[Asterisk-Dev] Re: 482 Loop Detected problem

[Nishi Kant]

Hi All.
Here is console logs for same problem

--------------------Logs------------------------
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.60.121:5050;branch=z9hG4bK11eeab88
From: "asterisk" <sip:asterisk at 192.168.60.121:5050>;tag=as4078a524
To: <sip:501 at 192.168.60.120>;tag=as4078a524
Contact: <sip:asterisk at 192.168.60.121:5050>
Content-Type: application/sdp
v=0
o=root 23293 23294 IN IP4 192.168.60.121
s=session
c=IN IP4 192.168.60.121
t=0 0
m=audio 11376 RTP/AVP 0 101

  rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


12 headers, 10 lines
    -- SIP/fwd-bc20 answered Zap/1-1
Dec  7 12:20:16 WARNING[23293]: res_features.c:382 ast_bridge_call: Bridge
faile
d on channels SIP/fwd-bc20 and Zap/2-1
    -- Hungup 'Zap/2-1'
  == Spawn extension (default, s, 1) exited non-zero on 'SIP/fwd-bc20'
Dec  7 12:20:16 WARNING[23293]: channel.c:732 ast_hangup: Hard hangup called
by
thread -1259086928 on SIP/fwd-bc20, while fd is blocked by
thread -1256866896 in
 procedure ast_waitfor_nandfds!  Expect a failure

------------------Logs-------------------------------

Thanks,
Nishi
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of Nishi Kant
Sent: Wednesday, December 07, 2005 10:41 AM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Re: 482 Loop Detected problem


Hi All,

I am also facing same "482 Loop Detected" issue. We are using Asterisks as
PSTN gateway and SIPX as sip PBX. Calls made from PSTN to SIP or from SIP to
PSTN works fine. When a call is made from PSTN to SIP UA and SIP UA forward
this call to another PSTN no, Asterisk gives "482 Loop detected"  and call
is dropped.

My observation is that Asterisk is confusing spiral for a Loop as request
URI of forwarded request is different from original request URI. To overcome
this problem I have commented if statement in chan_sip.c. After modifying
the code and remove loop detection logic found that asterisk is forwarding
call to Sip Proxy again instead of PSTN. To overcome this I have to modify
the extension.conf. After making all these changed I am able to ring called
PSTN no, but as soon as called party picked up the phone Asterisk crashes.
Any help on this issue will be highly appreciated.

Thanks,
Nishi

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of snacktime
Sent: Tuesday, December 06, 2005 1:26 PM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Re: 482 Loop Detected problem


One more thing.  With pedantic=yes in sip.conf the same call results
in no audio.  I'm using externip and localnet in sip.conf so maybe
pedantic has some effect on those settings?  I have a debug of the
same call with pedantic=yes if someone wants me to post that as well.

Chris
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