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

Tue Aug 16 12:04:15 CDT 2011

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 <alex9134 at gmail.com>
To: o o <bj_5150 at yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
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" <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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