeted at Anthony, however Brian or Michael may also be able to answer these. I am currently undertaking BT's SIP compliance testing on behalf of my client in the UK (on the latest production release of FreeSWITCH 1.4.20) and BT have requested the following off the software supplier (which could also be in the form of documentation, however if you simply state "Yes" for each of these then that should be enough): 1) RFC 3261 compliance a. I have already pointed BT to https://freeswitch.org/confluence/display/FREESWITCH/Specifications 2) RFC 3550 compliance 3) RTP clock-source accuracy (RTP clock sources shall be accurate to within 66ppm (ie. Shall not drift from a stratum 0 clock source by more than 20ms over the course of a 3 minute call) 4) When calculating the jitter estimate to be sent in a receiver report the variable holding this data shall be initialised to zero before the first incoming RTP packet is processed (i.e. the "s->jitter" variable in RFC 3550 [8] Appendix A.8) 5) RFC 2833 compliance a. I have already pointed BT to https://freeswitch.org/confluence/display/FREESWITCH/Specifications 6) RFC 4028 keep-alive timer compliant 7) Feature support: a. Call-hold reject b. RFC 3264 compliance c. Call-hold black-holing 8) G.711 Packet Loss Concealment (as described in G.711 Appendix I) Like I said above BT are just looking for a statement to say these are supported inside FreeSWITCH from the developers of FreeSWITCH. That is why a simple "yes" will suffice. One idea is to perhaps add all of these (the ones that are missing) to the https://freeswitch.org/confluence/display/FREESWITCH/Specifications page. Just an idea. Thanks in advance, Andrew Keil Visytel Pty Ltd

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webrtc/Sip issue
by Chris Mandra 03 Aug '15

03 Aug '15

Hi guys - one of my users keeps getting this error in fs_cli when he tries to connect via Chrome (web-rtc) on a macbook running yosemite. 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!] I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this: ********************* Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected sc-webrtc.js:931 _rejected called ******************* Any ideas? We're using latest FS and sip.js thanks, chris

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[ANN] openerp-freeswitch-connector new tagged release 20150803
by Trever L. Adams 03 Aug '15

03 Aug '15

Hello Everyone, I know several people had issues with this FreeSWITCH/OpenERP connector. Unfortunately, I have been out of communication for several months. I believe I have enhanced the documentation and fixed bugs and this release should work for many people now. Notible changes: * Provide documentation on obtaining and installing FreeSWITCH ESL python module. * Better documentation * Click to dial caller ID name clean-up * freeswitch_click2dial/scripts/get_caller_name.py works as a wsgi script, preferably on OpenERP server. See documentation in the script. * patches-for-external/README.md now clearly states that the program must be started AFTER OpenERP I do not know if this works with Odoo 8.0. I do know it works with OpenERP 7 as found in Fedora 23. https://github.com/treveradams/openerp-freeswitch-connector Anyone who would be willing to take pictures of the various aspects of configuring this and using it as found in the documentation and willing to make them available under some friendly license, I will add some wiki (github) and/or FreeSWITCH confluence entries based on the images and documentation. If not, I will do it myself when I have time. Thank you, Trever

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Passing Dialed Number to User Agent
by Rick Ewart 03 Aug '15

03 Aug '15

Hello. I am a relative newbie with FreeSwitch. Have been reading the FreeSWITCH 1.2 book and have managed to setup things and get them working without too much issue. I am basically trying to setup FreeSwitch to act as an intermediary (softswitch) between my carriers and my client's PBXs. I have the PBXs registering as extensions and am able to make calls and direct DIDs into them. The one thing I seem to be struggling with (at the moment anyhow) is how to pass the dialed number (i.e. DID dialed) to the internal extension I am transferring the call to. Because the one user will have multiple DIDs assigned to them and their system (an Asterisk PBX) needs to route the incoming call appropriately. But the "dialed number" being passed to the Asterisk PBX is the extension being called for the transfer. I have spent hours googling for an answer and never seem to get quite what I need. Seems I need to set a variable, but what I have done so far isn't working. Below is my definition for the DID. I am wondering if this is the correct way to go about this, or if I should be setting these up different? I do want the clients to register to my switch, which is why I have them setup as internal user extensions. Thanks in advance for any help you can provide. Rick <include> <extension name="test_did"> <condition field="destination_number" expression="^(19545551212)$"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="dialed_number=19545551212"/> <action application="transfer" data="1100 XML default"/> </condition> </extension> </include>

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FreeSWITCH Automated Testing
by Brian West 03 Aug '15

03 Aug '15

FreeSWITCHers, So after a bit discussion, a case of red bull, an approach to automating functional testing has now solidified. Audio Path is verified via DTMF in-band, Video Path is verified by using QR codes. Here are two VERY crude examples: https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0 https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0 Input? -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com 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_local_stream moh confusion
by Lappe, Adam 03 Aug '15

03 Aug '15

Hello, i am testing the latest 1.4.19 Version of FreeSWITCH. Currently we are running an old 1.2.7 Version. Everything seems to work fine, but there is 1 error that is very confusing: When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated. All I see is this error line: [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]! I don't use the local_stream module. freeswitch@internal> module_exists mod_local_stream false This error does not exists with the old version. Is this a bug, or am I missing something? Thanks in advance, Adam

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Can't make ClueCon? Want a chance to remote in?
by Anthony Minessale 03 Aug '15

03 Aug '15

Anyone who can call in to our video conf between 9am and 10am This Tuesday. Email consulting(a)freeswitch.org and register for a chance to participate. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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How to track call changes while running javascript?
by Thomas 01 Aug '15

01 Aug '15

Hello list! Does anyone know if there is a way to track changes to the call state, most notably a hangup, while executing Javascript? We're using JS to set up a call on a device connected to Freeswitch through portaudio. That call is initiated with sip/sofia executing the JS in the dialplan. If the SIP leg hangs up while the JS is running, nothing happens until the JS terminates. I wanted to check from the script, but haven't found a way to do that. "session.state" doesn't change, hangup hook doesn't fire, and without setTimeout or something similar I didn't find a way to "background" the JS. Is there any way to detect a hangup during JS execution? Cheers! Thomas

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Unpark call
by Carlos Ruiz Díaz 01 Aug '15

01 Aug '15

Hi, I'm looking for a way to unpark a call remotely. I have this scenario that involves Kamailio as registrar and proxy, and I'm sending calls to FS whenever an user is busy and can't pick up a call. This call is parked by FS. When Kamailio detects the user in question got available, I want to unpark the previous call and connect it to the original destination. Any ideas on how to do this in a clean way? Regards, -- Carlos http://caruizdiaz.com

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