[asterisk-bugs] [Asterisk 0018172]: Peer goes unreachable when placing a call from it.
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Oct 22 08:02:11 CDT 2010
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=18172
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Reported By: jordankirby
Assigned To:
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Project: Asterisk
Issue ID: 18172
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.8.0-rc5
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-10-20 06:23 CDT
Last Modified: 2010-10-22 08:02 CDT
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Summary: Peer goes unreachable when placing a call from it.
Description:
We are using 1.8RC5, Asterisk RealTime (MySQL) and realtime caching.
Both the server and the phones are behind separate NAT (phone in office,
server in data centre).
In this example, SIP/3001 calls SIP/3002 (via a gosub). The call proceeds
fine but asterisk sees SIP/3001 as unavailable for a period of time after
the initial invite.
This looks to be because asterisk starts sending OPTIONS and NOTIFIES to
the LAN IP address of the phone rather than it's NATed IP address.
In the attached trace:
10.50.0.47 is the server's LAN IP.
212.62.4.230 is the server's NATed IP.
192.168.1.231 is 3001's LAN IP.
195.59.152.66 is 3001's NATed IP.
Asterisk output:
== Using SIP RTP CoS mark 5
-- Executing [3002 at DLPN_All:1] Gosub("SIP/3001-000000aa",
"internal,3002,1")
-- Executing [3002 at internal:1] Dial("SIP/3001-000000aa", "SIP/3002")
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Called 3002
-- SIP/3002-000000ab is ringing
[Oct 20 12:11:05] NOTICE[24099]: chan_sip.c:24694 sip_poke_noanswer: Peer
'3001' is now UNREACHABLE! Last qualify: 37
== Spawn extension (internal, 3002, 1) exited non-zero on
'SIP/3001-000000aa'
-- Registered SIP '3001' at 195.59.152.66:17261
[Oct 20 12:11:55] NOTICE[24099]: chan_sip.c:19523
handle_response_peerpoke: Peer '3001' is now Reachable. (34ms / 2000ms)
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(0128312) schmidts (manager) - 2010-10-22 08:02
https://issues.asterisk.org/view.php?id=18172#c128312
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after talking to pabelanger we reopen this issue but we need more
information from you, like your sip.conf, realtime conf and also if
possible a mysql log what happens during this call.
and btw your nat really make some problems here too. i have seen in your
pcap file you have a cisco device in your network. If this is a cisco
router do you have disabled sip nat service (sip alg)?
if not just set the following parameters in your cisco router:
conf t
no nat service sip udp port 5060
write memory
thanks
schmidts
Issue History
Date Modified Username Field Change
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2010-10-22 08:02 schmidts Note Added: 0128312
2010-10-22 08:02 schmidts Status closed => feedback
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