- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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DESTINATION_OUT_OF_ORDER
by Jerry Richards 07 Jan '10

07 Jan '10

Hello, I have one FS instance that is working well with a PRI and running FS version 1.0.5pre9. Originally, it was running FS version 1.0.4 and I then upgraded it. Now, I am trying to bring up another FS instance (basically a clone of the first), but the PRI does not work. When I attempt to make an internal-to-PSTN call, I get a "502 Bad Gateway" reply. I verified that both servers are using the latest Sangoma Wanpipe driver, and I copied the conf XML file tree from the old server to the new one. I think the problem has to do with the openzap module, but I'm having difficulty isolating the problem. Could it have built the openzap module incorrectly? Another difference is that I installed 1.0.5pre9 from scratch on the new server (i.e. it never had 1.0.4 running on it). I put the FS log into the pastebin when an outbound call attempt is made: http://pastebin.freeswitch.org/11675 Could someone give me a pointer on what to try next? Thanks and Best Regards, Jerry

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Announcement: FSComm - The FreeSWITCH-based softphone
by Michael Collins 07 Jan '10

07 Jan '10

We are happy to announce a new project: FSComm, a FreeSWITCH-based softphone. Read the story here <http://bit.ly/4HLF71>. We look forward to watching this project grow and become a truly useful tool for VoIP users everywhere. -Michael

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UNIMRCP RECOGNIZE headers
by Neil Burgess 07 Jan '10

07 Jan '10

Hello, Wondering if anyone can help with a unimrcp question. We are using the UNIMRCP client in FreeSwitch to communicate with a commercial UNIMRCP server (Loquendo). We need to control such items as Speech Timeouts that are occurring on the server side, however we have been advised that the only way these can be affected is via headers in the MRCP requests. So, we need to be able to set headers such as Speech-Complete-Timeout, Speech-Incomplete-Timeout, Recognition-Timeout, etc in the MRCP RECOGNIZE request. Is there a mechanism in FreeSwitch which we can use to pass these down before, (or as) we issue the "detect_speech" command. We are happy to use a jscript interface, or whatever if such a capability is available. Many thanks, Neil

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