March 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to create IVR application in C#?
by Frankie Yiu 02 Mar '11

02 Mar '11

Hi there, I am newbie to FreeSwitch and I have question about creating an IVR application in C#, with a possibly of using VoiceXML. Could someone please points me to how I can get started or any example that I can look at? Thanks a lot!!! Frankie

3 2

Does FS handle three-way calling on the same POTS?
by GillesToo 01 Mar '11

01 Mar '11

Hello My ISP offers basic PBX features and free calls through when plugging a phone in an RJ11 on the ADSL modem. I'd like to hook up a PC to the modem with an FXO module, and use three-way calling so that the server calls me back on my cellphone, and then lets me call a number through the server before switching to a conference call. Poor man's callback system :-) I finally got Asterisk to make the first call, and put it on hold, but it stops there: It appears that Asterisk is unable to create a second channel on the same FXO module when a call has been put on hold ("app_dial.c:1310 dial_exec_full:Unable to create channel of type 'Dahdi' (cause 0 - Unknown)"). I was wondering if someone had successfully used Freeswitch in this scenario? FWIW, here's my extensions.conf: ============== [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(GLOBAL(CID)=${CALLERID(num)}) exten => s,n,Hangup() ;script waits and creates callfile exten => h,1,system(/var/tmp/test10.lua ${CID}&) ;context used by callfile [callback] exten => start,1,NoOp(In callback, CID is ${CID}) ;how to wait until remote phone picked up? exten => start,n,Wait(5) exten => start,n,Answer() exten => start,n,Playback(tt-monkeysintro) exten => start,n,Flash() ;BAD exten => start,n,Dial(Dahdi/1/${GSM}) ;does ring second number, but no actual voice channel exten => start,n,SendDTMF(456789) ============== Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-call… Sent from the freeswitch-users mailing list archive at Nabble.com.

3 4

Thoughts about mod_conference enter-sound, when wait-moderator has been set
by Peter Olsson 01 Mar '11

01 Mar '11

Good afternoon (at least for me)! Stupid question maybe - but here it goes :) When I configure a conference to wait for the moderator before connecting people, I've noticed that "enter-sound" is played synchronously. Is there any special reason for this? I'm thinking about adding a new parameter to allow it to play asynchronously, even when in this mode. My reasons for this is if there are 50 people logged in, and the moderator connects to the conference, there will be lots of time spent just to play the enter-sound for each member. 1 second of sound for each member will cause 50 seconds before it has played them all. It makes sense in the case when the caller's name is read, but for a common enter-sound I think an asynchronous playback would be good enough? /Peter

3 3

Compilation with -march options
by curriegrad2004 01 Mar '11

01 Mar '11

Is compiling FreeSwitch with the -march option in gcc a good idea? Because I just compiled the latest git using the -march option targeted for a Pentium 3 Processor and it did seem to increase the performance slightly in switching calls and transcoding calls. So far I haven't seen any adverse effects of doing this, but I was wondering if anybody else out there is also compiling FS with a march option and saw some performance increase or adverse effects...

2 4

T.38 Issues with passthrough handshaking
by Johannes Jakob 01 Mar '11

01 Mar '11

Hi, I'm having some serious trouble getting outbound t38 passthru to work in the following scenario: SIP Upstream <> FreeSWITCH <> Asterisk <> SIP ATA <> analog fax I'm not sure if it's a plain FS problem, but I have to start somewhere and I found at least one problem (I think) with FS's behaviour when negotiating with the gateway. Inbound T.38 faxes, when the ATA has to do the reinvite, work perfectly fine from one end to the other. Outbound faxes on the other hand, don't work at all. Here are the important config parts: dialplan before bridging to gateway: <action application="export" data="t38_passthru=true"/> <action application="set" data="proxy_media=true"/> <action application="set" data="bypass_media=false"/> directory-entry for the asterisk-registration: <param name="t38-passthru" value="true"/> and <variable name="proxy_media" value="true"/> <variable name="bypass_media" value="false"/> and finally the gateway's sip_profile: <param name="t38-passthru" value="true"/> <param name="inbound-proxy-media" value="true"/> So much for introduction... When trying to send a fax to 017286483798, this is what happens on the SBC: asterisk "client" is sending the plain audio INVITE to SBC: ----------------------------------------------------------------------------------------------------------------------- 9 37.668216 10.16.139.28 10.16.133.66 SIP/SDP 1202 Request: INVITE sip:017286483798@sbc1.mysip.net, with session description AkE`b?FG^^B%INVITE sip:017286483798@sbc1.mysip.net SIP/2.0 Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec Max-Forwards: 70 From: "0692386432" <sip:0692386432@mysip.net>;tag=as235c17b6 To: <sip:017286483798@sbc1.mysip.net> Contact: <sip:0692386432@10.16.139.28:5060> Call-ID: 6de3237b20a96ab010fb9b3f2436d305(a)mysip.net CSeq: 103 INVITE User-Agent: FPBX-2.8.1(1.8.2.4) Proxy-Authorization: Digest username="sipuser", realm="mysip.net", algorithm=MD5, uri="sip:017286483798@sbc1.mysip.net", nonce="aa9339fc-9e53-4017-85ff-fac5367cb733", response="96f4dea7267d436779cd106947cc25b7", qop=auth, cnonce="4e110d27", nc=00000001 Date: Thu, 24 Feb 2011 07:30:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 v=0 o=root 1297612317 1297612318 IN IP4 10.16.139.28 s=Asterisk PBX 1.8.2.4 c=IN IP4 10.16.139.28 t=0 0 m=audio 15000 RTP/AVP 0 8 3 101 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=ptime:20 a=sendrecv ----------------------------------------------------------------------------------------------------------------------- SBC is forwarding the request to the correct gateway: ----------------------------------------------------------------------------------------------------------------------- 11 37.716704 10.16.133.66 10.15.12.215 SIP/SDP 1232 Request: INVITE sip:+4317286483798@10.15.12.215, with session description =|$E%@^BWB{INVITE sip:+4317286483798@10.15.12.215 SIP/2.0 Via: SIP/2.0/UDP 10.16.133.66:5080;rport;branch=z9hG4bK33eQ6Na9D991D Max-Forwards: 69 From: "0692386432" <sip:0692386432@10.16.133.66>;tag=4NH77aQN2v7ZS To: <sip:+4317286483798@10.15.12.215> Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180 CSeq: 8930272 INVITE Contact: <sip:gw+gateway1@10.16.133.66:5080;transport=udp;gw=gateway1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 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: 269 X-FS-Support: update_display Remote-Party-ID: "0692386432" <sip:0692386432@10.16.133.66>;party=calling;screen=yes;privacy=off P-Asserted-Identity: <sip:06923864320@sip.gateway.de> v=0 o=FreeSWITCH 3022782041 3022782042 IN IP4 10.16.133.66 s=FreeSWITCH c=IN IP4 10.16.133.66 t=0 0 m=audio 18446 RTP/AVP 0 8 3 101 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=ptime:20 ----------------------------------------------------------------------------------------------------------------------- Gatway says, everything OK, audio call patched through: ----------------------------------------------------------------------------------------------------------------------- 22 39.760159 10.15.12.215 10.16.133.66 SIP/SDP 750 Status: 200 OK, with session description AkEw W^BSIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.133.66:5080;branch=z9hG4bK33eQ6Na9D991D;rport=5080 Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180 From: "0692386432"<sip:0692386432@10.16.133.66>;tag=4NH77aQN2v7ZS To: <sip:+4317286483798@10.15.12.215>;tag=h1hl6tsu-CC-39 CSeq: 8930272 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Contact: <sip:10.15.12.215:5060;user=phone> Content-Length: 217 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 3014926 3014927 IN IP4 10.15.12.215 s=Sip Call c=IN IP4 10.15.12.215 t=0 0 m=audio 15490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 ----------------------------------------------------------------------------------------------------------------------- SBC is telling the asterisk box, everything is fine ----------------------------------------------------------------------------------------------------------------------- 28 39.770349 10.16.133.66 10.16.139.28 SIP/SDP 1185 Status: 200 OK, with session description =|$Ew(a)1y^B^}aSIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.139.28:5060;branch=z9hG4bK3c5810ec From: "0692386432" <sip:0692386432@mysip.net>;tag=as235c17b6 To: <sip:017286483798@sbc1.mysip.net>;tag=crS25Ur70m0BK Call-ID: 6de3237b20a96ab010fb9b3f2436d305(a)mysip.net CSeq: 103 INVITE Contact: <sip:017286483798@10.16.133.66:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 222 Remote-Party-ID: "017286483798" <sip:017286483798@10.16.133.66>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 3022845683 3022845684 IN IP4 10.16.133.66 s=FreeSWITCH c=IN IP4 10.16.133.66 t=0 0 m=audio 28850 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 ----------------------------------------------------------------------------------------------------------------------- so, after the ACK which is passed along as well, there is the typical audio stuff, ringing... in both directions the right, pre-negotiated udp ports, everything is fine so far: ----------------------------------------------------------------------------------------------------------------------- 45 39.849484 10.15.12.215 10.16.133.66 RTP 216 PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352 User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 (18446) ----------------------------------------------------------------------------------------------------------------------- ----------------------------------------------------------------------------------------------------------------------- 46 39.849541 10.16.133.66 10.16.139.28 RTP 216 PT=ITU-T G.711 PCMU, SSRC=0x727E59A0, Seq=23045, Time=3032460352 User Datagram Protocol, Src Port: 28850 (28850), Dst Port: hydap (15000) ----------------------------------------------------------------------------------------------------------------------- Now, the magic happens, the gateway is sending it's T.38 re-INVITE to establish better fax connectivity... well, let's see what happens: The gateway is suggesting to move the audio stuff from port 15490 to 15492, and instead speak t38/udptl on port 15490, keep that in mind. ----------------------------------------------------------------------------------------------------------------------- 2099 50.121304 10.15.12.215 10.16.133.66 SIP/SDP 1026 Request: INVITE sip:gw+gateway1@10.16.133.66:5080;transport=udp;gw=gateway1, in-dialog, with session description AkEW^BwINVITE sip:gw+gateway1@10.16.133.66:5080;transport=udp;gw=gateway1 SIP/2.0 Via: SIP/2.0/UDP 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180 From: <sip:+4317286483798@10.15.12.215>;tag=h1hl6tsu-CC-39 To: "0692386432"<sip:0692386432@10.16.133.66>;tag=4NH77aQN2v7ZS CSeq: 2 INVITE Max-Forwards: 69 Contact: <sip:10.15.12.215:5060> Content-Length: 527 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 3014926 3014928 IN IP4 10.15.12.215 s=Sip Call c=IN IP4 10.15.12.215 t=0 0 m=image 15490 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 15492 RTP/AVP 8 0 127 103 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:127 PCMU/8000 a=gpmd:127 vbd=yes a=rtpmap:103 PCMA/8000 a=gpmd:103 vbd=yes a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax a=fmtp:101 0-15 ----------------------------------------------------------------------------------------------------------------------- FS is handing this INVITE to asterisk (and tells asterisk, it would accept audio and/or t.38, both on the same port 28850, don't think that's a problem, but it's at least different from HuaweiSoftX3000's behavior): ----------------------------------------------------------------------------------------------------------------------- 2101 50.122833 10.16.133.66 10.16.139.28 SIP/SDP 1398 Request: INVITE sip:0692386432@10.16.139.28:5060, in-dialog, with session description =|$Efw@0^B^R6INVITE sip:0692386432@10.16.139.28:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.133.66;rport;branch=z9hG4bKy5perv1F90acS Max-Forwards: 70 From: <sip:017286483798@sbc1.mysip.net>;tag=crS25Ur70m0BK To: "0692386432" <sip:0692386432@mysip.net>;tag=as235c17b6 Call-ID: 6de3237b20a96ab010fb9b3f2436d305(a)mysip.net CSeq: 8930278 INVITE Contact: <sip:017286483798@10.16.133.66:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 532 X-FS-Support: update_display P-Asserted-Identity: <sip:06923864320@sip.gateway.de> v=0 o=FreeSWITCH 3022845683 3022845685 IN IP4 10.16.133.66 s=FreeSWITCH c=IN IP4 10.16.133.66 t=0 0 m=image 28850 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 28850 RTP/AVP 8 0 127 103 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:127 PCMU/8000 a=rtpmap:103 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=gpmd:127 vbd=yes a=gpmd:103 vbd=yes a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax ----------------------------------------------------------------------------------------------------------------------- The asterisk box says this is fine (after of course successfully talking to the ATA, which is fine with it, too, but want's to have slower speed). The asterisk box is also changing the port it wants to get t38 data on from 15508 to 4676 and finally sets the udp port for audio 0 to disable it. Just plain t.38 in the new SDP description: ----------------------------------------------------------------------------------------------------------------------- 2107 50.165366 10.16.139.28 10.16.133.66 SIP/SDP 920 Status: 200 OK, with session description AkE`b?G^^^BtAbSIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.133.66;branch=z9hG4bKy5perv1F90acS;received=10.16.133.66;rport=5060 From: <sip:017286483798@sbc1.mysip.net>;tag=crS25Ur70m0BK To: "0692386432" <sip:0692386432@mysip.net>;tag=as235c17b6 Call-ID: 6de3237b20a96ab010fb9b3f2436d305(a)mysip.net CSeq: 8930278 INVITE Server: FPBX-2.8.1(1.8.2.4) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:0692386432@10.16.139.28:5060> Content-Type: application/sdp Content-Length: 307 v=0 o=root 1297612317 1297612319 IN IP4 10.16.139.28 s=Asterisk PBX 1.8.2.4 c=IN IP4 10.16.139.28 t=0 0 m=audio 0 RTP/AVP 8 0 127 103 101 m=image 4676 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:397 a=T38FaxUdpEC:t38UDPRedundancy ----------------------------------------------------------------------------------------------------------------------- BUT look at this! What does FreeSWITCH tell the gateway??? It sends 200 OK, but suddenly wants to receive only audio data and disables comfort noise? Same udp port as before, but no sign of t.38 in the SDP description! ----------------------------------------------------------------------------------------------------------------------- 2111 50.170760 10.16.133.66 10.15.12.215 SIP/SDP 902 Status: 200 OK, with session description =|$Ev%(a)^BWbA1SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.12.215:5060;branch=z9hG4bK4abf7e790396e9d932f59632f From: <sip:+4317286483798@10.15.12.215>;tag=h1hl6tsu-CC-39 To: "0692386432" <sip:0692386432@10.16.133.66>;tag=4NH77aQN2v7ZS Call-ID: e5c9ab65-ba8a-122e-21ad-863d7c249180 CSeq: 2 INVITE Contact: <sip:gw+gateway1@10.16.133.66:5080;transport=udp;gw=gateway1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7847289 2011-02-19 23-38-04 +0100 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 168 v=0 o=FreeSWITCH 3022782041 3022782043 IN IP4 10.16.133.66 s=FreeSWITCH c=IN IP4 10.16.133.66 t=0 0 m=audio 18446 RTP/AVP 8 0 127 103 101 m=audio 0 RTP/AVP 19 ----------------------------------------------------------------------------------------------------------------------- and look at this, even though we just told the gateway to only talk audio to it, we send a t38 packet! (it's this lonely one though!) ----------------------------------------------------------------------------------------------------------------------- 2112 50.174592 94.186.133.66 87.234.1.215 T.38 216 UDP: UDPTLPacket Seq=32768 t30ind: <unknown>[UNKNOWN PER: 10.9.3.8.1][Malformed Packet] User Datagram Protocol, Src Port: 18446 (18446), Dst Port: 15490 (15490) ----------------------------------------------------------------------------------------------------------------------- The gateway keeps sending normal audio to us on the specified and unchanged port, BUT from the udp port it originally told us it would only accept t.38 on... ----------------------------------------------------------------------------------------------------------------------- 2260 51.589980 10.15.12.215 10.16.133.66 RTP 216 PT=ITU-T G.711 PCMA, SSRC=0x727E59A0, Seq=23634, Time=3032554272 User Datagram Protocol, Src Port: 15490 (15490), Dst Port: 18446 (18446) ----------------------------------------------------------------------------------------------------------------------- While the gateway is sending us plain audio, we are talking t38 to the asterisk box (which is not responding). ----------------------------------------------------------------------------------------------------------------------- 2261 51.590063 10.16.133.66 10.16.139.28 T.38 216 UDP: UDPTLPacket Seq=32776 data:v8:[UNKNOWN PER: too long integer(per_integer)][Malformed Packet] User Datagram Protocol, Src Port: 28850 (28850), Dst Port: dhct-alerts (4676) ----------------------------------------------------------------------------------------------------------------------- For the record, on the asterisk (version 1.8.2.4) box I defined t38pt_udptl=yes,redundancy directmedia=no for the gateways and the ATA's extension. On both, the sbc and the asterisk box I compiled res_fax_spandsp, mod_spandsp with spandsp-0.0.6pre18. So... I've been spending far tooooo much time debugging this and I'm quite sure I'm just too stupid to find a solution for this. Is there any good pcap anonymizing utitlity, that can substitute application layer stuff as well? Well, *any* help/hint would be appreciated very much ;) Thanks in advance, John

3 9

problem with voice quality
by joy this 01 Mar '11

01 Mar '11

Dear all: The scenario is that, when I use 3CX (PCMA) to call XLite, 3CX will hear loud noise but XLite is fine. And I found that it would only happen when XLite supports PCMA and PCMU simultaneously. If XLite only support PCMA, everything is fine. Maybe it is because of the old version of Freeswitch, I download the latest version yesterday and re-test. The result is that 3CX hear nothing instead of noise. I use Wireshark to survey packages, and the server does not transfer UDP packages to 3CX. I check the log file, it is as below. -----3CX hears nothing----- 2011-02-24 20:17:30.724125 [DEBUG] sofia.c:4683 Channel sofia/internal/1000@thisjoy.x.x:5060 entering state [completed][200] 2011-02-24 20:17:30.724125 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000@thisjoy.x.x:5060 [BREAK] 2011-02-24 20:17:30.724125 [DEBUG] switch_channel.c:2790 (sofia/internal/1000@thisjoy.x.x:5060) Callstate Change EARLY -> ACTIVE 2011-02-24 20:17:30.724125 [NOTICE] switch_ivr_originate.c:3363 Channel [sofia/internal/1000@thisjoy.x.x:5060] has been answered 2011-02-24 20:17:30.724125 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Success: [sofia/internal/sip:1001@114.24.x.x:56922] 2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_originate.c:3408 Originate Resulted in Success: [sofia/internal/sip:1001@114.24.x.x:56922] 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:1001@114.24.x.x:56922 [BREAK] 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/1000@thisjoy.x.x:5060 [BREAK] 2011-02-24 20:17:30.739750 [DEBUG] switch_ivr_bridge.c:1234 (sofia/internal/sip:1001@114.24.x.x:56922) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-02-24 20:17:30.739750 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:1001@114.24.x.x:56922 [BREAK] 2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:1001@114.24.x.x:56922) Running State Change CS_EXCHANGE_MEDIA 2011-02-24 20:17:30.739750 [DEBUG] switch_core_state_machine.c:369 (sofia/internal/sip:1001@114.24.x.x:56922) State EXCHANGE_MEDIA 2011-02-24 20:17:30.739750 [DEBUG] mod_sofia.c:552 SOFIA EXCHANGE_MEDIA 2011-02-24 20:17:30.974125 [DEBUG] switch_rtp.c:2985 Correct ip/port confirmed. 2011-02-24 20:17:30.974125 [DEBUG] sofia.c:4683 Channel sofia/internal/1000@thisjoy.x.x:5060 entering state [ready][200] 2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal sofia/internal/sip:1001@114.24.x.x:56922 [BREAK] 2011-02-24 20:17:31.021000 [DEBUG] switch_core_session.c:771 Send signal sofia/internal/1000@thisjoy.x.x:5060 [BREAK] -----log end----- When XLite only supports PCMA, it would be an additional log: 2011-02-24 20:16:06.661625 [INFO] switch_rtp.c:2968 Auto Changing port from 192.168.1.1:51508 to 114.24.x.x:51508 And I test it again by using a public VoIP server ( sip.lowratevoip.com) there is no noise or silence situation, everything is fine. Does XLite get bugs? Why the public server can deal with the situation? Could anyone help me? Sincerely yours, thisjoy.

6 9

conference dialing and montly billing
by deniro 01 Mar '11

01 Mar '11

Hi All I would like to write some type of billing program that will collect the charges for each account monthly When someone calls into the (freeswitch) conference by dialing toll free number or local number and enters PIN number the program will recognize that and start collecting number of minutes and number of persons dialed in and calculate the amount of dollars. Lets say conference is 10cent/per minute /per person for a tool free number Each time people dial into conference it will calculate total amount by person and by minutes, and generate monthly billing. The PIN numbers may be different for conferences that belong to same account. How would I do such thing? Where do I start from? Is there any sample programs like it somewhere out there Which language would be best to it with I would pay for any professional services Thanks in advance deniro--

3 3

[ANNOUNCEMENT] EVERYONE PLEASE READ
by Anthony Minessale 01 Mar '11

01 Mar '11

Hi, I just want to give a reminder that we have a growing community and we want to put our resources to their best use. Here are some important things to consider. Many of you are already doing these things so if you already are, thank you, please help encourage others to follow your example. 1) We have a bug/issue tracker site using JIRA: http://jira.freeswitch.org This site is for reporting and discussing issues. Please use this tool to keep track of issues you have. Even if it's not a bug we can easily close it tagged "not a bug". Also please help out when you can by reviewing this site to see if you can provide answers to other people. 2) If you see someone complain about the WIKI please remind them that anyone can edit the WIKI and if it's wrong we can clean it up. It takes longer to complain about the wiki than it does to edit it to be correct or add the missing info. That is the idea behind wikis. 3) Please take the time to provide answers to any questions you may know when perusing the list. The more people who help the less work it is to keep everyone informed. 4) Take the time to test the latest development build. Its better to find bugs before we release than after. People tend to anchor themselves to the "official" releases and never try the BETA versions. The sad truth is the tagged release will begin to build a large list of known bugs starting 30 seconds after its tagged. You don't have to run HEAD in production every morning but you should stay up to speed with what's going on with the software and help to produce a stable release that truly is stable. 5) We will be branching 1.0 so it will slow down and taper off from upstream patches from HEAD. Anyone who wants to help maintain that via JIRA and GIT should let us know. Once we branch, most of our attention will still be focused on HEAD and only critical patches or bug fixes will be shared between 1.0 and 1.2 and we will rely on those interested in slower paced development to help keep that code base in order. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

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Global configuration to enable SRTP in the entire FS system
by Goutham BG 01 Mar '11

01 Mar '11

Hi All, We can enable SRTP for a particular channel/call by setting the variable sip_secure_media in the XML dialplan or in the dial string. Is there any global configuration to enable and enforce SRTP which would apply to all the extensions in the dialplan and all the channels that are created. Thanks Goutham B G

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How to run linux shell command from within
by Dmitry Saratsky 01 Mar '11

01 Mar '11

Thanks Christopher, I already have found why this was not working for me (it was quotes wrong usage). However I already have finished with my script with LUA’s (not FS’s) function: os.execute(execute_command); that also runs shell commands, but dose it directly from within LUA, but not through FreeSWITCH’s API. Do you see any problems with that? Thanks, Dmitry.

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