June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Writing FS apps via mod event socket + node.js
by Alex Lake 06 Jun '14

06 Jun '14

I'm interested in using node.js to write scripts to control Freeswitch-based telephony apps. This is partly because some of the apps we've been writing seem to be stretching the dialplan/Lua way of doing things - or at least my comprehension of the model - beyond what is commercially viable. What I mean is that writing and extending our apps (particularly when working with multiple call legs with complex inter-related state) is too expensive. It seems from reading around, that externalising the choreography of sessions is the choice of the fs cognoscenti. However, there seems to be very little by the way of documentation and example non-trivial "cookbook" style apps around - or maybe I've just not been looking in the right places. I get the strong impression that there are plenty of people here who've been there, done that and have the t-shirt. Just wondering what the chances are of a bit of sharing (obviously I'm happy to invest time into publishing my findings and creating new examples, etc)

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Can't make call from sip to skype.
by Russell Treleaven 06 Jun '14

06 Jun '14

I can do IM skype to sip and sip to skype I can call from skype to sip When I try to call from sip to skype it immediately goes to voicemailbox 1000(which does not exist). Actually about 1 in 20 times I can complete a call . dialplan <extension name="ext_skype_user"> <condition field="destination_number" expression="^skype/(.*)$" > <action application="log" data="DEBUG !in the skype extension"/> <action application="info"></action> <action application="bridge" data="skypopen/skype101/$1" ></action> </condition> </extension> log http://pastebin.freeswitch.org/22691

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Grandstream GXP2000 living-dead registrations - NAT config problem???
by Alex Lake 06 Jun '14

06 Jun '14

Got a strange problem that has only begun to manifest itself since we started using a host running FreeSWITCH 1.2.24. We were previously using a host running 1.2.13. Both hosts are hosted at Hetzner. I'm not 100% sure that the relevant parts of configuration haven't changed, although I don't think they have. The phones are on a NATted LAN connected via ADSL to the Internet. The problem is that, despite showing up as registered (and continuing to re-register), the phones can get into a state where they don't respond to inbound INVITEs. Other brands of phone (eg. Linksys and Yealink) appear NOT to suffer from this problem. The thing that baffles me is that this appears to be repeatable, but differs between the hosts. What parts of FS config should I be looking at for differences?

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Need some help with SIP Gateway
by deepak sujir 06 Jun '14

06 Jun '14

Hi, I am looking for some help with SIP gateway setup. I have a scenario where the SIP provider needs the *To-Header* and *Request-Line* to have their domain, but the request needs to be routed through a proxy. *Gateway configuration* <include> <gateway name="sip-domain.com"> <param name="realm" value="*sip-domain.com <http://sip-domain.com>*" /> <param name="proxy" value="*sip-proxy.com <http://sip-proxy.com>*"/> <param name="register" value="*false*"/> <param name="from-domain" value="sip-domain.com"/> </gateway> </include> *Dial plan * <action application="bridge" data="sofia/gateway/sip-domain.com/xxxx" /> Now when I send the call out through the gateway, the *To-Domain* and the *Request-Line *looks like *sip:xxxx@sip-proxy.com <sip%3Axxxx(a)sip-proxy.com>*, but i need it to be something like *sip:xxxx@sip-domain.com <sip%3Axxxx(a)sip-domain.com>* Is there a way where that I can set the To-domain? (may be something similar to param *from-domain* ) Thanks Deepak

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Freeswitch Multiple early-media
by Marino 06 Jun '14

06 Jun '14

Hi to all, i'm trying to implement something like this with FS but with bad result :( -)From a client i make parallel forking using FS; -)All receivers replies with 183 w/SDP; -)I would like that all 183 will arrive to client that can stream it's video to all receivers -)The first receiver that pick-up call establish connection With a simple call 1 to 1 everything is ok..but with forking FS get all the 183s messages but it doesn't forward it to the caller. I don't know if RFC allows this kind of scenario, but i would only know if it possible or i need to modify FS sourcecode. Many thanks to all

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5ess decoding
by Kasyanov Dmitry 06 Jun '14

06 Jun '14

Hi, Is there way to implement FS to decode binary 5ess data. I found some api to process 5ess and q931 data, but not quite sure how to use it. Sincerely Kasyanov Dmitry

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Is this SMS scenario possible in FreeSWITCH?
by Mario G 05 Jun '14

05 Jun '14

I checked the wiki and ML for this but could not determine if this is possible: Someone sends an SMS to an ITSP number into FreeSWITCH, then have FreeSWITCH forward the SMS to an outside number. I want to do with SMS what I now do with a bridge to cell phones. My current ITSP does not yet support SMS but would lie to make this work someday if possible. Thanks. Mario G

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FreeSWITCH 1.4.6 Released!
by Ken Rice 05 Jun '14

05 Jun '14

FreeSWITCH 1.4.6 has been released to address a few bugs. This bugs include a regression in re-invite parsing, a few more issues uncovered by our continuing Coverity scans of the code base, addressing some build issues mod_perl, and addressing a WebRTC issue with Chrome where it now requires a longer DTLS Password. This is a routine maintenance update release. Source Tarball is available at http://files.freeswitch.org/freeswitch-1.4.6.tar.gz Additional information on obtaining and installing FreeSWITCH is located at https://confluence.freeswitch.org/display/FREESWITCH/Installation -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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freeswitch how does mod_callcenter deal with 302 redirect
by Andrew Cassidy 05 Jun '14

05 Jun '14

As subject suggests, how does mod_callcenter deal with a 302 redirect from an agent's phone? We have a number of staff out of the office and they have set up call forwarding rather than logging out of the queue, just wondering if it's likely to cause issues. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F <info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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OpenSSL vulnerabilities.
by Michael Jerris 05 Jun '14

05 Jun '14

> A whole list of new vulnerabilities were announced today: > > https://www.openssl.org/news/secadv_20140605.txt > > We recommend everyone using openssl for anything upgrade to 1.0.1h immediately. > >

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