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Ang.: uuid_media hangs
by Peter Olsson 06 Jun '10

06 Jun '10

Update to latest git, Tony has fixed tjusiga... Peter Olsson ----- Reply message ----- Från: "peely" <freeswitch(a)peely.com> Datum: sön, jun 6, 2010 17:24 Rubrik: [Freeswitch-users] uuid_media hangs Till: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Hi, I'm trying to use the uuid_media command to control the media path after a call has been established. I can't seem to get it to respond wither through the console or through ESL. Whenever I issue a uuid_media {uuid} or uuid_media off {uuid} the console or connection just hangs, even when I have debug output I get nothing. I've tried this with both bridged and originated calls. I have enable-3pcc set to true on the profile. Effectively I am trying to set up an IVR through ESL then connect the caller on to a destination, once connected I want to push the media away. Looking at the SIP coming in from the far end I see invite and update is allowed so in theory this should be supported by the remote party. Could somebody please let me know what the conditions are to be able to use uuid_media? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/uuid-media-hangs-tp5145657p51… Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ 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 !DSPAM:4c0bbda332931523817751!

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Freeswicth nms
by Budi wibowo 05 Jun '10

05 Jun '10

Hi all I'm looking for nms that can do minimum alarm,fault, performance and statistic. Alarm could be sent via sms or email. I need the such solution due I have project to involve more than 10 fs box. Both free and commercial solution are ok Regards Budi

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Db auth
by Budi wibowo 05 Jun '10

05 Jun '10

Dear all Anybody knows how to do authentication based on mysql db? For easier maintenance I want to move the profile to database instead of file config. Any idea or turorial are highly appreciated and welcome. /b Budi

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Issue with Privacy: id and IVR
by David Ponzone 05 Jun '10

05 Jun '10

Hello all, I noticed a weird issue when receiving a call with Privacy: id on an IVR. Privacy: id is present on leg B, although I dont set it manually. On a regular DID (bridged directly to leg B without IVR), I would have to set Privacy: id manually. It's like it's forwarded from leg A automagically. My relevant dialplan is the following: <extension name="from_ipdirections_to_svi"> <condition field="destination_number" expression="^0174031891$"> <action application="ivr" data="svi"/> </condition> </extension> (that's the first extension in the dialplan) The IVR config is: <menu name="svi" greet-long="/usr/local/freeswitch/sounds/vera" timeout="50000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="1"> <entry action="menu-exec-app" digits="1" param="execute_extension set:ringback=/usr/local/freeswitch/sounds/ trying ,bridge:sofia/gateway/mosaica/0174031897 inline"/> <entry action="menu-exec-app" digits="2" param="bridge sofia/ gateway/mosaica/0174031800"/> </menu> I am running latest git. Behaviour is the same on a 1 month old version. Am I missing something ? Thank you David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Can freeswitch work with ODBC for Microsoft Access files?
by Frank Church 05 Jun '10

05 Jun '10

I'd just like to know if any brave souls have tested Freeswitch with Microsoft ODBC for JET 4.0 using Microsoft Access or any of the other file formats it supports, like DBF and Paradox db files. Anyone? -- Frank Church ======================= http://devblog.brahmancreations.com

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Question regarding table channels in core.db
by DJB INTERNATIONAL 05 Jun '10

05 Jun '10

I would like to know whether this is the normal behavior when i ran a query during the call for table channels: select * from channels I noticed that the ip_addr for inbound and outbound shows the same value. Should it show the bleg_ip_addr for outbound direction instead? git-0152706 2010-06-04 19-37-04 -0500 Thank you, Dorn B.

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Re: [Freeswitch-users] Set Privacy:id on Outbound Call
by Michael De Lorenzo 05 Jun '10

05 Jun '10

I tried setting the sip_h_Privacy via: freeswitch.Session("{sip_cid_type=pid,origination_caller_id_name=" .. caller_id_name .. ",origination_caller_id_number=+1" .. cid .. ",sip_h_Privacy=id,ignore_early_media=true}sofia/gateway/mygateway/" .. call_this_number) but in the SIP Capture Privacy is still none: INVITE sip:9735551234@5.6.7.8 <sip%3A9735551234(a)5.6.7.8> SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4:5080;rport;branch=z9hG4bKSaXt6Bva8UD5K Max-Forwards: 70 From: "Anonymous" <sip:9727289377@1.2.3.4 <sip%3A9727289377(a)1.2.3.4> ;transport=udp>;tag=QjjNSt2m3g0rB To: <sip:9735551234@5.6.7.8 <sip%3A9735551234(a)5.6.7.8>> Call-ID: 9b32aa4a-eaa3-122d-9bbb-000bdb94aab9 CSeq: 131718385 INVITE Contact: <sip:gw+verizon1@1.2.3.4:5080;transport=udp;gw=verizon1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22 15:06:05 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 291 X-FS-Support: update_display P-Asserted-Identity: "Anonymous" <sip:+19727289377@1.2.3.4<sip%3A%2B19727289377(a)1.2.3.4> > v=0 o=FreeSWITCH 1275645559 1275645560 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 27884 RTP/AVP 0 8 3 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Am I doing something incorrectly?

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Presence/BLF
by Troy Anderson 05 Jun '10

05 Jun '10

I am trying to get my head around presence/BLF indicators in a mixed endpoint environment. We have a couple Cisco 7940's and a couple Aastra phones (55i and 57i), and some Polycoms (450 and others). Presence works great if I am just using the Aastras, but gets wonky when I mix in other phones. I am using Wireshark to try to understand what's going on, but need some advise on what to look for. Is all this handled with SIP NOTIFY events? Is FS completely in charge of the BLF status on the phones? Thanks for any pointers! -Troy

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Help debugging INVITE with provider
by François Legal 04 Jun '10

04 Jun '10

Hello, I'm having trouble to send/receive calls with one of my providers. Freeswitch says the reply sent by provider has the session information incorrect, then the provider replies the calle is busy (which of course is not). The sip trace follows. Anybody can help ? François send 1413 bytes to udp/[MyProviderIp]:5060 at 15:27:29.696419: ------------------------------------------------------------------------ INVITE sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 642 X-FS-Support: update_display Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1275553793 1275553794 IN IP4 MyFSIpHere s=FreeSWITCH c=IN IP4 MyFSIpHere t=0 0 m=audio 25056 RTP/AVP 124 123 122 121 3 99 9 8 0 101 13 a=rtpmap:124 G726-16/8000 a=rtpmap:123 G726-24/8000 a=rtpmap:122 G726-32/8000 a=rtpmap:121 G726-40/8000 a=rtpmap:3 GSM/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=video 19186 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 H263-2000/90000 a=rtpmap:99 H264/90000 recv 420 bytes from udp/[MyProviderIp]:5060 at 15:27:29.770669: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 1030 bytes from udp/[MyProviderIp]:5060 at 15:27:29.774935: ------------------------------------------------------------------------ INVITE sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 453 v=0 o=FrancoisLegal 1275578849 1275578849 IN IP4 80.239.235.106 s=SIP Call c=IN IP4 80.239.235.106 t=0 0 m=audio 11180 RTP/AVP 2 3 9 8 0 101 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 m=video 11598 RTP/AVP 31 34 115 121 99 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:121 a=rtpmap:99 a=ptime:20 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:29.775380: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 552 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866610: ------------------------------------------------------------------------ ACK sip:gw+12voip.com@MyFSIpHere:5060;transport=udp;gw=12voip.com SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK3fecf431e27c437190b2844df71b49b0 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:0000000000@MyProviderIp:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ recv 423 bytes from udp/[MyProviderIp]:5060 at 15:27:29.866859: ------------------------------------------------------------------------ SIP/2.0 486 Busy here Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK freeswitch@tls-srv-01> From: "" ;tag=2SU9FHayaSXcr To: Contact: sip:MyProviderIp:5060 Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 320 bytes to udp/[MyProviderIp]:5060 at 15:27:29.867043: ------------------------------------------------------------------------ ACK sip:SomeAccount@provider.com SIP/2.0 Via: SIP/2.0/UDP MyFSIpHere;rport;branch=z9hG4bK2Ktev9DUUB7tK Max-Forwards: 70 From: "" ;tag=2SU9FHayaSXcr To: Call-ID: 5d107eb4-e9c7-122d-58a3-001517c07e1e CSeq: 131671088 ACK Content-Length: 0 ------------------------------------------------------------------------ send 605 bytes to udp/[MyProviderIp]:5060 at 15:27:30.276655: ------------------------------------------------------------------------ SIP/2.0 400 Bad Session Description Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------ recv 498 bytes from udp/[MyProviderIp]:5060 at 15:27:30.341335: ------------------------------------------------------------------------ ACK sip:77.72.169.134:5060 SIP/2.0 Via: SIP/2.0/UDP MyProviderIp:5060;branch=z9hG4bK6d186dd63c024b6190e5e139d10c5c48 From: "0000000000" ;tag=4e0113ac4c050c86c51a2 To: ;tag=32m2HcU171KZK Contact: sip:77.72.169.134:5060 Call-ID: 491618f252ba4d08b48c48ce92b0324c@MyProviderIp CSeq: 1 ACK Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 ------------------------------------------------------------------------

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check status of a number
by Madovsky 04 Jun '10

04 Jun '10

Hi, I'd like to use fs_cli to check if the callee number exists and get the status result. Is anyone already did that ? Thanks Franck

3 10