[Asterisk-Dev] Re: 482 Loop Detected problem

[Nishi Kant]

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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