Hi,
Both trixbox and Sipura are configured behind NAT and connected over WAN
Able to make outbound calls from pbx user the 3102 PSTN line.
Teixbox version 2.8. Sipura on 5.1 firmware.
When dialing SPA its just rings rings and the line busy outs. Though i can see the call coming to PBX, is not being picked up by the inbound route.
Logs of PBX when dialing Pots line connected to 3102
###################################
<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from UDP://121.246.xx.xx:5061 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.0.110:5061;branch=z9hG4bK-ad116899
From: Local Pstn sip:[email protected];tag=e1f6ac6bfcd95611o1
To: sip:[email protected]
Remote-Party-ID: Local Pstn sip:[email protected];screen=yes;party=calling
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“2pstn”,realm=“asterisk”,nonce=“5af999e7”,uri=“sip:[email protected]”,algorithm=MD5,response="346f974d4c184f2f680f51c2c3b0fd62"
Contact: Local Pstn sip:[email protected]:5061
Expires: 240
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 440
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 154881 154881 IN IP4 10.10.0.110
s=-
c=IN IP4 10.10.0.110
t=0 0
m=audio 16446 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (16 headers 20 lines) —
Sending to 121.246.xx.xx : 5061 (NAT)
Using INVITE request as basis request - [email protected]
Found user ‘2pstn’ for ‘2pstn’
<— Reliably Transmitting (NAT) to 121.246.xx.xx:5061 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.10.0.110:5061;branch=z9hG4bK-ad116899;received=121.246.xx.xx
From: Local Pstn sip:[email protected];tag=e1f6ac6bfcd95611o1
To: sip:[email protected];tag=as7b122c84
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.9-samy-r27
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
trixbox1*CLI>
<— SIP read from UDP://121.246.xx.xx:5061 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.0.110:5061;branch=z9hG4bK-ad116899
From: Local Pstn sip:[email protected];tag=e1f6ac6bfcd95611o1
To: sip:[email protected];tag=as7b122c84
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“2pstn”,realm=“asterisk”,nonce=“5af999e7”,uri=“sip:[email protected]”,algorithm=MD5,response="346f974d4c184f2f680f51c2c3b0fd62"
Contact: Local Pstn sip:[email protected]:5061
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
<------------->
— (11 headers 0 lines) —
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
trixbox1*CLI>
<— SIP read from UDP://10.10.50.3:5060 —>
<------------->
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
trixbox1*CLI>
##############################
Trunk configuration PBX.
disallow=all
allow=ulaw
type=friend
canreinvite=yes
insecure=very
context=from-trunk
dtmfmode=rfc2833
host=121.246.xx.xx
incominglimit=1
nat=yes
port=5061
qualify=yes
username=2pstn
secret=12345
SPA configured as per guide in the forums.
Please Help, i very important for me to resolve this in nxt 24 Hrs. i cant take paid support too.
Thanks