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 <[email protected]> > To: o o <[email protected]>; Asterisk Users Mailing List - Non-Commercial > Discussion <[email protected]> > 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:[email protected]>;tag=as438c582c >> To: <sip:um.outlook.com> >> Contact: <sip:[email protected]:5061;transport=TLS> >> Call-ID: [email protected]: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:[email protected]>;tag=as438c582c >> To: <sip:um.outlook.com>;tag=b4ec76231 >> Call-ID: [email protected]: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 '[email protected]: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:[email protected] SIP/2.0 >> Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 >> Max-Forwards: 70 >> From: "Test User" <sip:[email protected]>;tag=as746bc17a >> To: <sip:[email protected]> >> Contact: <sip:[email protected]:5061;transport=TLS> >> Call-ID: [email protected]: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:[email protected]>;tag=as746bc17a >> To: <sip:[email protected]> >> Call-ID: [email protected]: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:[email protected]>;tag=as746bc17a >> To: <sip:[email protected]>;tag=aprqngfrt-hm4td720000c6 >> Call-ID: [email protected]: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:[email protected] SIP/2.0 >> Via: SIP/2.0/TLS 1.2.3.4:5061;branch=z9hG4bK70149e02 >> Max-Forwards: 70 >> From: "Test User" <sip:[email protected]>;tag=as746bc17a >> To: <sip:[email protected]>;tag=aprqngfrt-hm4td720000c6 >> Contact: <sip:[email protected]:5061;transport=TLS> >> Call-ID: [email protected]: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 '[email protected]:5061' Method: INVITE >> >> >> TIA >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? 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