[asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?

Tue Aug 16 06:57:49 CDT 2011

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
> --
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