Hello.
I am configuring a trunk with IPBrick 5.2 (Lisbon) and Asterisk 11.2.1 (Oporto).
At Oporto I have sip.conf
[lisboa]
context=internal
type=friend
insecure=invite
username=lisboa
secret=<password>
nat=yes
host=<lisbonIP>
port=<lisbonPort>
qualify=yes
canreinvite=no
At Lisbon I have sip.conf
[PortoLisboa]
type=peer
username=lisboa
secret=<password>
fromuser=lisboa
fromdomain=<oportoIP>
host=<oportoIP>
dtmfmode=rfc2833
authuser=lisboa
insecure=port,invite
qualify=no
canreinvite=no
context=local_noauth_processing
port=<oportoPort>
call-limit=0
and when I dial I have
[code]1373899903.60292(1373899903.60292) Call from user1 to 116 (initially to 62116)
– Executing [s@macro-CDRData2:33] MacroExit(“SIP/user1-000031a7”, “”) in new stack
– Executing [62116@local_processing:4] Dial(“SIP/user1-000031a7”, “SIP/116@InosatPortoLisboa|3600|TM(CDRData^SIP/user1-000031a7^1373899903.60292)”) in new stack
Audio is at port 5034
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to ::
INVITE sip:116@: SIP/2.0
Via: SIP/2.0/UDP :;branch=z9hG4bK00abdf78
From: “User” <sip:portolisboa@>;tag=as7922b9f7
To: <sip:116@:>
Contact: <sip:portolisboa@:>
Call-ID: 7dea5f752d2b79c832211c4a2af3a0e9@
CSeq: 102 INVITE
User-Agent: IPBrick
Max-Forwards: 70
Date: Mon, 15 Jul 2013 14:51:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 463
v=0
o=root 6757 6757 IN IP4
s=session
c=IN IP4
t=0 0
m=audio 5034 RTP/AVP 0 8 3 97 110 111 7 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:110 speex/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
– Called 116@InosatPortoLisboa
sip*CLI>
<— SIP read from :49220 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP :;branch=z9hG4bK00abdf78
From: “User” <sip:portolisboa@>;tag=as7922b9f7
To: <sip:116@:>;tag=1E255E54-77AFCF33
CSeq: 102 INVITE
Call-ID: 7dea5f752d2b79c832211c4a2af3a0e9@
Contact: sip:polycomporto@192.168.2.248
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.4.0061
Content-Length: 0
<------------->
— (9 headers 0 lines) —
sip*CLI>
<— SIP read from :49220 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP :;branch=z9hG4bK00abdf78
From: “User” <sip:portolisboa@>;tag=as7922b9f7
To: <sip:116@:>;tag=1E255E54-77AFCF33
CSeq: 102 INVITE
Call-ID: 7dea5f752d2b79c832211c4a2af3a0e9@
Contact: sip:polycomporto@192.168.2.248
User-Agent: PolycomSoundStationIP-SPIP_6000-UA/3.0.4.0061
Allow-Events: talk,hold,conference
Call-Info: sip:sip..com;appearance-index=1
Content-Length: 0 [/code]
Although I have nothing at Oporto’s Asterisk so the extension 116 won’t really ring.
May you please tell me what am I doing wrong here? thanks