December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Scanning my firewall for open UDP ports?
by Fred-145 17 Dec '09

17 Dec '09

Hello Does someone know of a free service on the web that can check whether the UDP ports on my firewall are open after Freeswitch is up and running? ShieldsUp only scans TCP ports. Thank you. -- View this message in context: http://old.nabble.com/Scanning-my-firewall-for-open-UDP-ports--tp26808383p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Sofia performance
by Yehavi Bourvine 17 Dec '09

17 Dec '09

Hello, In the WIKI page that talks about Freeswitch performance there is a sentence: *libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles* How can I enable more than one profile on the same interface? Won't they colide when using the same IP and port? Thanks! __Yehavi:

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What are the solutions for G729 support ?
by Oscav 17 Dec '09

17 Dec '09

Hi, What are the solutions to support the G729/G723 codec within FreeSwitch ? Thanks -- View this message in context: http://old.nabble.com/What-are-the-solutions-for-G729-support---tp26777181p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Basic Question on the Internal Profile
by Costa Zikalala 16 Dec '09

16 Dec '09

Hi All I understand that to connect to a SIP Provider you have to (amongst other things) define a Gateway on the External Profile. But some gateways may be defined on the Internal Profile. What kind of gateways would these be and what would be their purpose as most gateways are External? Hope my question makes a bit of sense. Thanks CZ

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Where is that codec list coming from?
by Kristian Kielhofner 16 Dec '09

16 Dec '09

Hello everyone, Pastebin here: http://pastebin.freeswitch.org/11525 I've got my pjsip profile configured for G722 only: <param name="codec-prefs" value="G722"/> Yet whenever I send calls using that profile it (mysteriously) indicates support for PCMU in the INVITE. The pastebin includes both the INVITE and "sofia status profile pjsip" to show that only G722 has been enabled. Where is PCMU coming from? -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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build errors :(
by RR 16 Dec '09

16 Dec '09

Hello All, I know you will probably ask me to check out a fresh copy from svn trunk and all, but I assure you I have done that yet I keep getting these errors on make: creating freeswitch cc1: warnings being treated as errors libs/esl/fs_cli.c: In function âcompleteâ: libs/esl/fs_cli.c:440: warning: format â%ldâ expects type âlong intâ, but argument 4 has type âintâ libs/esl/fs_cli.c:440: warning: format â%ldâ expects type âlong intâ, but argument 4 has type âintâ make[2]: *** [fs_cli-fs_cli.o] Error 1 Any ideas? Thanks RR

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Click-to-call and click-to-dial
by John Platts 16 Dec '09

16 Dec '09

How can I perform click-to-call or click-to-dial in FreeSWITCH? Do you have any recommendations on programs capable of click-to-call or click-to-dial from Microsoft Outlook or Microsoft Excel? _________________________________________________________________ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/

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xml_rpc.conf
by Nameer Kazzaz 16 Dec '09

16 Dec '09

Hi all, Can I set xml_rpc server to run on a specific interface I can set the port but not the ip address to bind to. I have a linux server with more then one interface. I don't want to use iptables to block it. Thanks Nameer

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hardware requirement to run voip application
by ovvenkat 16 Dec '09

16 Dec '09

Hi to all, I dont know, whether I can post this question here or not, I dont have any other options . I hope some one will help me here. Here is my question? 1. I have developed my voip application on top of freeSwitch, Its working fine with soft phone. 2. Now, I need to test my application by calling my landline. I am living in India. I did some googling, I got to this hardware, *Linksys SPA3102 *. Is this only hardware enough to run my voip application or need more hardware? I am very new to this platform, and not having knowledge on hardware. Can anyone please suggest me which hardware which is freeswitch friendly? -- Regards Venkatesan OV.

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SIP Error Message 480
by bcxml 16 Dec '09

16 Dec '09

I have Freeswitch and Microsoft Speech Server 2007 on the same box When Speech Server initiates a call, I get a sip error message 480 Here is the internal profile trace... freeswitch@HD-T2253CN> freeswitch@HD-T2253CN> recv 958 bytes from tcp/[209.172.55.154]:1431 at 20:04:05 .445011: ------------------------------------------------------------------------ INVITE sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@219.175.50.104:5060;transport=tcp> CSEQ: 2 INVITE CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTACT: <sip:HD-T2253CN:1415;transport=Tcp;maddr=209.172.55.154;ms-opaque=be 704290e5b4e03b>;automata CONTENT-LENGTH: 340 USER-AGENT: RTCC/3.0.0.0 CONTENT-TYPE: application/sdp ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 209.172.55.154 s=Microsoft Speech Server session c=IN IP4 209.172.55.154 t=0 0 m=audio 35840 RTP/AVP 114 115 4 0 8 97 101 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ send 369 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 TO: <sip:19059183027@219.175.50.104:5060;transport=tcp> CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Content-Length: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_channel.c:613 New Channel sofia/inter nal/12482578002@127.0.0.1:5080 [4ce7c8ed-6970-1c45-acc6-b7ee03c7e506] 2009-12-15 15:04:05.445011 [INFO] mod_dialplan_xml.c:408 Processing 12482578002- >19059183027 in context public 2009-12-15 15:04:05.445011 [NOTICE] switch_core_state_machine.c:187 Hangup sofia /internal/12482578002@127.0.0.1:5080 [CS_EXECUTE] [NORMAL_CLEARING] send 822 bytes to tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692;rport=1431 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;epid=55D003BB53;tag=25bf 436a29 To: <sip:19059183027@219.175.50.104:5060;transport=tcp>;tag=gr4aF6aS8tZ0j CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe CSEQ: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15826M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, RE FER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-descrip tion, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "19059183027" <sip:19059183027@209.172.55.154> ------------------------------------------------------------------------ recv 383 bytes from tcp/[209.172.55.154]:1431 at 20:04:05.445011: ------------------------------------------------------------------------ ACK sip:19059183027@219.175.50.104:5060;transport=tcp SIP/2.0 FROM: <sip:12482578002@127.0.0.1:5080;transport=tcp>;tag=25bf436a29;epid=55D0 03BB53 TO: <sip:19059183027@219.175.50.104:5060;transport=tcp>;tag=gr4aF6aS8tZ0j CSEQ: 2 ACK CALL-ID: 59a030fa-58d9-4811-bc4c-d9a5314ec9fe MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 209.172.55.154:1431;branch=z9hG4bK8a203692 CONTENT-LENGTH: 0 ------------------------------------------------------------------------ 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1154 Session 6 (sofia/ internal/12482578002@127.0.0.1:5080) Ended 2009-12-15 15:04:05.445011 [NOTICE] switch_core_session.c:1156 Close Channel sof ia/internal/12482578002@127.0.0.1:5080 [CS_DESTROY] Can anyone point me in the right direction ? Thanks Brian -- View this message in context: http://old.nabble.com/SIP-Error-Message-480-tp26801000p26801000.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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