April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Disabling REFER on a single gateway
by Tomáš Boros 17 Apr '15

17 Apr '15

Hello freeswitchers, I am trying to disable transfer on a single gateway I am calling bridge using {origination_uuid=${out_uuid},sip_cid_type=pid,disable-transfer=true}sofia/gateway/telekom-ip/${destination_number} and I have also defined param <param name="disable-transfer" value="true"/> for the gateway. When I make calls I can clearly see, that the variable is set "[disable-transfer]=[true]" but in the siptrace I still can see "Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY" and "Allow-Events: talk, hold, conference, refer". REFER should not be there as transfers are disabled. Although the REFER message is there, REFER requests are denied with a 403 forbidden message. Do you think it is a bug? Are there any other purposes of the REFER message than transferring calls? Thank you, -- Tomáš Boros System & Network Administrator VM Telecom, s.r.o. Panenská 8 811 03 Bratislava Slovakia www.vmtele.com <http://www.vmtele.com> | boros(a)vmtele.com <mailto:boros@vmtele.com> Tel: +421 2 20 28 99 79 | Fax: +421 2 20 28 20 30 <http://www.vmtele.com/>

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OpenSIPS Summit May 12 & 13 in Amsterdam!
by Alex Goulis 17 Apr '15

17 Apr '15

The OpenSIPS Project is excited to announce our first European summit on May 12 & 13 2015 at the Westcord Fashion Hotel in Amsterdam! With the release of OpenSIPS 2.1 we'll have so much to cover that we've decided to make this a 2 day event. You can get more information about the summit at: http://www.opensips.org/Community/Summit-2015Amsterdam-Schedule For beginners, get answers to questions like: * why OpenSIPS is the right choice for you network * what are the OpenSIPS capabilities, features and limits * how to extend, expand and enhance your SIP service with OpenSIPS Seasoned professionals can: * Learn about new modules and features * Discuss advanced topics * Discuss the development roadmap for OpenSIPS Speakers include: * Alexandr Dubovikov - Founder of Homer Sipcapture project * Pete Kelly - Sourcevox * Dan Christian Bogos - ITsysCOM * Tijmen de Mes - AG Projects Hotel information: http://www.westcordhotels.com/sea/westcord-fashion-hotel-amsterdam Conference schedule: http://www.opensips.org/Community/Summit-2015Amsterdam-Schedule#toc1 Information on previous summits: http://www.opensips.org/Community/Summits We look forward to seeing you in Amsterdam! -The OpenSIPS Project Team

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 17 Apr '15

17 Apr '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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nohz=off CentOS 6.6 needed?
by Martin Tomczyk 17 Apr '15

17 Apr '15

We are experiencing high cpu load after setting up a CentOS 6.6 (fully updated; kernel 2.6.32) with a stable 1.4.15 freeswitch on a vmware 5.1 esxi server. Is the nohz=off setting still applicable? Thanks a lot for any help. Much appreciated Kind regards Martin

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start_dtmf_generate and bind_digit_action
by Podrigal, Aron 16 Apr '15

16 Apr '15

I have noticed that if I use start_dtmf_generate on the channel, bind_digit_action doesn't bind. Is this by design? Aron

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WEBRTC
by Ali Jibran 16 Apr '15

16 Apr '15

Is there anyway I can use WEBRTC through a SBC? I'm using Kamailio as a load balancer. And I want to make WEBRTC calls come in through Kamailio and not directly to freeSWITCH. I'm using sip.js. Can someone guide me in the right deprecation on this? Thanks :)

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Encoding for received faxes changed in Ruby-ESL library?
by Peter Steinbach 16 Apr '15

16 Apr '15

Hello, we are receiving faxes via Freeswitch, and while receiving an ESL event (hangup, rxfaxresult) we are storing the fax information into a database. We have some problems with some special characaters there (German Umlauts). While sending a Fax via Freeswitch and receiving via Freeswitch we always have the following (correctly) encoded string in the Freeswitch events, while grepping them on the network interface. variable_fax_ident: Mein-B%81ro where %81 represents the germany character "ü". But after evaluating this string in the Ruby(1.8) ESL library, this %81 is changed to mein-B%\3401ro, means "%81" is changed to "%\3401". I have looked into the code, but could not find any encoding change. Anybody has a hint, which encoding is used here, so we may encode it back? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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mod_call_center consultation
by Podrigal, Aron 16 Apr '15

16 Apr '15

Hi, I'm looking for consultation by someone who is actively using mod_call_center and has good expedience with it. Please email me in private. Thanks. -- Aron Podrigal - //Be happy :-)

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pcapsipdump
by Podrigal, Aron 16 Apr '15

16 Apr '15

Hi, I'm running pcapsipdump on fs server and I notice that I'm missing captures for lots of calls. I suspect it might be that the kernel processes the packets on another interface that I'm capturing on (weak_host_model). any other thoughts? Aron

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FS + WinSvr2008 - OK; FS + WinSvr2012 - NOT SO GOOD
by T Fred Farmington 16 Apr '15

16 Apr '15

I know that many of you are using Linux with FS, but a few of us are using Windows OS's. I had a virtual server running FS + Windows Server 2008 up and running OK, but we wanted to put FS into a 'real' physical server. And we are currently using FS 1.5.12b (32bit) As a result of wanting to make the server change in mind I set up a new physical server. And wanting to run the most recent OS, I set it up to run Windows Server 2012 BUT I had LOTS of problems trying to get that running with SIP phones. The most recent problem was reported as: The error messages are: Error Creating SIP UA for profile: external-ipv6....... Error Creating SIP UA for profile: internal-ipv6....... The likely causes for this are.... 1. Another application is already listening on the specific address 2. The IP the profile is attempting to bind to is not local to this system But I had repeatedly confirmed that neither of the 'likely causes' was valid. Out of frustration I changed the server's OS to Windows Server 2008 (consistent with the previous virtual server) That, by itself, resolved the problem. Now without any change to the server's previous IP and no change to the FS Port, the new FS server was working. There MUST be something wrong with that version of FS running under Windows Server 2012. If anyone knows what the problem might be and how to resolve it, I'd appreciate learning what it might be. Thanks, TF

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