o o
2011-Aug-16 00:40 UTC
[asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?
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:Unknown at 1.2.3.4>;tag=as438c582c To: <sip:um.outlook.com> Contact: <sip:Unknown at 1.2.3.4:5061;transport=TLS> Call-ID: 67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061 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:Unknown at 1.2.3.4>;tag=as438c582c To: <sip:um.outlook.com>;tag=b4ec76231 Call-ID: 67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061 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 '67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061' 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:999 at um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: "Test User" <sip:210 at 1.2.3.4>;tag=as746bc17a To: <sip:999 at um.outlook.com> Contact: <sip:210 at 1.2.3.4:5061;transport=TLS> Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 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:210 at 1.2.3.4>;tag=as746bc17a To: <sip:999 at um.outlook.com> Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 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:210 at 1.2.3.4>;tag=as746bc17a To: <sip:999 at um.outlook.com>;tag=aprqngfrt-hm4td720000c6 Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 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:999 at um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: "Test User" <sip:210 at 1.2.3.4>;tag=as746bc17a To: <sip:999 at um.outlook.com>;tag=aprqngfrt-hm4td720000c6 Contact: <sip:210 at 1.2.3.4:5061;transport=TLS> Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 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 '535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061' Method: INVITE TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110815/88697813/attachment-0001.htm>
Alex Vishnev
2011-Aug-16 11:57 UTC
[asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?
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:Unknown at 1.2.3.4>;tag=as438c582c > To: <sip:um.outlook.com> > Contact: <sip:Unknown at 1.2.3.4:5061;transport=TLS> > Call-ID: 67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061 > 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:Unknown at 1.2.3.4>;tag=as438c582c > To: <sip:um.outlook.com>;tag=b4ec76231 > Call-ID: 67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061 > 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 '67f260947dae7c27121ca30e5ee9d3ef at 1.2.3.4:5061' 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:999 at um.outlook.com SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" <sip:210 at 1.2.3.4>;tag=as746bc17a > To: <sip:999 at um.outlook.com> > Contact: <sip:210 at 1.2.3.4:5061;transport=TLS> > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 > 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:210 at 1.2.3.4>;tag=as746bc17a > To: <sip:999 at um.outlook.com> > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 > 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:210 at 1.2.3.4>;tag=as746bc17a > To: <sip:999 at um.outlook.com>;tag=aprqngfrt-hm4td720000c6 > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 > 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:999 at um.outlook.com SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" <sip:210 at 1.2.3.4>;tag=as746bc17a > To: <sip:999 at um.outlook.com>;tag=aprqngfrt-hm4td720000c6 > Contact: <sip:210 at 1.2.3.4:5061;transport=TLS> > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061 > 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 '535c9a2d211f19be5528b2ce7dbce0d4 at 1.2.3.4:5061' 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110816/301d297f/attachment.htm>