Re: [asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?
does that mean you try setting dtmfmode=inband and made sure that 101 was no longer present in SDP? Still you got 488? good luck with that ;-) On Aug 16, 2011, at 1:04 PM, o o wrote: > Alex, >Thanks for the pointers. Digging through some Cisco documentation linked > to as a guide for configuring CCM 8.0 with Office 365, it states that they > support 711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with > no luck. > > Trying to get someone with a brain at MS to work with me on this. > > > From: Alex Vishnev > To: o o ; Asterisk Users Mailing List - Non-Commercial > Discussion > Sent: Tuesday, August 16, 2011 4:57 AM > Subject: Re: [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" ;tag=as438c582c >> To: >> Contact: >> Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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" ;tag=as438c582c >> To: ;tag=b4ec76231 >> Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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@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:9...@um.outlook.com SIP/2.0 >> Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 >> Max-Forwards: 70 >> From: "Test User" ;tag=as746bc17a >> To: >> Contact: >> Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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" ;tag=as746bc17a >> To: >> Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4:5061 >> CSeq: 102 INVITE >> Content-Length: 0 >> >> <-> >> [2011-08-11 23
Re: [asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?
Alex, Thanks for the pointers. Digging through some Cisco documentation linked to as a guide for configuring CCM 8.0 with Office 365, it states that they support 711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with no luck. Trying to get someone with a brain at MS to work with me on this. From: Alex Vishnev To: o o ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, August 16, 2011 4:57 AM Subject: Re: [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" ;tag=as438c582c To: Contact: Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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" ;tag=as438c582c To: ;tag=b4ec76231 Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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@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:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: "Test User" ;tag=as746bc17a To: Contact: Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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" ;tag=as746bc17a To: Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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" ;tag=as746bc17a To: ;tag=aprqngfrt-hm4td72c6 Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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:9...@um.outlook.com SIP/2.0 Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 Max-Forwards: 70 From: "Test User" ;tag=as746bc17a To: ;tag=aprqngfrt-hm4td72c6 Contact: Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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@1.2.3.4:5061' Method: INVITE > > >TIA >-- >__
Re: [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" ;tag=as438c582c > To: > Contact: > Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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" ;tag=as438c582c > To: ;tag=b4ec76231 > Call-ID: 67f260947dae7c27121ca30e5ee9d3ef@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@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:9...@um.outlook.com SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" ;tag=as746bc17a > To: > Contact: > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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" ;tag=as746bc17a > To: > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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" ;tag=as746bc17a > To: ;tag=aprqngfrt-hm4td72c6 > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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:9...@um.outlook.com SIP/2.0 > Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 > Max-Forwards: 70 > From: "Test User" ;tag=as746bc17a > To: ;tag=aprqngfrt-hm4td72c6 > Contact: > Call-ID: 535c9a2d211f19be5528b2ce7dbce0d4@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@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 -- _ -- Bandwidth and Coloca
