[Asterisk-Users] SIP - Loop detected
Trond Andersen
trond.andersen at tandberg.net
Thu Nov 17 00:19:19 MST 2005
Hi.
I have a small problem and was hoping for some pointers in the right
direction...
I have two sip phones registered at my asterisk 1.2.rc2 server.
They both use the asterisk IP in their SIP-URI:
phone1: SIP-URI=trond at asterisk.tsip.lab
phone2: SIP-URI=20170 at asterisk.tsip.lab
If I use the following dial command in my extensions.conf
-------------
exten => s,2,Dial(SIP/${ARG1},30,Cf)
-------------
the To: field is set to To <SIP:asterisk.tsip.lab>
That will not work in my case because I need the full SIP-URI to arrive
at the receiving end.
Therefore I have changed my dialplan to have the following dial command:
-------------
exten => s,2,Dial(SIP/${ARG1}@${SIPDOMAIN},30,Cf)
-------------
Now Asterisk tries
To <SIP:20170 at asterisk.tsip.lab>
But the result is a 482: Loop detected.....
What am I doing wrong? Anyone...
Trond
I add the sip debug. Asterisk has
IP: 10.47.8.89 which equals asterisk.tsip.lab
=============================================================================
<-- SIP read from 10.47.5.174:5060:
INVITE sip:20170 at 10.47.8.89 SIP/2.0
Via: SIP/2.0/UDP
10.47.5.174:5060;branch=z9hG4bKa7dc32132336e031fee20e54a036538e.1;rport
Call-ID: 0100ad00ac00f900 at 10.47.5.174
CSeq: 100 INVITE
Contact: <sip:trond at 10.47.5.174>
From: "tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A
To: <sip:20170 at 10.47.8.89>
Max-Forwards: 70
Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW))
Content-Type:application/sdp
Content-Length:319
v=0
o=tandberg 0 1 IN IP4 10.47.5.174
s=-
c=IN IP4 10.47.5.174
b=CT:768
t=0 0
m=audio 5600 RTP/AVP 9 8 0
c=IN IP4 10.47.5.174
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=video 5602 RTP/AVP 31
c=IN IP4 10.47.5.174
a=sendrecv
a=rtpmap:31 H261/90000
a=fmtp:31 cif=1;qcif=1
--- (12 headers 17 lines)---
Using INVITE request as basis request - 0100ad00ac00f900 at 10.47.5.174
Sending to 10.47.5.174 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.47.5.174:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.47.5.174:5060;branch=z9hG4bKa7dc32132336e031fee20e54a036538e.1;rport;received=10.47.5.174
From: "tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A
To: <sip:20170 at 10.47.8.89>;tag=as35edd3f0
Call-ID: 0100ad00ac00f900 at 10.47.5.174
CSeq: 100 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:20170 at 10.47.8.89>
Proxy-Authenticate: Digest realm="asterisk.tsip.lab", nonce="23e7f721"
Content-Length: 0
---
Scheduling destruction of call '0100ad00ac00f900 at 10.47.5.174' in 15000
ms
Found user 'trond'
asterisk1*CLI>
<-- SIP read from 10.47.5.174:5060:
ACK sip:20170 at 10.47.8.89 SIP/2.0
Via: SIP/2.0/UDP
10.47.5.174:5060;branch=z9hG4bKa7dc32132336e031fee20e54a036538e.1;rport
Call-ID: 0100ad00ac00f900 at 10.47.5.174
CSeq: 100 ACK
From: "tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A
To: <sip:20170 at 10.47.8.89>;tag=as35edd3f0
User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW))
Content-Length:0
--- (8 headers 0 lines)---
<-- SIP read from 10.47.5.174:5060:
INVITE sip:20170 at 10.47.8.89 SIP/2.0
Via: SIP/2.0/UDP
10.47.5.174:5060;branch=z9hG4bKdd4d38efc0ffcb9330aa4600b3a1dd8c.1;rport
Call-ID: 0100ad00ac00f900 at 10.47.5.174
CSeq: 101 INVITE
Contact: <sip:trond at 10.47.5.174>
From: "tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A
To: <sip:20170 at 10.47.8.89>
Max-Forwards: 70
Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW))
Proxy-Authorization: Digest nonce="23e7f721", realm="asterisk.tsip.lab",
qop="", username="trond", uri="sip:asterisk.tsip.lab",
response="7986e098509760d4aa60fc8d0b902172", algorithm=MD5
Content-Type:application/sdp
Content-Length:319
v=0
o=tandberg 0 1 IN IP4 10.47.5.174
s=-
c=IN IP4 10.47.5.174
b=CT:768
t=0 0
m=audio 5600 RTP/AVP 9 8 0
c=IN IP4 10.47.5.174
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
m=video 5602 RTP/AVP 31
c=IN IP4 10.47.5.174
a=sendrecv
a=rtpmap:31 H261/90000
a=fmtp:31 cif=1;qcif=1
--- (13 headers 17 lines)---
Using INVITE request as basis request - 0100ad00ac00f900 at 10.47.5.174
Sending to 10.47.5.174 : 5060 (non-NAT)
Found user 'trond'
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found video format unknown
Peer audio RTP is at port 10.47.5.174:5600
Peer video RTP is at port 10.47.5.174:5602
Found description format G722
Found description format PCMA
Found description format PCMU
Found description format H261
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|
g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0xc (ulaw|
alaw)/video=0x40000 (h261), combined - 0x4000c (ulaw|alaw|h261)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 20170 in from-internal (domain 10.47.8.89)
list_route: hop: <sip:trond at 10.47.5.174>
Transmitting (no NAT) to 10.47.5.174:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.47.5.174:5060;branch=z9hG4bKdd4d38efc0ffcb9330aa4600b3a1dd8c.1;rport;received=10.47.5.174
From: "tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A
To: <sip:20170 at 10.47.8.89>
Call-ID: 0100ad00ac00f900 at 10.47.5.174
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:20170 at 10.47.8.89>
Content-Length: 0
---
-- Executing Macro("SIP/trond-c6b8", "dial-direct-sip|
20170 at tandberg.net|20170") in new stack
-- Executing Macro("SIP/trond-c6b8", "test") in new stack
-- Executing Macro("SIP/trond-c6b8", "dumpvars") in new stack
-- Executing NoOp("SIP/trond-c6b8", "BLINDTRANSFER=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CALLERID="Trond G. Andersen"
<Trond>") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CALLERIDNAME=Trond G.
Andersen") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CALLERIDNUM=Trond") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "CALLINGPRES=0") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CHANNEL=SIP/trond-c6b8") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "CONTEXT=macro-dumpvars") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "DIALEDPEERNAME=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "DIALEDPEERNUMBER=") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "DIALSTATUS=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "EXTEN=s") in new stack
-- Executing NoOp("SIP/trond-c6b8", "SIPURI=sip:trond at 10.47.5.174")
in new stack
-- Executing NoOp("SIP/trond-c6b8", "SIPDOMAIN=10.47.8.89") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "SIP_CODEC=") in new stack
-- Executing NoOp("SIP/trond-c6b8",
"SIPCALLID=0100ad00ac00f900 at 10.47.5.174") in new stack
-- Executing NoOp("SIP/trond-c6b8", "SIPUSERAGENT=TANDBERG/46
(F3.2Beta10 PAL (TEST SW))") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CDR(src)=Trond") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CDR(dstchannel)=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "CDR(dcontext)=from-internal")
in new stack
-- Executing NoOp("SIP/trond-c6b8", "BLINDTRANSFER=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "BRIDGEPEER=") in new stack
-- Executing NoOp("SIP/trond-c6b8",
"SIPCHANINFO(recvip)=10.47.5.174") in new stack
-- Executing NoOp("SIP/trond-c6b8", "ARG1=20170 at tandberg.net") in
new stack
-- Executing NoOp("SIP/trond-c6b8", "ARG2=20170") in new stack
-- Executing NoOp("SIP/trond-c6b8", "ARG3=") in new stack
-- Executing NoOp("SIP/trond-c6b8", "SIPPEER-trond=10.47.5.174") in
new stack
-- Executing SIPGetHeader("SIP/trond-c6b8", "TMP=To") in new stack
-- Executing SIPGetHeader("SIP/trond-c6b8", "TMP2=From") in new
stack
-- Executing SIPGetHeader("SIP/trond-c6b8", "TMP3=Contact") in new
stack
-- Executing SIPGetHeader("SIP/trond-c6b8", "TMP4=Call-ID") in new
stack
-- Executing NoOp("SIP/trond-c6b8", "TMP=<sip:20170 at 10.47.8.89>") in
new stack
-- Executing NoOp("SIP/trond-c6b8", "TMP2="tga1000"
<sip:trond at asterisk.tsip.lab>;tag=4c003d001800db00;epid=TAA00506001E02A") in new stack
-- Executing NoOp("SIP/trond-c6b8", "TMP3=<sip:trond at 10.47.5.174>")
in new stack
-- Executing NoOp("SIP/trond-c6b8",
"TMP4=0100ad00ac00f900 at 10.47.5.174") in new stack
-- Executing SIPAddHeader("SIP/trond-c6b8", "Trond: <was here>") in
new stack
-- Executing NoOp("SIP/trond-c6b8", "SIPPEER-trond-codec0=0x1f07ff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|
h263|h263p)") in new stack
-- Executing Dial("SIP/trond-c6b8", "SIP/20170 at 10.47.8.89|30|Cf") in
new stack
We're at 10.47.8.89 port 9666
Video is at 10.47.8.89 port 9948
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x10000 (jpeg) to SDP
Adding codec 0x20000 (png) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
14 headers, 23 lines
Reliably Transmitting (no NAT) to 10.47.8.89:5060:
INVITE sip:20170 at 10.47.8.89 SIP/2.0
Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK75b38feb
From: "Trond G. Andersen" <sip:Trond at 10.47.8.89>;tag=as5feed5c6
To: <sip:20170 at 10.47.8.89>
Contact: <sip:Trond at 10.47.8.89>
Call-ID: 6f6b5c8f3f8edf7c3eb56fc937c33a8b at 10.47.8.89
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 17 Nov 2005 07:51:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Trond: <was here>
Content-Type: application/sdp
Content-Length: 547
v=0
o=root 10114 10114 IN IP4 10.47.8.89
s=session
c=IN IP4 10.47.8.89
t=0 0
m=audio 9666 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -
m=video 9948 RTP/AVP 26 31 34 103
a=rtpmap:26 JPEG/90000
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
---
-- Called 20170 at 10.47.8.89
<-- SIP read from 10.47.8.89:5060:
INVITE sip:20170 at 10.47.8.89 SIP/2.0
Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK75b38feb
From: "Trond G. Andersen" <sip:Trond at 10.47.8.89>;tag=as5feed5c6
To: <sip:20170 at 10.47.8.89>
Contact: <sip:Trond at 10.47.8.89>
Call-ID: 6f6b5c8f3f8edf7c3eb56fc937c33a8b at 10.47.8.89
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 17 Nov 2005 07:51:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Trond: <was here>
Content-Type: application/sdp
Content-Length: 547
v=0
o=root 10114 10114 IN IP4 10.47.8.89
s=session
c=IN IP4 10.47.8.89
t=0 0
m=audio 9666 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=silenceSupp:off - - - -
m=video 9948 RTP/AVP 26 31 34 103
a=rtpmap:26 JPEG/90000
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
--- (14 headers 23 lines)---
Transmitting (no NAT) to 10.47.8.89:5060:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP
10.47.8.89:5060;branch=z9hG4bK75b38feb;received=10.47.8.89
From: "Trond G. Andersen" <sip:Trond at 10.47.8.89>;tag=as5feed5c6
To: <sip:20170 at 10.47.8.89>;tag=as5feed5c6
Call-ID: 6f6b5c8f3f8edf7c3eb56fc937c33a8b at 10.47.8.89
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:Trond at 10.47.8.89>
Content-Length: 0
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