March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch and android
by Brian May 22 Mar '10

22 Mar '10

Hello, Anybody here got freeswitch working with Android? If so, what app did you use? I tried sipdroid which looked very good, but unfortunately it seems it answers incoming phone calls immediately and fakes a ring tone. I filled a bug report, but it was closed as invalid, so far without explanation: http://code.google.com/p/sipdroid/issues/detail?id=383 Thanks -- Brian May <brian(a)microcomaustralia.com.au>

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

For example, I set rtp-start and end port in switch.conf.xml from 16000-32000 but When I register a phone it uses port above 32000. example on a register sip trace : Via: SIP/2.0/UDP 10.10.10.10:52641;rport=52641 Any idea ? ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Sunday, March 21, 2010 11:12 PM Subject: Re: [Freeswitch-users] rtp timeout yet > Hi Metik, > > Thanks for your answer. Just wondering which rtp port FS use really. > on the sip trace I can see often rtp port up to 32768 (like 50434 etc..) > so it should be a problem since I use apache with port 50050:60100 on > the same IP as FS. > > > ----- Original Message ----- > From: "Metik" <freeswitch-users-list(a)metik.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Sent: Sunday, March 21, 2010 3:46 PM > Subject: Re: [Freeswitch-users] rtp timeout yet > > >>I haven't followed this particular thread very closely but are you sure >>there are no port conflicts exhibited in the SDP? This is UDP after all >>and unlike TCP, binding is exclusive and port selection would then need to >>take in account those ports already in use (the endpoint and FS would be >>pretty much flying blind if the endpoint is not a local process). >> >> I would suggest reviewing your endpoint configurations and making sure >> that they are using unique (rtp/media) port ranges. One would think that >> if the endpoint shares the same global IP address as used by FS and it is >> not a (softphone) process running on the same physical box, then it would >> be communicating with FS using its private IP. Does both the signaling >> and media traffic traverse (i.e. routed) through a broadband router that >> sits in front of both FS and the endpoint or is it sent (i.e. forwarded) >> directly? >> >> -metik >> >> Madovsky wrote: >>> last trunk, >>> remove current conf folder and reinstalled the default one after make >>> install. >>> I stopped iptables, connect 1000 to 1001 on port 5060, but >>> after rtp-timeout it closes yet. >>> sip trace after the rtp timeout >>> SIP/2.0 481 Call Does Not Exist >>> I start to be very frustrated.... >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org

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rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated....

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 22 Mar '10

22 Mar '10

Apparently there s a timeout everytime I register a phone on a public IP. for phone behind a firewall (outside FS network) it works well. ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, March 21, 2010 3:09 PM Subject: rtp timeout yet last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated.... -------------------------------------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Downloading Free Switch
by Shaik basha 22 Mar '10

22 Mar '10

Dear All, Good Afternoon every one in this forum. I am new to FS. I tried to understand about FS, but could not understand completely. Hence would any one help me from where I can download FS and what FS can do exactly. Your early response in this regard would be very much appreciated. Thanking you in advance. Regards, shaikh.bashaa

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Open source softphone that can handle g729
by Fraser Redmond 21 Mar '10

21 Mar '10

We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.) Does anyone have any they'd suggest? Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.) Cheers, Fraser

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 21 Mar '10

21 Mar '10

Apparently there s a timeout everytime I register a phone on a public IP. for phone behind a firewall it works well. ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, March 21, 2010 3:09 PM Subject: rtp timeout yet last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated....

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Freeswitch SPA3000 HUP Not Sent
by Ben Langfeld 21 Mar '10

21 Mar '10

Hey, I have a small 7 seat mostly softphone based PBX installation in place using freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside from PSTN HUP detection, outbound calls are fine setup to directly dial the SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely. Unfortunately inbound calls aren't so successful. Calls are sent using the SPA dialplan to an internal freeswitch extension, and the calls connect fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to the ATA. Is there a reason for this? Can anyone give me any idea how I can get freeswitch to instruct the SPA to drop the line? Regards, Ben Langfeld Wave > Email

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Paging
by Sean Holt 21 Mar '10

21 Mar '10

Hello Dev team, Freeswitch rocks!! I have a question about the paging application. I¹ve followed the instructions on the wiki on how to set it up, but when I do a sip trace I¹m noticing a Proxy authentication message when try to connect to the paging extension. Every time I dial the extension it hungs up. I¹ve checked the acl xml file and my subnet is in the access list. Any thoughts on how to correct the authentication issue, if that¹s issue. Thanks Sean

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not registered external softphone's with internal profile
by Vladimir Elizarov 21 Mar '10

21 Mar '10

Hello, I'm trying to register on the fs external softphone (with public ip). Copied profine internal to internal5090 (http://pastebin.freeswitch.org/12506) When you register get 405 Method Not Allowed or 481 Subscription Does Not Exist (http://pastebin.freeswitch.org/12507 and http://pastebin.freeswitch.org/12508) Depending on what softphone use. sofia_contact shows that the username is not registered (error/user_not_registered), although it is in sofia status internal5090 (http://pastebin.freeswitch.org/12509) What causes the appearance of 405, 481 and the absence of registration? -- Best regards, Vladimir Elizarov

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