I can make all kinds of calls, inbound from outside to DID assigned on Skype for Business through FreePBX, I can call from FreePBX extension to Skype for Business ext, I can call from that extension to any number outside.
But only when I am trying to call from Skype for Business to outside through freePBX it says the number doesn’t exist check the number and try again.
I am calling from INVITE 3072481662 to +13201009553
The 307 number is assigned on Skype for Business DID.
Target Number: 3201009553 is PSTN Number
My Skype for Business server is 192.168.20.25
My FreePBX IP is 10.0.1.3
Call Logs are attached:>
freepbx*CLI> sip set debug on
SIP Debugging enabled<— SIP read from TCP:192.168.20.25:54060 —>
INVITE sip:[email protected];user=phone SIP/2.0
FROM: "adminM"sip:+13072481662;[email protected];user=phone;epid=075CFA3A59;tag=ba65c54576
TO: sip:[email protected];user=phone
CSEQ: 83955 INVITE
CALL-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b
CONTACT: sip:SBG-FE01.SkypeForBusiness.net:5060;transport=Tcp;maddr=192.168.20.25;ms-opaque=72f0d0298b3a73fe
CONTENT-LENGTH: 345
SUPPORTED: 100rel
USER-AGENT: RTCC/6.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
P-ASSERTED-IDENTITY: "adminM"sip:[email protected],tel:+13072481662;ext=44124
Privacy: id
Allow: CANCEL,BYE,INVITE,PRACK,UPDATEv=0
o=- 27555 1 IN IP4 192.168.20.25
s=session
c=IN IP4 192.168.20.25
b=CT:1000
t=0 0
m=audio 54564 RTP/AVP 97 101 13 0 8
c=IN IP4 192.168.20.25
a=rtcp:54565
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
— (16 headers 18 lines) —
Sending to 192.168.20.25:54060 (NAT)
Sending to 192.168.20.25:54060 (NAT)
Using INVITE request as basis request - b2e83a43-1878-4d99-9d59-c86ade39cc93
No matching peer for ‘+13072481662;ext=44124’ from ‘192.168.20.25:54060’
Found RTP audio format 97
Found RTP audio format 101
Found RTP audio format 13
Found RTP audio format 0
Found RTP audio format 8
Found unknown media description format RED for ID 97
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 13
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.20.25:54564
Looking for +13201009553 in from-sip-external (domain 10.0.1.3)
sip_route_dump: route/path hop: sip:SBG-FE01.SkypeForBusiness.net:5060;transport=Tcp;maddr=192.168.20.25;ms-opaque=72f0d0298b3a73fe<— Transmitting (NAT) to 192.168.20.25:54060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b;received=192.168.20.25;rport=54060
From: "adminM"sip:+13072481662;[email protected];user=phone;epid=075CFA3A59;tag=ba65c54576
To: sip:[email protected];user=phone
Call-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
CSeq: 83955 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060;transport=tcp
Content-Length: 0<------------>
[2019-02-20 14:05:18] WARNING[18850][C-000001c8]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from 192.168.20.25”
Audio is at 19398
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP<— Reliably Transmitting (NAT) to 192.168.20.25:54060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bKdcd9998b;received=192.168.20.25;rport=54060
From: "adminM"sip:+13072481662;[email protected];user=phone;epid=075CFA3A59;tag=ba65c54576
To: sip:[email protected];user=phone;tag=as3fb52f4e
Call-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
CSeq: 83955 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060;transport=tcp
Content-Type: application/sdp
Content-Length: 278v=0
o=root 1366665202 1366665202 IN IP4 66.23.202.104
s=Asterisk PBX 13.22.0
c=IN IP4 66.23.202.104
t=0 0
m=audio 19398 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv<------------>
<— SIP read from TCP:192.168.20.25:54060 —>
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
FROM: sip:+13072481662;[email protected];user=phone;epid=075CFA3A59;tag=ba65c54576
TO: sip:[email protected];user=phone;tag=as3fb52f4e
CSEQ: 83955 ACK
CALL-ID: b2e83a43-1878-4d99-9d59-c86ade39cc93
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.20.25:54060;branch=z9hG4bK69fead56
CONTENT-LENGTH: 0
USER-AGENT: RTCC/6.0.0.0 MediationServer