- FreeSWITCH-users - FreeSWITCH Mailing Lists

Retrieve conference member state using cli?
by Lon Baker 28 Oct '09

28 Oct '09

Just like you can call conference list for conferences, is there a way to retrieve the profile state of a conference member using the cli or xml/rpc? Lon

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reload from managed
by srinivasula reddy 28 Oct '09

28 Oct '09

Hi, how can i call reload from managed code? and i called Csharp_switch_xml_open_root, this is working first time, then onwards it is not working. is this the correct function to call reload? -- Srinivasula Reddy K

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switch_xml_open_root
by srinivasula reddy 28 Oct '09

28 Oct '09

** Hi, when i am calling switch_xml_open_root(1,err) . i am getting this warning message. HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap( 00160000, 100E9FD0 ). can any know. please help me. Thanks Srinivasula Reddy K

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how to config FS with two net interface?
by Lei Tang 28 Oct '09

28 Oct '09

Hi all, I run FS on a machine with two net interface, each interface has a ip addr, one of the them connect to public network(has ip addr A), the other connect to a private network(has ip addr B), FS server as a SIP server for public through A, all outbound call will bridge to a softswitch in private network through B. here is my sofia config file and diaplan config: sofia internal.xml .... <param name="rtp-ip" value="A"/> <param name="sip-ip" value="A"/> .... sofia external.xml .... <param name="rtp-ip" value="B"/> <param name="sip-ip" value="B"/> .... dialplan ...... <extension name="OUTBOUND"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="set" data="effective_caller_id_number=xxxxxxx"/>  <action application="bridge" data="sofia/external/${destination_number}@xxxxx"/> </condition> </extension> ..... then call seq is sipAgent --> [internal -->(bridge)-->external] -->softswith FREESWITCH the question is, when sipAgent make a outbound call, FS can't recevie the caller's up audio stream, I traced the SIP packets, found that FS has return addr B in SDP when ack the invite request from sipAgent, the ack packet is =============== SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208 From: "1000" <sip:xxxx@A>;tag=cb4d3c4e To: "65960581" <sip:xxxx@A>;tag=DtvSc0QX01yKN Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI. CSeq: 2 INVITE Contact: <sip:xxxxxx@B:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1256598185 1256598186 IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the private network s=FreeSWITCH c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the private network t=0 0 m=audio 31066 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ================ I think FS should return A in SDP, not the external binding addr (B), does somebody known how to solve this problem? -- Lei.Tang lei.tlfly(a)gmail.com

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FS Training
by Jerry Richards 27 Oct '09

27 Oct '09

Did the voting booth close? I was unable to vote. I'm not sure what link to click and I have had some strange issues with my FS account today. I would be interested in paid training. Do you have plans for offering a training session at your locale? Or would you travel onsite to provide training? Best Regards, Jerry

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Forcing endpoint registration
by Lars Zeb 27 Oct '09

27 Oct '09

Is the following the correct command to force registration of an endpoint below which is not showing up in 'sofia status profile internal' currently, but when it does it looks like the following? Call-ID: mjkg8c3zlqwjzfd.hrt5 User: 1019(a)192.168.10.29 Contact: "user" <sip:1019@192.168.10.103;line=3852> Agent: snom-m3-SIP/02.02 (MAC=0004132A31C7; HW=1) Status: Registered(UDP)(unknown) EXP(2009-10-27 10:46:07) Host: fs IP: 192.168.10.103 Port: 5060 Auth-User: 1019 Auth-Realm: 192.168.10.29 MWI-Account: 1019(a)192.168.10.29 sofia profile internal flush_inbound_reg <mailto:1019@192.168.10.29> 1019(a)192.168.10.29 Thanks Lars

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FreeSWITCH Weekly Conference Call Agenda - Oct 30th
by Michael Collins 27 Oct '09

27 Oct '09

Greetings all! I just wanted to say thank you to those who have been contributing to the documentation and cleanup efforts. Please keep up the good work. FYI, we still have lots of janitorial stuff to do. Just keep checking the latest conf call agenda for updates. This week's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30 We are always looking for people to help out with documentation and other little projects so please feel free to ask me if you would like to assist. Also, we are working on having members of the community give brief presentations during the weekly conference calls. We want people to join the conference and share with the group information about how they're using FreeSWITCH in production, or they can give little tutorials on how to use various parts of FS. We are also working on getting a web-cast enabled so that we can have a little video to go with the audio. If you have something you'd like to discuss with group please let me know. Thanks for being such a great community! -Michael

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Playing Background music as well as a file
by lakshmanan ganapathy 27 Oct '09

27 Oct '09

Hi all, I've done experimenting with the uuid_displace and mux. What mux does is playing a file when the conversation is also happening. But I've a different requirement. I need to play a background music to a UUID, that will get played continuously and also I need to play some other voice message to that uuid. Is it possible in freeswitch? If so please guide me on how to do that!

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Resend: Issues with SIP + TCP?
by Kristian Kielhofner 27 Oct '09

27 Oct '09

I originally sent this last Friday but I've been unable to confirm it ever made it to the list. Hello everyone, I'm having some issues with SIP and TCP. I've used it before with success but I'm seeing some strange behavior... Level 7 debugs with siptrace on both profiles. UDP invite from softphone comes in on port 5062, it's supposed to bridge to 10.70.0.62. When configured to use UDP FS sends an INVITE (nothing currently answers) while TCP doesn't send anything (confirmed with siptrace and packet sniffer). I confirmed this behavior with a gateway configured for TCP and appending ;transport=tcp to a bridge line. This is trunk rev 15211 on an Intel Mac running Snow Leopard. I've also confirmed this behavior on an Intel Linux machine running Ubuntu (not sure of version ATM). TCP: http://pastebin.freeswitch.org/10825 UDP: http://pastebin.freeswitch.org/10826 dialplan (UDP): <extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887(a)10.70.0.62"/> </condition> </extension> dialplan (TCP): <extension name="smhpbx"> <condition field="destination_number" expression="^(7887)$"> <action application="set" data="call_timeout=60"/> <action application="set" data="effective_caller_id_name=Voalte Test"/> <action application="set" data="effective_caller_id_number=19412848354"/> <action application="bridge" data="sofia/avaya/7887(a)10.70.0.62;transport=tcp"/> </condition> </extension> Any thoughts? Thanks! -- 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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How to load user account from databse ?
by Lei Tang 27 Oct '09

27 Oct '09

Hi All: I'm a newbie to FS. I'm using FS as a sbc and have about 20000 user account . Does somebody can tell me how to make FS load use account information from a database such as mssql or mysql? Could you give me a sample configuration file? Thanks a lots.

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