Hi,
I am try to configure Asterisk as PBX system with two interfaces as
shown below. One interface pointing to the local subnet with a SIP phone
and another interface pointing to the external ISP SIP Sever.
SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external-
intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld
I am able to setup a call from the Phone to the outside world and I have
the audio (RTP packets) coming from the outside world being routed to my
phone
but the audio from my Phone IP(X.X) is not going out to the SIP-Server.
In fact I think it is not even reaching the Asterisk server because the
SDP in the 183 going to the phone has the IP address of the
external-inf(Z.Z.247.106) of the Asterisk PBX when it should actually
(Y.Y.47.149)
<--- Transmitting (NAT) to X.X.141.32:5060 --->
SIP/2.0 183 Session Progress^M
Via: SIP/2.0/UDP
X.X.141.32;branch=z9hG4bK87468d20000002f44b86a00400006f2b00000166;receiv
ed=X.X.141.32;rport=5060^M
From: "Irfan Lateef"
<sip:[email protected]>;tag=327f290e2e7^M
To: <sip:[email protected]>;tag=as24228e21^M
Call-ID: 876BAA6B36F644F7B4EF7BE5D4B7E8BD0x87468d20^M
CSeq: 2 INVITE^M
User-Agent: Asterisk PBX 1.6.0.17^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO^M
Supported: replaces, timer^M
Require: timer^M
Session-Expires: -1;refresher=uas^M
Contact: <sip:[email protected]>^M
Content-Type: application/sdp^M
Content-Length: 315^M
^M
v=0^M
o=root 1021147583 1021147583 IN IP4 Z.Z.247.106^M
s=Asterisk PBX 1.6.0.17^M
c=IN IP4 Z.Z.247.106^M
t=0 0^M
m=audio 18702 RTP/AVP 0 8 3 101^M
I have the following in the sip_nat.conf
localnet=Y.Y.47.149/255.255.0.0
externhost=Z.Z.247.106
externrefresh=10
fromdomain=att.com
nat=yes
qualify=yes
canreinvite=no
I think the SDP should have give the Y.Y.47.149 IP on the local net side
to the phone. But I am unable to figure how make it do that.
The Asterisk log shows this.
[Feb 25 11:06:30] VERBOSE[1449] logger.c: --
Executing [...@macro-dialout-trunk:19]
^[[1;36;40mDial^[[0;37;40m("^[[1;35;40mSIP/2005-19dc0db8^[[0;37;40m",
"^[[1;35;40mSIP/ATT-alpi016-IPFlex1/19084611234,300,^[[0;37;40m") in new
stack
[Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP
RTP TOS bits 184
[Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP
RTP CoS mark 5
[Feb 25 11:06:30] VERBOSE[1449] logger.c: -- Called
ATT-alpi016-IPFlex1/19084611234
[Feb 25 11:06:32] VERBOSE[1449] logger.c: --
SIP/ATT-alpi016-IPFlex1-19dda0f8 is making progress passing it to
SIP/2005-19dc0db8
[Feb 25 11:06:32] VERBOSE[1449] logger.c: Audio is at
Z.Z.247.106 port 18702
[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x4 (ulaw) to SDP
[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x8 (alaw) to SDP
[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x2 (gsm) to SDP
[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding
non-codec 0x1 (telephone-event) to SDP
[Feb 25 11:06:32] VERBOSE[1449] logger.c:
Any help is greatly appreciated.
Thanks and Regards,
Irfan Lateef
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