Asterisk -> Office 365 Unified Messaging… anyone done it?

Home » Asterisk Users » Asterisk -> Office 365 Unified Messaging… anyone done it?
Asterisk Users No Comments

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: <a href="mailto:67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4">67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4</a>: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: <a href="mailto:67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4">67f260947dae7c27121ca30e5ee9d3ef@1.2.3.4</a>: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:999@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: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>: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: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>: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-hm4td720000c6
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>: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@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-hm4td720000c6
Contact:
Call-ID: <a href="mailto:535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4">535c9a2d211f19be5528b2ce7dbce0d4@1.2.3.4</a>: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

This could be an unsupported codec. 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.

TIA