January 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

robo caller
by Sean Devoy 17 Jan '15

17 Jan '15

Does anyone have a sample RoboCaller script? Perhaps I am using the wrong name and that is why I can't find one. I have a doctor's office that wants to automate the reminder calls about appointments to their patients. I am curious how people handle answering machine detection as well? Thanks, Sean

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Re: [Freeswitch-users] Any CDR tool for Freeswitch that won't remove/overwrite existing config?
by DanB 17 Jan '15

17 Jan '15

Hi Sina, CGRateS is a billing engine and I think it has what you are looking for. It does not touch/influence your FreeSWITCH configuration files. It can serve you as simple CDR server or complete charging solution with some pretty fancy prepaid/postpaid scenarios. Once calculated, CDRs will be accessible to be exported or retrieved via JSON-RPC calls. DanB

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Aalternative to PlayAndGetDigits
by kthofer 16 Jan '15

16 Jan '15

HI There does someone of you have a alternative to the play and get digit function problem is that all dtmf events are released. which kills our scripts. -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer(a)inteli-sim.com

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SIP over Websocket VS SIP over TCP
by Adam Ben-Ayoun 16 Jan '15

16 Jan '15

Hi, We are developing a mobile client that will use the WebRTC media stack and Freeswitch as an MCU (only for conference calls). My question is, since we build a native app, can we use SIP over TCP for signalling? In other words, if Freeswitch receives the WebRTC kind of SDP, will it be able to communicate in the same way as if we were using the SIP over Websocket (the other Freeswitch option)? Any corner cases/considerations with this? Our goal is to avoid implementing SIP over Websocket on the client as much as possible. Thanks, Adam

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Polycom busy lamp with IVR?
by Alex Balashov 16 Jan '15

16 Jan '15

Hi, Sorry if it's a mundane question: I've got two Polycom phones sharing a registration appearance and SLA / busy lamp works fine with them. What I would like is to create a second line appearance whose busy lamp lights up when a call comes into our main IVR. Is this possible? How would I do it? Are there presence hints that can be arbitrarily directed from the dialplan? I wasn't able to discern this from the Polycom presence or general Polycom documentation. Thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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mod_opus bug?
by Thomas Auge 16 Jan '15

16 Jan '15

Hello list, I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at all, and it seems to be enabled. My C skills are limited, but could this be a bug: static opus_codec_settings_t default_codec_settings = { .... /*.usedtx */ 1, .... }; By default, usedtx is enabled, correct? However, in static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) there's this: if (settings->usedtx) { snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); } Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because that's the default setting? Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. Cheers! Thomas P.S.: If my interpretation is correct, it'll also affect useinbandfec.

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User tagging against IP ACL
by Muhammad Naseer Bhatti 16 Jan '15

16 Jan '15

When a call is authenticated via an ACL, is there a way we can flag the user who’s being authenticated? This is needed to billing/identification purpose. Since the user does not exists in the directory and only allowed by an IP ACL, how can we know which user the IP belongs to? — Thanks, Muhammad Naseer Bhatti

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Asterisk's nat=yes feature
by Zoltán Szabó 16 Jan '15

16 Jan '15

Hi, I have a scenario where my softphone is behind nat and freeswitch is not behind nat. With asterisk I could make the audio work with nat=yes in the user's configuration. What is the settings for this on freeswitch? I have tried everything from the following pages: http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA http://wiki.freeswitch.org/wiki/NAT http://wiki.freeswitch.org/wiki/NAT_Traversal http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios (btw these are linking back to each other) Is it not that just easy as net=yes in asterisk? Many thanks, Zoltan

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freeswitch reject calling without registration
by rentmycoder rentmycoder 16 Jan '15

16 Jan '15

Hi all, How is it possible to to reject calling without registration? In asterisk there is no option for this... Thanks, John

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 16 Jan '15

16 Jan '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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