January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Skypiax and Socket API
by Max Bridgewater 08 Jan '10

08 Jan '10

Hey, I see that it's possible to send chat messages to Skype users using sk on top of socket api and skypiax. But how do I received chat messages via the socket api? Are there events generated by Skypiax? If so, do we have a list somewhere? Thanks, Max.

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Skypiax on CentOS
by Max Bridgewater 08 Jan '10

08 Jan '10

Hi, Has anybody installed Skypiax on CentOS 5 lately? The documentation is based on Skype 2.0.0 (skype-2.0.0.72-centos.i586.rpm) which apparently is not available online anymore. I am hoping that you guys can help me either 1) get Skypiax to run with the latest Skype or 2) share this old version so i can follow the existing documentation. My preference would be the second option though ;) Thanks, Max.

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hw:dummy not visible
by Max Bridgewater 08 Jan '10

08 Jan '10

Hi, I got a few more Skypiax questions. Please bear with me. 1) Skype should use hw:dummy as audio device. But where do I set this on Skype? In Options>Sound Devices, the only devices I see are; "Default device" and "hdmi". My guess was that alsa-utils or some other ALSA related lib would install this. But it seems this is not happening. Am i missing something? 2) To create the configuration directory, I connect to the FreeNX XServer running on my remote machine. I use NX to connect to the remote Xserver. I can start all sort of GUI applications this way. But when I try to run skypiax_auth, I get the following error message: "Cannot open X Display ':0.0', exiting". Any idea? Thanks again, Max.

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How to suspend certain users from the system using flush_inbound_reg
by katarina djakovic 08 Jan '10

08 Jan '10

Hi, we are trying to figure out how to suspend certain subscribers from our system and we have some problems with removing thier registrations. The UAs are ATAs. This is what we do: 1) We remove the subscriber extension from the conf\directory .xml files 2) We do reloadxml 3) We flush user's registration with flush_inbound_reg but, the users are still able to make calls as if they were still registered. To make it clearer, their registrations are removed from the registration list (checked with sofia status), but they system still accepts the calls from them. From this, it seems that if ATA is never rebooted - we are not able to ban these users from the system. Only after the ATA is rebooted user is not able to make calls any more, as the ATA can not register any more - since they users are removed from the directory. But before we reboot ATA everything works as nothing had been done. Does anyone have an idea what are we doing wrong? We expect that after the registration is removed from the FS the UA should not be able to make a call but this is not what happnes. Can anybody help please? Thanks, Katarina _________________________________________________________________ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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Re: [Freeswitch-users] Connecting GSM Cards to FreeSWITCH
by Steven Brown 08 Jan '10

08 Jan '10

Hi, I have used Portech gateways with good results, you can get cards and external gateways with various numbers of GSM channels, see http://www.portech.com.tw/p3-product1_1.asp?Pid=13 Regards Steve

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IP PBX and NAT firewalls
by Fred-145 08 Jan '10

08 Jan '10

Hello, I read a couple of thorough articles on SIP, and I'd like to make sure I got things right when it comes to using SIP with NAT routers. I know that, ideally, the IP PBX should be located in the DMZ to void NAT-related issues in SIP, but SOHO routers don't necessarily support this, so I'll assume that the SIP caller "Alice" and the IP PBX (eg. Freeswitch or Asterisk) server are located in a non-routable, private LAN, while the remote callee "Bob" is located on the Internet (either behind their own NAT router, or connected with a public, routable address). The SIP phone of Alice and Bob are both logged on to the Freeswitch server: http://img46.imageshack.us/img46/5120/sipnatrouters.jpg 1. When Alice wants to call Bob, her SIP phone sends an SIP packet to the Freeswitch server with her private IP address and a UDP port that it opened to let incoming RTP packets from Bob 2. Freeswitch rings Bob's phone through the UDP port is used to register with Freeswitch (usually, UDP5060). Bob's phone replies to Freeswitch with his public IP address and the RTP port it chose to receive voice packets from Alice 3. Once Bob picks up the phone, RTP voice packets flow directly between Alice and Bob, while Freeswitch remains in the loop to handle call signaling such as closing the connection when someone hangs up the call. Provided this is how things work... there are three issues when one or all SIP end-points are located in a (different) private LAN: 1. End-points use their private IP and a private UDP port for RTP. A server has to translate this into a routable IP address, and... 2. it must negotiate with the NAT firewall to make sure this RTP port is available, and if not, open some other port, and... 3. the server must rewrite the SDP packet to use this public port I have a couple of questions: 1. Can Freeswitch/Asterisk handle this rewriting/negotiation? 2. Provided the NAT firewall doesn't support UPnP/NAT-PMP, does it mean I must a) enable STUN in Freeswitch, b) set SIP end-points so that they use a fixed port for RTP, and c) configure the NAT firewall to map this UDP port to point to the SIP end-point? 3. Should SIP end-points be configured to use STUN/NAT, or should I let the server handle the IP/port rewriting itself? Thank you for any help. -- View this message in context: http://old.nabble.com/IP-PBX-and-NAT-firewalls-tp27075600p27075600.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Connecting GSM Cards to FreeSWITCH
by Matthew Fong 08 Jan '10

08 Jan '10

Does anyone have any hardware recommendations for setting up 4+ GSM cell phone lines to work with FreeSWITCH?

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Re cording call into existing file
by shehzad p 08 Jan '10

08 Jan '10

Hi, while recording a file using session_record, can i continue the existing recorded file? So that the existing record will remain as it is and new recording will be added into that file? Thanks msp -- View this message in context: http://old.nabble.com/Recording-call-into-existing-file-tp26975973p26975973… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Segmentation fault (core dumped) on "shutdown" if mod_skypiax loaded
by Scott Torr 08 Jan '10

08 Jan '10

Hi, I'm getting a Segmentation fault on "shutdown" if mod_skypiax is loaded on the current build. The audio on skypiax calls is also extremely choppy. This only occurred after a "make current" from 15787. Has anyone else experienced this problem? VMware Server Version 2.0.1 ubuntu-8.04.3-server-amd64.iso (update/upgrade) FreeSWITCH Version 1.0.trunk (16172) skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb mod_skypiax http://jira.freeswitch.org/browse/MODSKYPIAX-68 Regards, Scott

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SPA3102 Help
by Joseph L. Casale 08 Jan '10

08 Jan '10

Reading the wiki, I have Line 1 (the fxs port) configured as ext 1001 and PSTN User (the fxo port) configured as ext 1000. I am a bit unsure of the dial plan part of the wiki's config? Under "VoIP-To-PSTN Gateway Setup" I set "Line 1 VoIP Caller DP:" to a dial plan with (xx.) which dials whatever is sent to it? Under "PSTN-To-VoIP Gateway Setup" I set "PSTN Caller Default DP:" to a dial plan with (<:ABCD>S0) where ABCD could be a group dial for example like 2000 as in the default config? Do I understand this correctly? I would try this out, but I don't have a pstn here and want to pass this off configured for delivery tomorrow and be prepared for any adjustments via ssh if need be... Thanks! jlc

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