February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Polycom G722 negotiation
by Spencer Thomason 03 Mar '11

03 Mar '11

Hello all, I have several Polycom IP-650s. On the external profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT PCMU,G729 On the internal profile I have CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM The Polycoms have a similar codec priority set: v=0 o=- 1167611879 1167611879 IN IP4 10.59.1.243 s=Polycom IP Phone c=IN IP4 10.59.1.243 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 The issue I have is that if I leave the internal profile to generous and a user places a call from the PSTN on hold which is PCMU, when the user tries to pick up the call, the Polycoms' preference then changes the call to G722 and there is no audio. If I set the profile to greedy, since the call is already PCMU Freeswitch keeps it at PCMU and audio is fine. I realize this is the intended behavior for greedy vs generous but my question is why is there no audio when switching the codecs? I didn't want to clutter this up any more but I'm happy to provide traces however the only real difference is the codec order. Thanks, Spencer -- Spencer Thomason 5Nine Solutions LLC e. sales(a)5ninesolutions.com p. +1.888.271.7959 f. +1.310.510.6980

4 11

mod_dingaling and google voice
by envelopes envelopes 02 Mar '11

02 Mar '11

Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered? thanks!

5 11

HANGUP_CAUSE after bridge in outbound ESL
by Dmitry Sytchev 02 Mar '11

02 Mar '11

Hi All What is correct method to get bridge status after execute("bridge") in ESL? I have inbound call that gets bridged to SIP endpoint. I need to know whether it was ORIGINATOR_CANCEL or BUSY or something else, but if I use sync, seems I can't determine ORIGINATOR_CANCEL status, because it is set on a-leg which is destroyed first after hangup. Is there a way to get bridge status without messing with events and event source filtering by channel uuid or event type? -- Best regards, Dmitry Sytchev, IT Engineer

3 4

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

disabling ptime warning message
by Malay Thakershi 01 Mar '11

01 Mar '11

Hello, I updated to the latest FS version last week. I started getting the following warning when speech / sound is played on the call. "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC payload 7 added to sdp wanting ptime 90 but it's already 20 (G7221:115:20), disabling ptime." I read sections on codecs and negotiations. Following are the settings from vars.xml (I have not changed them): <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> Also, there is no codec related setting in sip_profiles files and sofia.conf.xml file. I am playing audio files using Cepstral TTS during the call. Can someone please help me understand these settings? And if they are appropriate? Thank you. Malay

3 16

t38_gateway self - no re-Invite?
by Clemens Ebentheuer 01 Mar '11

01 Mar '11

Hi to all from austria, I´m very new to freeswitch and I´m trying to get t38_gateway self working - but no success. for testing I use zoiper or spa2102 as t38 "device" - i have troubles in outbound direction: t38 device(internal) -> FS:t38_gateway self -> sip-provider (external - no t38 support) freeswitch sends no re-invite, I guess and Zoiper BYEs with Reason: SIP;description="Stale re-Invite" also see: http://pastebin.freeswitch.org/15439 inbound (t38_gateway peer) works fine - also when I do t38 device -> FS:rxfax my simple dialplan for the outbound call: <extension name="scpa" > <condition field="destination_number" expression="^1(.*)$"> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="execute_on_answer=t38_gateway self"/> <action application="bridge" data="sofia/gateway/scpa567/$1"/> </condition> </extension> Maybe I have to mention that I´m using Fusionpbx?! Thanks for any advice! I allready tried different examples I found, but no luck... Clemens

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FreeSwitch Installation Error
by George Lee 28 Feb '11

28 Feb '11

Hi, I changed the installation directory to /opt/freeswitch > ./configure --prefix=/opt/freeswitch Then I (as root) ran make; make install and it gave me this error: /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la' libtool: install: error: cannot install `ftmod_zt.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la' libtool: install: error: cannot install `ftmod_skel.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la' libtool: install: error: cannot install `ftmod_analog.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la' libtool: install: error: cannot install `ftmod_analog_em.la' to a directory not ending in /usr/local/freeswitch/mod What did I setup wrong? Thanks, George

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