I don't know as there's a way in sipXecs. I just do it in my gateway device and phones by preferring G.711uLaw and G.729 only (leave G.711aLaw right out of the picture).
Since I started with sipXecs a few years ago I've always set my gateways to use G.711uLaw and had better results. It isn't supposed to matter but I found it works better in most cases. Mike On Tue, 2009-04-21 at 07:37 +0200, Weigel, Stefan wrote: > Hi Mike, > > > > do you know how to change the primary codec the ACD server is using ? > > > > > > Best regards > > > > Stefan > > > > Von: Picher, Michael [mailto:[email protected]] > Gesendet: Montag, 20. April 2009 11:54 > An: Weigel, Stefan; [email protected] > Betreff: RE: [sipx-users] [SOLVED] white-noise problems with ACD calls > > > > > I’m not sure why there seems to be a problem with G.711a and the ACD > service. I just knew I had a customer start having trouble and the > only thing I had changed were some fax things that I had been working > with Patton on. I went back and noticed that G.711a was the primary > codec… On a hunch changed it to G.711u. > > > > In 4.0 Freeswitch is not handling the ACD or the Voicemail (maybe at > some point). Only the conference bridge. > > > > Mike > > > > From: Weigel, Stefan [mailto:[email protected]] > Sent: Monday, April 20, 2009 5:24 AM > To: Picher, Michael; [email protected] > Subject: AW: [sipx-users] [SOLVED] white-noise problems with ACD calls > > > > > Hi Mike, > > > > > > I checked again all the values. Patton / Thelephones / SipXecs is now > primary using G.711uLaw codec. Now the noise is gone. I'm very > impressed, in the logs I see the capabilities for using PCMA, PCMU, > telephone-event, so why is it only working with PCMU ? And the final > and importent question: how can I change the ACD server config to use > other codecs or to prefer other codecs ??!? I thought with 4.0 the > audio stuff is handled by freeswitch, but changing the config doesn't > impress the ACD server... > > > > > > Thanks Mike! > > > > > > > > > Best regards > > > > > > Stefan > > > > > > > ______________________________________________________________________ > Von: Picher, Michael [[email protected]] > Gesendet: Freitag, 17. April 2009 17:51 > An: Weigel, Stefan; [email protected] > Betreff: RE: [sipx-users] white-noise problems with ACD calls > > > Oddly enough... > > > > I just had a customer call with the same issue. I had done some > changes to my gateway that Patton requested earlier in the week. > > > > The result was that my default codec had changed to G.711aLaw. > > > > I changed the gateway back to prefer G.711uLaw and the static was > gong. > > > > Give it a whirl and report back please. > > > > Thanks, > > Mike > > > > > ______________________________________________________________________ > From: Weigel, Stefan [mailto:[email protected]] > Sent: Friday, April 17, 2009 11:25 AM > To: '[email protected]' > Cc: Picher, Michael > Subject: AW: [sipx-users] white-noise problems with ACD calls > > Hi Michael, > > > > two different environments: > > > > - productive: HP ProLiant DL360 G5, 4 GB RAM, SipXecs > 3.10.2-013143, dual Intel(R) Xeon(R) CPU E5345 @ 2.33GHz > > - testing: VMWare guest (2.0) on a HP ProLiant DL360 G5 (20GB > RAM total), 2 GB RAM assigned fix (no sharing), SipXecs 4.1.0-015201, > Intel(R) Xeon(R) CPU E5345 @ 2.33GHz > > > > I read a mailing list entry regarding memory leak, so we triggered a > daily cron job to reboot the machine each night, but without success. > > > > > > Best regards > > > > Stefan > > > > > > Von: Picher, Michael [mailto:[email protected]] > Gesendet: Freitag, 17. April 2009 15:24 > An: Weigel, Stefan; [email protected] > Betreff: RE: [sipx-users] white-noise problems with ACD calls > > > > > What kind of server do you have it on? Virtual? > > > > Processor speed / RAM? > > > > > ______________________________________________________________________ > From:[email protected] > [mailto:[email protected]] On Behalf Of Weigel, > Stefan > Sent: Friday, April 17, 2009 8:56 AM > To: '[email protected]' > Subject: [sipx-users] white-noise problems with ACD calls > > Hi, > > > > we have a strange problem with white-noise with calls going through > the ACD. > > We tested various setups and SipXecs versions (stable to source > repository), always with the same result. > > > > The setup: > > > > VoIP Gateway (sip.qsc.de) ßà Internet ß à Patton SN4638 ß à SipXecs > (with ACD) ß à Polycom Soundstation IP 560 phones / Snom 300 phone > (only for testing) > > > > I configured the phones to use various codecs, in the beginning the > G729/G722, later PCMU,PCMA. If no hotline is called, the sound is > clear, no white-noise, absolutely brilliant. As soon as I call a > hotline I hear a white-noise while ringing. > > When I pick up the phone, the white-noise disappears after ~ 3 > seconds. If I transfer the call from the Polycom Soundstation to > another phone, I hear the white-noise again for the whole call. With > the Snom I haven’t that problem. > > > > I did a SIP trace, changed to DEBUG log level. I assume that I may > have a codec problem. When calling the hotline, the ACD always answers > with the following settings (sipxacd.log): > > > > sipxacd:"sipxConfigSetAudioCodecPreferences number of Codec = 3 for > hInst=0x91ae058" > > sipxacd:"sipxConfigSetAudioCodecPreferences: PCMU PCMA > audio/telephone-event" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU > PCMA audio/telephone-event with NumReject 0" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = > PCMU PCMA audio/telephone-event with NumReject 0" > > sipxacd:"sipxConfigSetVideoCodecPreferences hInst=0x91ae058 > bandWidth=3" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU > PCMA audio/telephone-event with NumReject 0" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = > PCMU PCMA audio/telephone-event with NumReject 0" > > sipxacd:"sipxConfigSetVideoCodecPreferences:" > > sipxacd:"sipxConfigSetVideoCodecPreferences: Setting 3 failed, falling > back to preference 0" > > sipxacd:"sipxConfigSetVideoCodecPreferences hInst=0x91ae058 > bandWidth=0" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU > PCMA audio/telephone-event with NumReject 0" > > sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = > PCMU PCMA audio/telephone-event with NumReject 0" > > > > > > I changed the settings for codecs in the mediaserver-config, > sipxpark-config, freeswitch.xml + sipX_profile.xml. But still I get > always the log entries from above. > > > > Here’a SIP trace from the SNOM 300 phone calling a hotline: > > > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:41:836 (1074 > bytes): > > > > INVITE sip:[email protected];user=phone SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone> > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 1 INVITE > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > P-Key-Flags: keys="3" > > User-Agent: snom300/7.3.14 > > Accept: application/sdp > > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO > > Allow-Events: talk, hold, refer, call-info > > Supported: timer, 100rel, replaces, from-change > > Session-Expires: 3600;refresher=uas > > Min-SE: 90 > > c: application/sdp > > l: 372 > > > > v=0 > > o=root 105497526 105497526 IN IP4 192.168.150.158 > > s=call > > c=IN IP4 192.168.150.158 > > t=0 0 > > m=audio 60184 RTP/AVP 18 9 8 0 3 99 4 101 > > a=rtpmap:18 g729/8000 > > a=rtpmap:9 g722/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:3 gsm/8000 > > a=rtpmap:99 g726-32/8000 > > a=rtpmap:4 g723/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > a=sendrecv > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:41:898 (288 > bytes): > > > > SIP/2.0 100 Trying > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone> > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 1 INVITE > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 > > Content-Length: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:011 (520 > bytes): > > > > SIP/2.0 407 Proxy Authentication Required > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone>;tag=8350a339 > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 1 INVITE > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 > > Proxy-Authenticate: Digest realm="sipxecs.domain.test", > nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31" > > Server: sipXecs/4.1.0 sipXecs/sipXproxy (Linux) > > Date: Fri, 17 Apr 2009 12:46:41 GMT > > Content-Length: 0 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:017 (379 bytes): > > > > ACK sip:[email protected];user=phone SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone>;tag=8350a339 > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 1 ACK > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > l: 0 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:032 (1306 > bytes): > > > > INVITE sip:[email protected];user=phone SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone> > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 2 INVITE > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > P-Key-Flags: keys="3" > > User-Agent: snom300/7.3.14 > > Accept: application/sdp > > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO > > Allow-Events: talk, hold, refer, call-info > > Supported: timer, 100rel, replaces, from-change > > Session-Expires: 3600;refresher=uas > > Min-SE: 90 > > Proxy-Authorization: Digest > username="974",realm="sipxecs.domain.test",nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31",uri="sip:[email protected];user=phone",response="78c0b7444abc7c39436512a6d259e3d6",algorithm=MD5 > > c: application/sdp > > l: 372 > > > > v=0 > > o=root 105497526 105497526 IN IP4 192.168.150.158 > > s=call > > c=IN IP4 192.168.150.158 > > t=0 0 > > m=audio 60184 RTP/AVP 18 9 8 0 3 99 4 101 > > a=rtpmap:18 g729/8000 > > a=rtpmap:9 g722/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:3 gsm/8000 > > a=rtpmap:99 g726-32/8000 > > a=rtpmap:4 g723/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=ptime:20 > > a=sendrecv > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:040 (520 > bytes): > > > > SIP/2.0 407 Proxy Authentication Required > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone>;tag=8350a339 > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 1 INVITE > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 > > Proxy-Authenticate: Digest realm="sipxecs.domain.test", > nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31" > > Server: sipXecs/4.1.0 sipXecs/sipXproxy (Linux) > > Date: Fri, 17 Apr 2009 12:46:41 GMT > > Content-Length: 0 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:041 (379 bytes): > > > > ACK sip:[email protected];user=phone SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone>;tag=8350a339 > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 1 ACK > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > l: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:102 (288 > bytes): > > > > SIP/2.0 100 Trying > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone> > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 2 INVITE > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 > > Content-Length: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:203 (720 > bytes): > > > > SIP/2.0 180 Ringing > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone>;tag=624039693 > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 2 INVITE > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 > > Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.% > 2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> > > Contact: > <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> > > Date: Fri, 17 Apr 2009 12:46:42 GMT > > User-Agent: sipXacd (4.1.0-015201) (Linux) > > Accept-Language: en > > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, > SUBSCRIBE, NOTIFY > > Supported: replaces > > Content-Length: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:297 (985 > bytes): > > > > SIP/2.0 200 OK > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone>;tag=624039693 > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 2 INVITE > > Contact: > <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> > > Content-Type: application/sdp > > Content-Length: 214 > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 > > Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.% > 2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> > > Session-Expires: 3600 > > Date: Fri, 17 Apr 2009 12:46:42 GMT > > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, > SUBSCRIBE, NOTIFY > > User-Agent: sipXacd (4.1.0-015201) (Linux) > > Accept-Language: en > > Supported: replaces > > > > v=0 > > o=sipXecs 5 5 IN IP4 192.168.150.19 > > s=phone-call > > c=IN IP4 192.168.150.19 > > t=0 0 > > m=audio 10000 RTP/AVP 8 0 101 > > a=sendrecv > > a=rtpmap:8 pcma/8000/1 > > a=rtpmap:0 pcmu/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:307 (528 bytes): > > > > ACK > sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d > SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-4zetnkpayupe;rport > > Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom% > 7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone>;tag=624039693 > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 2 ACK > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > l: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:425 (985 > bytes): > > > > SIP/2.0 200 OK > > From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > To: <sip:[email protected];user=phone>;tag=624039693 > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 2 INVITE > > Contact: > <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> > > Content-Type: application/sdp > > Content-Length: 214 > > Via: SIP/2.0/UDP > 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 > > Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.% > 2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> > > Session-Expires: 3600 > > Date: Fri, 17 Apr 2009 12:46:42 GMT > > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, > SUBSCRIBE, NOTIFY > > User-Agent: sipXacd (4.1.0-015201) (Linux) > > Accept-Language: en > > Supported: replaces > > > > v=0 > > o=sipXecs 5 5 IN IP4 192.168.150.19 > > s=phone-call > > c=IN IP4 192.168.150.19 > > t=0 0 > > m=audio 10000 RTP/AVP 8 0 101 > > a=sendrecv > > a=rtpmap:8 pcma/8000/1 > > a=rtpmap:0 pcmu/8000/1 > > a=rtpmap:101 telephone-event/8000/1 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:427 (528 bytes): > > > > ACK > sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d > SIP/2.0 > > v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-4zetnkpayupe;rport > > Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom% > 7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> > > f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > t: <sip:[email protected];user=phone>;tag=624039693 > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 2 ACK > > Max-Forwards: 70 > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > l: 0 > > > ______________________________________________________________________ > Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:48:834 (692 > bytes): > > > > BYE sip:[email protected]:2052;line=t7xm4yiz;x-sipX-nonat SIP/2.0 > > From: <sip:[email protected];user=phone>;tag=624039693 > > To: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > Call-Id: 3c433d50d289-bfx9jv1z1nk1 > > Cseq: 1 BYE > > Contact: > <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d;x-sipX-nonat> > > Date: Fri, 17 Apr 2009 12:46:48 GMT > > Max-Forwards: 19 > > User-Agent: sipXacd (4.1.0-015201) (Linux) > > Accept-Language: en > > Supported: replaces > > Via: SIP/2.0/UDP > 192.168.150.19;branch=z9hG4bK-sipXecs-00836321a900fb0fc453a9b44c18ad50cc4b > > Via: SIP/2.0/UDP > 192.168.150.19:5150;branch=z9hG4bK-sipXecs-00be1158fe64df32a8248079ea542964915e > > Content-Length: 0 > > > ______________________________________________________________________ > Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:48:880 (607 bytes): > > > > SIP/2.0 200 OK > > v: SIP/2.0/UDP > 192.168.150.19;branch=z9hG4bK-sipXecs-00836321a900fb0fc453a9b44c18ad50cc4b > > v: SIP/2.0/UDP > 192.168.150.19:5150;branch=z9hG4bK-sipXecs-00be1158fe64df32a8248079ea542964915e > > f: <sip:[email protected];user=phone>;tag=624039693 > > t: "TestSnom1" <sip:[email protected]>;tag=bj35colkje > > i: 3c433d50d289-bfx9jv1z1nk1 > > CSeq: 1 BYE > > m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 > > User-Agent: snom300/7.3.14 > > RTP-RxStat: > Total_Rx_Pkts=320,Rx_Pkts=320,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 > > RTP-TxStat: Total_Tx_Pkts=330,Tx_Pkts=330,Remote_Tx_Pkts=319 > > l: 0 > > > > > ______________________________________________________________________ > > > > > > > Am I missing something, is it a coding problem or why do I always here > that noise when calling a hotline (doesn’t matter if it’s an internal > call or it comes from external) ? Help would be really appreciated! > > > > > > > > > > Best regards > > > > Stefan > > > > > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
