this could be an unsupported codec. Do you know if Office365 supports PCMU? I would also try to get rid of 101 (rfc2833) and see if that makes a difference On Aug 15, 2011, at 8:40 PM, o o wrote:
> Trying to make this work, and Office 365 support is useless, giving me the > following response when I asked them for help troubleshooting a 488 Not > Acceptable Here. > > Regarding your service request about configuring your PBX system with Office > 365, we do not support specific setups for PBX systems for Unified Messaging. > Please contact the vendor for more specific instructions and configurations. > > Here is a SIP debug: > > [2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no > NAT) to 65.55.174.100:5061: > OPTIONS sip:um.outlook.com SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 > Max-Forwards: 70 > From: "Unknown" <sip:[email protected]>;tag=as438c582c > To: <sip:um.outlook.com> > Contact: <sip:[email protected]:5061;transport=TLS> > Call-ID: [email protected]:5061 > CSeq: 102 OPTIONS > User-Agent: FPBX-2.8.1(1.8.5.0) > Date: Fri, 12 Aug 2011 06:00:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Length: 0 > > > --- > [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: > <--- SIP read from TLS:65.55.174.100:5061 ---> > SIP/2.0 200 OK > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162 > From: "Unknown" <sip:[email protected]>;tag=as438c582c > To: <sip:um.outlook.com>;tag=b4ec76231 > Call-ID: [email protected]:5061 > CSeq: 102 OPTIONS > ACCEPT: application/sdp > CONTENT-LENGTH: 0 > ALLOW: INVITE > ALLOW: BYE > ALLOW: CANCEL > ALLOW: OPTIONS > ALLOW: ACK > ALLOW: INFO > ALLOW: NOTIFY > SERVER: RTCC/3.5.0.0 > > <-------------> > [2011-08-11 23:00:27] VERBOSE[17375] chan_sip.c: --- (16 headers 0 lines) --- > [2011-08-11 23:00:27] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog > '[email protected]:5061' Method: OPTIONS > [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Audio is at 5061 > [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding codec 0x4 (ulaw) to > SDP > [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Adding non-codec 0x1 > (telephone-event) to SDP > [2011-08-11 23:00:47] VERBOSE[17999] chan_sip.c: Reliably Transmitting (no > NAT) to 65.55.174.100:5061: > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" <sip:[email protected]>;tag=as746bc17a > To: <sip:[email protected]> > Contact: <sip:[email protected]:5061;transport=TLS> > Call-ID: [email protected]:5061 > CSeq: 102 INVITE > User-Agent: FPBX-2.8.1(1.8.5.0) > Date: Fri, 12 Aug 2011 06:00:47 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 238 > > v=0 > o=root 1381221379 1381221379 IN IP4 1.2.3.4 > s=Asterisk PBX 1.8.5.0 > c=IN IP4 1.2.3.4 > t=0 0 > m=audio 17688 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > --- > [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: > <--- SIP read from TLS:65.55.174.100:5061 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > From: "Test User" <sip:[email protected]>;tag=as746bc17a > To: <sip:[email protected]> > Call-ID: [email protected]:5061 > CSeq: 102 INVITE > Content-Length: 0 > > <-------------> > [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- > [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: > <--- SIP read from TLS:65.55.174.100:5061 ---> > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > From: "Test User" <sip:[email protected]>;tag=as746bc17a > To: <sip:[email protected]>;tag=aprqngfrt-hm4td720000c6 > Call-ID: [email protected]:5061 > CSeq: 102 INVITE > Content-Length: 0 > > <-------------> > [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: --- (7 headers 0 lines) --- > [2011-08-11 23:00:47] VERBOSE[17375] chan_sip.c: Transmitting (no NAT) to > 65.55.174.100:5061: > ACK sip:[email protected] SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" <sip:[email protected]>;tag=as746bc17a > To: <sip:[email protected]>;tag=aprqngfrt-hm4td720000c6 > Contact: <sip:[email protected]:5061;transport=TLS> > Call-ID: [email protected]:5061 > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.5.0) > Content-Length: 0 > > > --- > [2011-08-11 23:00:48] VERBOSE[17000] chan_sip.c: Really destroying SIP dialog > '[email protected]:5061' Method: INVITE > > > TIA > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
