<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">does that mean you try setting dtmfmode=inband and made sure that 101 was no longer present in SDP? Still you got 488?<div>good luck with that ;-)</div><div><br><div><div>On Aug 16, 2011, at 1:04 PM, o o wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>Alex,</span></div><div><span> 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. <br></span></div><div><br><span></span></div><div><span>Trying to get someone with a brain at MS to work with me on this.</span></div><div><br><span></span></div><div><span></span></div><div><br></div><div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"><div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"><font size="2" face="Arial"><hr size="1"><b><span style="font-weight:bold;">From:</span></b> Alex Vishnev <<a href="mailto:alex9134@gmail.com">alex9134@gmail.com</a>><br><b><span style="font-weight:
bold;">To:</span></b> o o <<a href="mailto:bj_5150@yahoo.com">bj_5150@yahoo.com</a>>; Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br><b><span style="font-weight: bold;">Sent:</span></b> Tuesday, August 16, 2011 4:57 AM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] Asterisk -> Office 365 Unified Messaging... anyone done it?<br></font><br><div id="yiv1162198228">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<br><div><div>On Aug 15, 2011, at 8:40 PM, o o wrote:</div><br class="yiv1162198228Apple-interchange-newline"><blockquote type="cite"><div><div style="color:#000;background-color:#fff;font-family:times new roman, new york, times, serif;font-size:12pt;"><div>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.<br></div><div><br></div><div>
</div><div style="margin:0in 0in 0.0001pt;"><span style="">Regarding
your service request </span><span style="font-size:11.0pt;"></span><span style="">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.</span></div><div style="margin:0in 0in 0.0001pt;"><br><span style=""></span></div><div style="margin:0in 0in 0.0001pt;"><span style="">Here is a SIP debug:</span></div><div style="margin:0in 0in 0.0001pt;"><br><span style=""></span></div><pre style="width:1156px;" dir="ltr">[2011-08-11 23:00:26] VERBOSE[17000] chan_sip.c: Reliably Transmitting (no NAT) to 65.55.174.100:5061:
OPTIONS <a rel="nofollow">sip:um.outlook.com</a> SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK16884162
Max-Forwards: 70
From: "Unknown" <<a rel="nofollow">sip:Unknown@1.2.3.4</a>>;tag=as438c582c
To: <<a rel="nofollow">sip:um.outlook.com</a>>
Contact: <<a rel="nofollow">sip:Unknown@1.2.3.4:5061;transport=TLS</a>>
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" <<a rel="nofollow">sip:Unknown@1.2.3.4</a>>;tag=as438c582c
To: <<a rel="nofollow">sip:um.outlook.com</a>>;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 <a rel="nofollow">sip:999@um.outlook.com</a> SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
Max-Forwards: 70
From: "Test User" <<a rel="nofollow">sip:210@1.2.3.4</a>>;tag=as746bc17a
To: <<a rel="nofollow">sip:999@um.outlook.com</a>>
Contact: <<a rel="nofollow">sip:210@1.2.3.4:5061;transport=TLS</a>>
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" <<a rel="nofollow">sip:210@1.2.3.4</a>>;tag=as746bc17a
To: <<a rel="nofollow">sip:999@um.outlook.com</a>>
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" <<a rel="nofollow">sip:210@1.2.3.4</a>>;tag=as746bc17a
To: <<a rel="nofollow">sip:999@um.outlook.com</a>>;tag=aprqngfrt-hm4td720000c6
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 <a rel="nofollow">sip:999@um.outlook.com</a> SIP/2.0
Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02
Max-Forwards: 70
From: "Test User" <<a rel="nofollow">sip:210@1.2.3.4</a>>;tag=as746bc17a
To: <<a rel="nofollow">sip:999@um.outlook.com</a>>;tag=aprqngfrt-hm4td720000c6
Contact: <<a rel="nofollow">sip:210@1.2.3.4:5061;transport=TLS</a>>
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</pre>
<div><span><br><br>TIA<br></span></div>
</div></div>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a rel="nofollow" target="_blank" href="http://www.api-digital.com/">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a rel="nofollow" target="_blank" href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a rel="nofollow" target="_blank" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br></div><br><br></div></div></div></div></blockquote></div><br></div></body></html>