July 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Any one have experience for Asterisk register on freeswitch ?
by Shaw Xu 17 Jul '14

17 Jul '14

I am newbee of freeswitch.( many year experience on asterisk / Elastix / freepbx) I checked freeswitch is great as a soft switch(means works as the VoIP router for many PBXes) , and Asterisk (Elastix / freepbx / PIAF and so on) have more feature as the pbx. So my idea is using freeswitch as the the main hub ( to SIP trunk, which will provide hundreds DID to many PBXes like asterisk / freepbx/elastix.) and freeswitch will collect all minutes to IMS (sip trunk provider) I checked forum, looks the wiki only has documents on FreeSwitch and Asterisk directly connect(dial exten each other) , which my intention is using Elastix / PIAF (on top of freepbx) as sub PBX, and the call routered from FreeSwitch should go through the Elastix inbound Routes rules instead the direct extension as in the wiki https://confluence.freeswitch.org/display/FREESWITCH/Asterisk system environment is CentOS 5.10 with follow the wiki install guide to install freeswitch.( by the way, the MOH not working after install the MOH sound, because the directory is not correct, I copy to the right location manually, and need start the moh for first time.) Any ideas are appreciated.

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e911
by Joli Martinez 17 Jul '14

17 Jul '14

Hello I am looking for an open source e911 service where I can manage multiple locations centrally. Does anyone have any experience with any software that can do this? Thanks,

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SIM bank
by Joli Martinez 17 Jul '14

17 Jul '14

Hello, I am looking for a way to connected a Freeswitch to a small SIM bank for 1-2 SIM's. The goal is to be able to have inbound calls hit the SIM and then Freeswitch route the calls, and then use that SIM for outbound based on my LCR table. Does any one have experience with something like this and if so what brands/models would you recommend? Thanks,

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fs core dsn, agents, tiers and queues
by David Villasmil 16 Jul '14

16 Jul '14

Hello Guys, I configured core-dsn with mysql and it's working fine. I also configured mod_callcenter. When I restarted FS, it created the tables "agents" and "tiers", but did not create "queues"... is this by desing? Thanks!

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Stuck at branch v1.4 commit 9479729
by Victor Chukalovskiy 16 Jul '14

16 Jul '14

Good day, not sure what's wrong here. My FS installation appears stuck at this version: > FreeSWITCH Version 1.4.6+git~20140603T193516Z~9479729340~64bit (git > 9479729 2014-06-03 19:35:16Z 64bit) "make current" and/or "git pull" and recompiling don't help. However, if I look in fisheye there was lots of activity since 2014-06-03 in branch v1.4. Thanks!

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Git.freeswitch.org now points at stash!
by Ken Rice 16 Jul '14

16 Jul '14

Hey FreeSWITCHers, If you have not updated the remotes for you local git checkouts, you will no longer be able to git pull. The old git server has been retired and everything now points at the stash server. Visit https://stash.freeswitch.org/ for the repo that you are looking for and click the clone button to get the correct URL you need to update your remotes to! See you @ ClueCon! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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ClueCon Weekly! Today Bogdan-Andrei Iancu of OpenSIPS!
by Ken Rice 16 Jul '14

16 Jul '14

Join us today at 1PM Eastern, 10A Pacific for ClueCon Weekly! Today¹s special guest is Bogdan-Andrei Iancu, project leader of OpenSIPS! Drop in and find out the latest on OpenSIPS and get a preview of what Bogdan may be talking about at ClueCon 2014! For Access: Visit https://conference.freeswitch.org/ and use your headset for Verto and HD Audio (OPUS/48K/Stereo)!! Call sip:888@conference.freeswitch.org from your sip phones Or +1-919-386-9900 from the PSTN Want to know more about OpenSIPS? Come To ClueCon! Monday @ ClueCon is the OpenSIPS Summit visit http://www.opensips.org/Community/Summit-2014Chicago for more info! And use the discount code ³clucon2014² to get a 25% discount for OpenSIPS Summit! Then stay for ClueCon and get the FreeSWITCH Training on Friday! Visit ClueCon.com for more info! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_httapi caching and sample rate
by Neil Patel 16 Jul '14

16 Jul '14

I have a Lua IVR app that is reading files from a remote server. mod_httapi is caching files to local storage. However, I see this error in the logs frequently: freeswitch.log.2014-07-14-20-00-06.1:2014-07-14 19:39:40.690935 [DEBUG] switch_core_file.c:216 File /usr/local/freeswitch/storage/http_file_cache/7b1120ea3a0ebce2718d6f5588846b9e.wav sample rate 44100 doesn't match requested rate 8000 Does this mean there is a cache miss? I am using session:playAndGetDigits to playback the file, how do I adjust my request to ensure a cache hit? Thanks, Neil

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hangup_time zero
by Aqs Younas 16 Jul '14

16 Jul '14

I am new to freeswitch.I am facing problem in cdr. json_cdr module is giving perfect call start time and duration but hangup_time is 0. Why i am getting hangup_time zero but correct duration of call? Any help would be appreciated. Thanks

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Out of band Signaling (RFC2833) getting lost between legs
by Örn Arnarson 16 Jul '14

16 Jul '14

Hello, I have an inbound call to FS with RFC2833 signaling. FreeSWITCH bridges the call to another profile. A pcap shows a certain DTMF sequence on the inbound leg, and another on the outbound leg. The sequence on the outbound leg is missing some digits from the sequence. Seemingly at random -- I can generate the same sequence twice and a different sequence will be put through. The interval between the digits does not seem to matter either. Firstly, has anyone ever heard of something similar? Secondly, I was informed by someone on #freeswitch on freenode that FS actually regenerates the signaling on the second leg. Can I make FS just blindly forward the RTP events instead of regenerating them to avoid this problem? Regards, Örn

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