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List Info
Thread: SIP Not Showing Disconnect
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| SIP Not Showing Disconnect |
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2007-07-22 14:20:38 |
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I've been having some problems recently with Asterisk thinking a SIP phone is still connected. These are GrandStream Budge Tone 102's and even after someone hangs up the AGI script I have running for them is still looping. The AGI script keeps playing "beep" untill it's told to continue by the user. ; But if the user has hung up the phone, the script just keeps looping and looping, and that causes a bunch of problems for me. Is there any way to do something on the line that would make Asterisk realize the phone isn't connected anymore? I would have thought something like playing a file would fix it all. Thought Asterisk would require a response from the SIP phone saying it got the message, and when the SIP phone didn't reply it would hang up the line.
=======================
& nbsp; -- AGI Script Executing Application: (PLAYBACK) Options: (beep)
-- <SIP/josh-08be5690> Playing 'beep' (language 'en9;)
-- AGI Script Executing Application: (PLAYBACK) Options: (beep)
-- <SIP/josh-08c391b8> Playing 'beep' (language 'en9;)
www*CLI> sip show peers
Name/username ; Host ; Dyn Nat ACL Port ; Status
josh/josh ; ;
66.205.135.100 D N 1024 ; UNREACHABLE
=======================
As you can see I've even unplugged the SIP phone and Asterisk is still keeping TWO (2) lines open to user "josh" even though Josh isn't reachable anymore. So even removing the phone from the network doesn't make Asterisk realize the phone isn't there anymore. Suggestions please!
--
/Nick
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| Re: SIP Not Showing Disconnect |
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2007-07-23 08:35:18 |
On Sunday 22 July 2007 21:20, Nicholas Blasgen wrote:
> I've been having some problems recently with Asterisk
thinking a SIP phone
> is still connected. These are GrandStream Budge Tone
102's and even after
> someone hangs up the AGI script I have running for them
is still looping.
FWIW, we set up bt102's as not needing registration i.e. we
set host=<actual
ip> instead of dynamic and configured the phone to allow
calls without
registration and all our problems went away. Before that,
the bt102's would
randomly lose connection with asterisk (V1.2 BTW)
Paul
--
Paul Hewlett Technical Director
Global Call Center Solutions Ltd, 2nd Floor, Milnerton Mall
Cnr Loxton & Koeberg Roads, 7435 Milnerton
www.gccs.co.za
Tel: +27 86 111 3433 Fax: +27 86 111 3520 Cel: +27 76 072
7906
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| Re: SIP Not Showing Disconnect |
 
United States |
2007-07-23 15:19:33 |
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At 12:20 PM -0700 2007/7/22, Nicholas Blasgen wrote:
I've been having some problems recently
with Asterisk thinking a SIP phone is still connected. These are
GrandStream Budge Tone 102's and even after someone hangs up the AGI
script I have running for them is still looping. The AGI script
keeps playing "beep" untill it's told to continue by the
user. But if the user has hung up the phone, the script just
keeps looping and looping, and that causes a bunch of problems for
me. Is there any way to do something on the line that would make
Asterisk realize the phone isn't connected anymore? I would have
thought something like playing a file would fix it all. Thought
Asterisk would require a response from the SIP phone saying it got the
message, and when the SIP phone didn't reply it would hang up the
line.
=======================
-- AGI Script
Executing Application: (PLAYBACK) Options: (beep)
-- <SIP/josh-08be5690> Playing 'beep'
(language 'en')
-- AGI Script Executing Application: (PLAYBACK)
Options: (beep)
-- <SIP/josh-08c391b8> Playing 'beep'
(language 'en')
www*CLI> sip show peers
Name/username ;
Host ;
Dyn Nat ACL Port ; Status
josh/josh ; ; 66.205.135.100 D
N ; 1024 ;
UNREACHABLE
=======================
As you can see I've even unplugged the
SIP phone and Asterisk is still keeping TWO (2) lines open to user
"josh"; even though Josh isn't reachable anymore. So
even removing the phone from the network doesn't make Asterisk realize
the phone isn't there anymore. Suggestions please!
--
/Nick
I'm uncertain if you've implemented the user-prompted method
correctly, since you do not include enough data to discern what is
happening with your system (and I would not suggest forwarding it to
this list, as it does not sound like your core question is a
development issue.)
There is an existing solution for this which does not require
prompts if you don't mind the media travelling through your Asterisk
system instead of directly between endpoints - look at the
descriptions for "rtptimeout"; in your sip.conf file. ;
That can be discussed at length in the Asterisk-Users mailing list if
further explanation is required.
The second solution is to put a maximum call duration on calls so
that Asterisk hangs up the calls regardless of what the state of the
endpoint is - see the description of the function TIMEOUT(absolute).
That can be discussed at length in the Asterisk-Users mailing list if
further explanation is required.
The last method (that I can think of, at least) would require
Session-Timers, which currently do not exist in Asterisk. That
has been discussed on this list (asterisk-dev) within the past six
days:
http://lists.digium.com/pipermail/asterisk-dev/2007-July/028574.html
If you are interested in creating a patch to support these
features, it would be greatly appreciated. However, if your
methods or questions regard any of the first three methods (prompts,
RTPtimeout, or absolute timeout) then I would suggest that
asterisk-users is a more appropriate forum for discussion.
JT
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