[Asterisk-Users] Outgoing SIP to SER causes LOOP BACK message

[David Waugh]

> Hello fellow asterisk people!
> 
> I have Asterisk listening on port 5061 and SER on port 5060.
> 
> Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
> 
> My problems are with SIP. I can make incoming calls from SIP to asterisk
> and to any of the other networks, but when I try to make an outgoing call
> from Asterisk to SER I see the following in Asterisk:
> 
  sip debug
SIP Debugging Enabled
    -- Executing Ringing("H323/ip$192.219.85.57:2488/23038", "") in new
stack
    -- Executing Dial("H323/ip$192.219.85.57:2488/23038",
"sip/290 at sip_proxy-out|20|r") in new stack
We're at 192.219.85.57 port 15916
Answering/Requesting with root capability 0x4 (ulaw)
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:290 at fedcore2.eicon.com SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>
Contact: <sip:Asterisk at 192.219.85.57:5061>
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 25 Jul 2005 08:51:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 17396 17396 IN IP4 192.219.85.57
s=session
c=IN IP4 192.219.85.57
t=0 0
m=audio 15916 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
 (no NAT) to 192.219.85.57:5060
    -- Called 290 at sip_proxy-out


Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.3 (i386/linux))
Content-Length: 0
Warning: 392 192.219.85.57:5060 "Noisy feedback tells:  pid=30409
req_src_ip=192.219.85.57 req_src_port=5061 in_uri=sip:290 at fedcore2.eicon.com
out_uri=sip:290 at 192.219.85.57:5061 via_cnt==1"


9 headers, 0 lines


Sip read:
INVITE sip:290 at 192.219.85.57:5061 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:192.219.85.57;ftag=as533da407;lr=on>
Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>
Contact: <sip:Asterisk at 192.219.85.57:5061>
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 25 Jul 2005 08:51:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 17396 17396 IN IP4 192.219.85.57
s=session
c=IN IP4 192.219.85.57
t=0 0
m=audio 15916 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

15 headers, 8 lines
Transmitting (no NAT):
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>;tag=as533da407
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:Asterisk at 192.219.85.57:5061>
Content-Length: 0


 to 192.219.85.57:5060


Sip read:
ACK sip:290 at 192.219.85.57:5061 SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
To: <sip:290 at fedcore2.eicon.com>;tag=as533da407
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.9.3 (i386/linux))
Content-Length: 0


8 headers, 0 lines


Sip read:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>;tag=as533da407
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:Asterisk at 192.219.85.57:5061>
Content-Length: 0


10 headers, 0 lines
    -- Got SIP response 482 "Loop Detected" back from 192.219.85.57
Transmitting:
ACK sip:290 at fedcore2.eicon.com SIP/2.0
Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805
From: "223" <sip:Asterisk at 192.219.85.57:5061>;tag=as533da407
To: <sip:290 at fedcore2.eicon.com>;tag=as533da407
Contact: <sip:Asterisk at 192.219.85.57:5061>
Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118 at 192.219.85.57
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0 

> Want I want to happen is the call to go out through Asterisk - to SER (as
> SER knows where the SIP extension is) - and then onto the extension of the
> person to call.
> 
> In my sip.conf I have the following:
> 
 [general]
context=sip-incoming                    ; Default context for incoming calls
autocreatepeer=yes

recordhistory=yes               ; Record SIP history by default
;realm=fedcore2.eicon.com       ; Realm for digest authentication
port=5061                       ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=no                    ; Enable DNS SRV lookups on outbound calls
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference

register =>Asterisk:asterisk at fedcore2.eicon.com

[sip_proxy-out]
type=peer
secret=asterisk
username=Asterisk
fromuser=Asterisk
host=fedcore2.eicon.com
dtmfmode=inband 


> In my extensions.conf I have 
> 
> exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r)
> 
> So that dialing an extension 5XXX rings sip extension XXX.
> 
> I also the following context to catch incoming SIP calls.
> [sip-incoming]
> exten=>s,1,Wait,1
> exten =>s,2,Goto(default,384220,1)
> exten =>5000,1,Goto(default,384220,1)
> exten =>_9.,1,Goto(default,${EXTEN:1},1)
> 
> Why am I unable to make outgoing SIP calls?
> 
> I have also not made any changes to my DNS SVR settings (in case I need
> to???)
> 
> Many thanks for your help. I am probably doing something obvious wrong!
> 
> Thanks
> David
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