Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf
[general]
register =>
user:secret:user@sipserver.com:8080<http://user:secret:user@sipserver.com:8080>
as long as I have just the above entry, I am able to receive incoming calls.
Now I would like to setup outgoing calls too. So I create a new section in
sip.conf
[sipserverout]
type=peer
secret=secret
username=user
fromuser=user
fromdomain=sipserver.com <http://sipserver.com>
host=sipserver.com <http://sipserver.com>
port=8080
context=default
with the above configuration I can successfully dial out using dial(
SIP/{$EXTEN}@sipserverout)
but now when I call my incoming number, I get a busy or invalid number
signal. If I coment out sipserverout section, I could receive incoming calls
again.
So I turned on sip debug on CLI. and it appears to me that the following is
happening. astreisk takes the incoming call and tries to match it with a
section with the same hostname. Now the reverse IP lookup on
109.147.41.48<http://109.147.41.48>return
sipserver.com <http://sipserver.com> (which is correct), so it is trying
to
send the call to sipserverout which is essentially back to the same server
where it came from (Notice the statement "Found peer
'sipserverout'" in the
sip debug logs below). This creates an endless loop and the equipment at the
other end terminates the call.
According to all the examples I have seen, my setup is the correct setup and
everyone seems to be using it. but it does not work for me. I am deperately
looking for a solution. Please help.
I am using asterisk 1.2.0 beta 1 on FC1.
Here is the sip debug dump when a call is coming.
<-- SIP read from 109.147.41.48:8080 <http://109.147.41.48:8080>:
INVITE sip:s@66.197.70.80:5050 SIP/2.0
Record-Route: <sip:209.47.41.48:80
<http://209.47.41.48:80>;ftag=2C996308-10F9;lr=on>
Via: SIP/2.0/UDP 209.47.41.48:80
<http://209.47.41.48:80>;branchz9hG4bK03a4.da6a926.0
Via: SIP/2.0/UDP 209.47.41.61:5060
<http://209.47.41.61:5060>;rport=53084;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK4BB6EA6
From: <sip:0000123456@209.47.41.61>;tag=2C996308-10F9
To: <sip:16166739282@209.47.41.48>
Date: Thu, 06 Oct 2005 08:13:58 GMT
Call-ID: FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61
Supported: timer
Min-SE: 1800
Cisco-Guid: 4208765565-896995802-2793406481-2459445924
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 4
Remote-Party-ID:
<sip:0000123456@109.147.41.48>;party=calling;screen=yes;privacy=off
Timestamp: 1128586438
Contact: <sip:0000123456@109.147.41.48:53084>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 369
hint: NAThelper
hint: SDP rewritten
hint: usrloc applied
hint: NAT...
v=0
o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4
209.47.41.61<http://209.47.41.61>
s=SIP Call
c=IN IP4 109.147.41.48 <http://109.147.41.48>
t=0 0
m=audio 53870 RTP/AVP 0 8 18 3 101
c=IN IP4 109.147.41.48 <http://109.147.41.48>
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes
--- (26 headers 16 lines)---
Using INVITE request as basis request -
FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61
Sending to 109.147.41.48 <http://109.147.41.48> : 80 (non-NAT)
Found peer 'sipserverout'
Reliably Transmitting (no NAT) to 209.47.41.48:80
<http://209.47.41.48:80>:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 209.47.41.48:80
<http://209.47.41.48:80>;branchz9hG4bK03a4.da6a926.0
Via: SIP/2.0/UDP 209.47.41.61:5060
<http://209.47.41.61:5060>;x-route-tag="tgrp:sroutetor1";branch=z9hG4bK4BB6EA6
From: <sip:0000123456@109.147.41.48 >;tag=2C996308-10F9
To: <sip:16166739282@109.147.41.48 >;tag=as1b7fff99
Call-ID: FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:s@66.197.70.80:5050>
Proxy-Authenticate: Digest realm="asterisk",
nonce="6d00a83d"
Content-Length: 0
---
Scheduling destruction of call '
FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61' in 15000 ms
<-- SIP read from 109.147.41.48:8080 <http://109.147.41.48:8080>:
ACK sip:s@66.197.70.80:5050 SIP/2.0
Via: SIP/2.0/UDP 109.147.41.48:8080
<http://109.147.41.48:8080>;branchz9hG4bK03a4.da6a926.0
From: <sip:0000123456@109.147.41.48>;tag=2C996308-10F9
Call-ID: FADDE365-357711DA-80F5C727-E0F535F0@209.47.41.61
To: <sip:16166739282@109.147.41.48>;tag=as1b7fff99
CSeq: 101 ACK
User-Agent: Phone Server 1
Content-Length: 0
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