July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Disk error / database error help
by clive engelberg 06 Jul '15

06 Jul '15

Hi Guys. I have had this happen a few times already, where Freeswitch seems to have a database error. This starts off slowly, and within a short time, it completely fills the disk drive with log error files. the log files look like this: 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [disk I/O error] BEGIN EXCLUSIVE 2015-07-01 15:19:35.470881 [CRIT] switch_core_sqldb.c:1565 ERROR [disk I/O error] 2015-07-01 15:19:35.470881 [ERR] switch_core_sqldb.c:583 NATIVE SQL ERR [cannot commit - no transaction is active] Any ideas what is causing this. I suspect maybe a hardware disk error.. not sure. Thanks in advance Clive ____________________________________________________________ South Africas premier free email service - www.webmail.co.za

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New module mod_smpp merged into FreeSWITCH
by William King 04 Jul '15

04 Jul '15

Just merged it in today into the master branch. It's not in the FS unstable debian repo yet(have to get libsmpp34 packaged first), but here is a quick write up of how to use it: https://quentustech.com/smpp-support-in-freeswitch.html -- William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king(a)quentustech.com

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Quality VOIP provider with SMS support
by Mitch Capper 04 Jul '15

04 Jul '15

After using several providers from vitelity to flowroute and callwithus, I figured it was time to move on from vitelity given some very bad SMS support as of late. Looked at Nexmo, voxbone, and a few others (noticed nexmo mentioned in the new mod_snmpp thread). I figured I would see what people suggest. As a bonus right now do a lot of faxing with FS and flowroute and getting some great results for both incoming and outgoing. While happy to keep using many providers for different things if able to converge on one all the better:) ~Mitch

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OpenSIPS Workshop @ClueCon August 3, 2015
by Alex Goulis 03 Jul '15

03 Jul '15

It's almost that time again! The OpenSIPS team is coming to ClueCon 2015! The OpenSIPS Workshop @ClueCon has been scheduled for *3rd of August 2015, at InterContinental Chicago, Chicago, IL, USA*. The OpenSIPS Workshop is a new format that will focus more on providing attendees with more of a technical approach to integrating OpenSIPS rather than case studies - the sessions will provide working setups (with explanations) for OpenSIPS in different scenarios . The Workshops will cover new OpenSIPS release and features, FreeSwitch integration, Edge Proxy setup, Asynchronous I/O support and Fraud prevention. The Official schedule is still under construction: http://www.opensips.org/Community/Workshop-2015Chicago-Schedule But we're happy to announce the following workshop leaders: Bogdan-Andrei Iancu - Founder OpenSIPS project Dan Christian Bogos - ITsysCOM Vlad Paiu - OpenSIPS Project Flavio Goncalves - SIPPulse Alex Goulis - CID(name) Pete Kelly - SourceVox Whether you are a beginner or a seasoned professional, the OpenSIPS Workshop @ClueCon has something for you. For beginners, get answers to questions like: * how to extend, expand and enhance FreeSwitch with OpenSIPS * why OpenSIPS is the right choice for your network * what are the capabilities, features and limits of OpenSIPS Seasoned professionals can: * Learn about new modules and features for the 2.x release * Discuss advanced topics * Discuss the development roadmap for OpenSIPS The registration fees (per person) are: $199 for those attending only the OpenSIPS Workshop $149 for those attending ClueCon as well or having a discount code (*CLUECON2015*) or are registering 3 or more participants Space is limited, so don't wait too long to register. More information can be found at: http://www.opensips.org/Community/Workshop-2015Chicago We're looking forward to seeing you in Chicago! - The OpenSIPS Team

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Asterisk canreinvite (directmedia) equivalent in freeswitch
by Ričardas Stoma 03 Jul '15

03 Jul '15

Does freeswitch have anything similar to canreinvite (asterisk sip setting)? I know there is bypass_media but it does not work in the same way as canreinvite in asterisk. It looks like bypass_media always tells freeswitch to send media directly between originator and terminator even though they do not have common codecs. I want to use bypass_media only when leg A and leg B have common codecs. If they do not have then freeswitch should stay in media path and handle codec transcoding. Is there an easy way to achieve this in freeswitch?

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 03 Jul '15

03 Jul '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Why ringback is not generated?
by Sergey Safarov 03 Jul '15

03 Jul '15

In dialplan I set ringback variable <extension name="setRingback" continue="true"> <condition> <action application="set" data="ringback=${us-ring}"/> </condition> </extension> And diaplan terminate call on extension <extension name="agent_login"> <condition field="${destination_number}" expression="^agent_login$"> <action application="set" data="ignore=${callcenter_config(agent set status ${caller_id_number}@${sip_to_host} 'Available')}" inline="true"/> <action application="ring_ready"/> <action application="pre_answer"/> <action application="sleep" data="2500"/> <action application="answer"/> <action application="sleep" data="700"/> </condition> </extension> But ringback is not generaded. Why is it posible? Sergey

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Make current failing on Certificates
by Guillermo Ruiz Camauer 03 Jul '15

03 Jul '15

I just tried to "make current" and got this: error: server certificate verification failed. CAfile: /etc/ssl/certs/ca-certificates.crt CRLfile: none while accessing https://stash.freeswitch.org/scm/fs/freeswitch.git/info/refs fatal: HTTP request failed make[1]: *** [update] Error 1 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 -- Guillermo Ruiz Camauer

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mod_sms duplicate messages
by sangdrax8 03 Jul '15

03 Jul '15

I am trying to test out mod_sms for the chatplans and it seems to be sending two messages when I call a send. Can someone tell me if there is some configuration I am doing wrong, of if I need to file a Jira bug? If I unload the mod_sms module, only one messages i sent as expected but I was looking for the chatplan functionality. I am currently testing on Version 1.4.19 git 73f45e3, as this was the suggested branch for production usage. my chatplan is very basic: <include> <context name="public"> <extension name="send"> <condition> <action application="send"/> </condition> </extension> </context> </include> when I send a message I get the following in the CLI and two sip messages are sent. 2015-07-02 18:03:38.950750 [INFO] mod_sms.c:336 Processing text message 19995550008->19995550001 in context public Chatplan: 19995550001 parsing [public->send] continue=false Chatplan: 19995550001@demo Absolute Condition [send] Chatplan: 19995550001@demo Action send() With siptrace on I can see two send events that are identical but happen less than a millisecond apart. send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965479: send 1173 bytes to tls/[192.168.0.40]:54156 at 18:03:38.965878:

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Video playback in 1.7
by Stanislav Sinyagin 02 Jul '15

02 Jul '15

the newest 1.7 freeswitch successfully played an .mp4 file with "playback" application, and the picture was sent to an VP8 client (linphone on Android). The playback took about 20% CPU usage on a Xeon core -- probably because of resising work. The source file was taken from http://www.quirksmode.org/html5/tests/video.html Question: what other file formats are supported? I tried .ogv and .webm, but I got "Invalid file format" error. thanks

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