Freepbx and Exchange 2013 UM Gateway

cbotts · January 17, 2014, 4:17pm

Ok so I we installed freepbx 32 bit. Setup free pbx and micrsoft to talk to each other via a trunk.

from the local extensions we are able to call into exchange and from extension to extension we are able to leave voicemails and everything is working as expected.

However when and outside call comes in and rings and then tries to go to the um mailbox, the rtp stops working and freepbx hangs up the call because of lack of rtp.

We tried all the canreinvite options and things to no end.

Does anyone have any tricks they have seen with this issue?

Both the freepbx server and the exchange are on a public ip address and within the same network as well.

dicko · January 17, 2014, 6:38pm

Does your server agree with your asterisk box on which range to accept the audio payload (rtp) on? In asterisk it is defined in /etc/asterisk/rtp.conf

cbotts · January 17, 2014, 8:49pm

So after some more testing, it appears to work if we answer the call and then transfer then after talking to someone. Here are to captures. One from the server to the carrier and one from the server to the exchange.

Exchange log -------------------
INVITE sip:x.x.x.68:5065 SIP/2.0.
Via: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK129af145.
Max-Forwards: 70.
From: “WIRELESS CALLER” sip:[email protected];tag=as7cf2fa3b.
To: sip:x.x.x.68:5065.
Contact: sip:[email protected]:5060;transport=TCP.
Call-ID: [email protected]:5060.
CSeq: 102 INVITE.
User-Agent: FPBX-2.11.0(11.3.0).
Date: Fri, 17 Jan 2014 20:35:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Diversion: tel:9546;reason=no-answer;screen=no;privacy=off.
Content-Type: application/sdp.
Content-Length: 282.
.
v=0.
o=root 957332019 957332019 IN IP4 x.x.x.77.
s=Asterisk PBX 11.3.0.
c=IN IP4 x.x.x.77.
t=0 0.
m=audio 25888 RTP/AVP 0 8 3 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

T x.x.x.68:5065 -> x.x.x.77:33014 [AP]
SIP/2.0 100 Trying.
FROM: "WIRELESS CALLER"sip:[email protected];tag=as7cf2fa3b.
TO: sip:x.x.x.68:5065.
CSEQ: 102 INVITE.
CALL-ID: [email protected]:5060.
VIA: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK129af145.
CONTENT-LENGTH: 0.
.

T x.x.x.68:5065 -> x.x.x.77:33014 [AP]
SIP/2.0 180 Ringing.
FROM: "WIRELESS CALLER"sip:[email protected];tag=as7cf2fa3b.
TO: sip:x.x.x.68:5065;tag=f8fccfb21b;epid=DDAE5739E9.
CSEQ: 102 INVITE.
CALL-ID: [email protected]:5060.
VIA: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK129af145.
CONTACT: sip:SN-EX2013.corp.syndeonetworks.com:5065;transport=Tcp;maddr=x.x.x.68;automata;text;audio;video;image.
CONTENT-LENGTH: 0.
ALLOW: CANCEL.
ALLOW: BYE.
ALLOW: UPDATE.
ALLOW: PRACK.
P-ASSERTED-IDENTITY: tel:9546;phone-context=dialstring.
SERVER: RTCC/5.0.0.0 MSExchangeUM/15.00.0712.000.
.

T x.x.x.68:5065 -> x.x.x.77:33014 [AP]
SIP/2.0 200 OK.
FROM: "WIRELESS CALLER"sip:[email protected];tag=as7cf2fa3b.
TO: sip:x.x.x.68:5065;tag=f8fccfb21b;epid=DDAE5739E9.
CSEQ: 102 INVITE.
CALL-ID: [email protected]:5060.
VIA: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK129af145.
CONTACT: sip:SN-EX2013.corp.syndeonetworks.com:5065;transport=Tcp;maddr=x.x.x.68;automata;text;audio;video;image.
CONTENT-LENGTH: 304.
PRIORITY: Normal.
SUPPORTED: Replaces.
SUPPORTED: timer.
SUPPORTED: 100rel.
CONTENT-TYPE: application/sdp.
ALLOW: ACK.
P-ASSERTED-IDENTITY: tel:9546;phone-context=dialstring.
SERVER: RTCC/5.0.0.0 MSExchangeUM/15.00.0712.000.
Content-ID: 22b30268-0207-4c23-9b1b-917951db096b.
Allow: CANCEL,BYE,INVITE,MESSAGE,INFO,SERVICE,OPTIONS,BENOTIFY,NOTIFY,PRACK,UPDATE.
Session-Expires: 1800;refresher=uas.
Min-SE: 90.
.
v=0.
o=- 21 0 IN IP4 x.x.x.68.
s=session.
c=IN IP4 x.x.x.68.
b=CT:1000000.
t=0 0.
m=audio 31570 RTP/AVP 0 8 101.
c=IN IP4 x.x.x.68.
a=rtcp:31571.
a=label:main-audio.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16,36.
a=ptime:20.

T x.x.x.77:33014 -> x.x.x.68:5065 [AP]
ACK sip:SN-EX2013.corp.syndeonetworks.com:5065;transport=Tcp;maddr=x.x.x.68 SIP/2.0.
Via: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK461299d9.
Max-Forwards: 70.
From: “WIRELESS CALLER” sip:[email protected];tag=as7cf2fa3b.
To: sip:x.x.x.68:5065;tag=f8fccfb21b.
Contact: sip:[email protected]:5060;transport=TCP.
Call-ID: [email protected]:5060.
CSeq: 102 ACK.
User-Agent: FPBX-2.11.0(11.3.0).
Content-Length: 0.
.

T x.x.x.77:33014 -> x.x.x.68:5065 [AP]
OPTIONS sip:x.x.x.68 SIP/2.0.
Via: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK78526c9f.
Max-Forwards: 70.
From: “Unknown” sip:[email protected];tag=as1570c45d.
To: sip:x.x.x.68.
Contact: sip:[email protected]:5060;transport=TCP.
Call-ID: [email protected]:5060.
CSeq: 102 OPTIONS.
User-Agent: FPBX-2.11.0(11.3.0).
Date: Fri, 17 Jan 2014 20:35:54 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.

T x.x.x.68:5065 -> x.x.x.77:33014 [AP]
SIP/2.0 405 Method Not Allowed.
FROM: "Unknown"sip:[email protected];tag=as1570c45d.
TO: sip:x.x.x.68;tag=f312691940.
CSEQ: 102 OPTIONS.
CALL-ID: [email protected]:5060.
VIA: SIP/2.0/TCP x.x.x.77:5060;branch=z9hG4bK78526c9f.
CONTENT-LENGTH: 0.
ALLOW: NOTIFY.
ALLOW: BENOTIFY.
SERVER: RTCC/5.0.0.0 MSExchangeUM/15.00.0712.000.
.

Carrier Log ---------------
U x.x.x.125:5060 -> x.x.x.77:5060
INVITE sip:[email protected]:5060 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.125:5060;branch=z9hG4bK+ed552f2b782c8fb3bc2de2d7ccce09ac1+sip+1+a6ca1c85.
From: “WIRELESS CALLER” sip:[email protected]:5060;tag=x.x.x.125+1+a9a5bf7d+a272deed.
To: sip:[email protected]:5060.
CSeq: 12021 INVITE.
Expires: 180.
Supported: 100rel.
Content-Length: 259.
Call-Info: sip:x.x.x.125:5060;method=“NOTIFY;Event=telephone-event;Duration=2000”.
Contact: sip:[email protected]:5060.
Content-Type: application/sdp.
Call-ID: 0gQAAC8WAAACBAAALxYAAFk9vqaXmCLMG/[email protected].
Max-Forwards: 68.
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE, INFO.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
P-Asserted-Identity: “WIRELESS CALLER” sip:[email protected]:5060.
.
v=0.
o=Sonus_UAC 4690592818814 4690592818814 IN IP4 x.x.x.125.
s=-.
c=IN IP4 x.x.x.125.
t=0 0.
m=audio 41808 RTP/AVP 0 8 101.
a=sendrecv.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=maxptime:20.

U x.x.x.77:5060 -> x.x.x.125:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP x.x.x.125:5060;branch=z9hG4bK+ed552f2b782c8fb3bc2de2d7ccce09ac1+sip+1+a6ca1c85;received=x.x.x.125.
From: “WIRELESS CALLER” sip:[email protected]:5060;tag=x.x.x.125+1+a9a5bf7d+a272deed.
To: sip:[email protected]:5060.
Call-ID: 0gQAAC8WAAACBAAALxYAAFk9vqaXmCLMG/[email protected].
CSeq: 12021 INVITE.
Server: FPBX-2.11.0(11.3.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:[email protected]:5060.
Content-Length: 0.
.

U x.x.x.77:5060 -> x.x.x.125:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP x.x.x.125:5060;branch=z9hG4bK+ed552f2b782c8fb3bc2de2d7ccce09ac1+sip+1+a6ca1c85;received=x.x.x.125.
From: “WIRELESS CALLER” sip:[email protected]:5060;tag=x.x.x.125+1+a9a5bf7d+a272deed.
To: sip:[email protected]:5060;tag=as0ab27ca9.
Call-ID: 0gQAAC8WAAACBAAALxYAAFk9vqaXmCLMG/[email protected].
CSeq: 12021 INVITE.
Server: FPBX-2.11.0(11.3.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:[email protected]:5060.
Content-Length: 0.
.

U x.x.x.77:5060 -> x.x.x.125:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.125:5060;branch=z9hG4bK+ed552f2b782c8fb3bc2de2d7ccce09ac1+sip+1+a6ca1c85;received=x.x.x.125.
From: “WIRELESS CALLER” sip:[email protected]:5060;tag=x.x.x.125+1+a9a5bf7d+a272deed.
To: sip:[email protected]:5060;tag=as0ab27ca9.
Call-ID: 0gQAAC8WAAACBAAALxYAAFk9vqaXmCLMG/[email protected].
CSeq: 12021 INVITE.
Server: FPBX-2.11.0(11.3.0).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Contact: sip:[email protected]:5060.
Content-Type: application/sdp.
Content-Length: 237.
.
v=0.
o=root 1936555700 1936555700 IN IP4 x.x.x.77.
s=Asterisk PBX 11.3.0.
c=IN IP4 x.x.x.77.
t=0 0.
m=audio 15130 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.

U x.x.x.125:5060 -> x.x.x.77:5060
ACK sip:[email protected]:5060 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.125:5060;branch=z9hG4bK+24b74fc53dd10f5b4e17d7720b33e5e01+sip+1+a6ca1ca3.
Call-ID: 0gQAAC8WAAACBAAALxYAAFk9vqaXmCLMG/[email protected].
From: “WIRELESS CALLER” sip:[email protected]:5060;tag=x.x.x.125+1+a9a5bf7d+a272deed.
To: sip:[email protected]:5060;tag=as0ab27ca9.
CSeq: 12021 ACK.
Contact: sip:[email protected]:5060.
Content-Length: 0.
Max-Forwards: 69.
.

U x.x.x.77:5060 -> x.x.x.125:5060
OPTIONS sip:x.x.x.125 SIP/2.0.
Via: SIP/2.0/UDP x.x.x.77:5060;branch=z9hG4bK02ac395f.
Max-Forwards: 70.
From: “Unknown” sip:[email protected];tag=as69037181.
To: sip:x.x.x.125.
Contact: sip:[email protected]:5060.
Call-ID: [email protected]:5060.
CSeq: 102 OPTIONS.
User-Agent: FPBX-2.11.0(11.3.0).
Date: Fri, 17 Jan 2014 20:35:55 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.

U x.x.x.125:5060 -> x.x.x.77:5060
SIP/2.0 200 OK.
From: “Unknown” sip:[email protected];tag=as69037181.
To: sip:x.x.x.125;tag=sip+1+5a970008+72a3dd4f.
Via: SIP/2.0/UDP x.x.x.77:5060;received=x.x.x.77;branch=z9hG4bK02ac395f.
Server: SIP/2.0.
Max-Forwards: 70.
Contact: sip:[email protected]:5060.
Call-ID: [email protected]:5060.
CSeq: 102 OPTIONS.
User-Agent: FPBX-2.11.0(11.3.0).
Date: Fri, 17 Jan 2014 20:35:55 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Content-Length: 0.
.

cbotts · January 21, 2014, 3:28pm

So another twist in the story. If I route a call through an IVR, and then it transfers to voicemail it works as intended. But if I route through a ring group and then directly to the voicemail for exchange. I get no audio.