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rentorbuy at yahoo.com
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 Posted: Tue Mar 25, 2008 5:32 am Post subject: [asterisk-users] force soft hangup |
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How can I "force" soft hangup (if that makes sense)?
"show channels" reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to have
worked but it doesn't for the analog/ATA phone. "show
hints" also shows that it's InUse. But of course it
isn't and noone can contact this extension since I
disabled call waiting for it (I also rebooted the ATA
and forced re-registration of the ATA SIP client).
So how can I "kill" this channel without restarting
the asterisk daemon?
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gordon+asterisk at dro...
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| Posted: Tue Mar 25, 2008 6:18 am Post subject: [asterisk-users] force soft hangup |
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On Tue, 25 Mar 2008, Vieri wrote:
| Quote: | How can I "force" soft hangup (if that makes sense)?
"show channels" reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to have
worked but it doesn't for the analog/ATA phone. "show
hints" also shows that it's InUse. But of course it
isn't and noone can contact this extension since I
disabled call waiting for it (I also rebooted the ATA
and forced re-registration of the ATA SIP client).
So how can I "kill" this channel without restarting
the asterisk daemon?
|
Strangely enough the command in the CLI is:
soft hangup <channel>
Just type soft hangup then push the TAB key and it will auto complete for
you ...
Gordon |
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davies147 at gmail.com
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| Posted: Tue Mar 25, 2008 7:23 am Post subject: [asterisk-users] force soft hangup |
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On 25/03/2008, Gordon Henderson <gordon+asterisk at drogon.net> wrote:
| Quote: | On Tue, 25 Mar 2008, Vieri wrote:
| Quote: | How can I "force" soft hangup (if that makes sense)?
"show channels" reveals a stale sip channel. It's of
an analog phone behind a Grandstream ATA which was
communicating with another SIP softphone. The latter
crashed. A soft hangup of the softphone seems to have
worked but it doesn't for the analog/ATA phone. "show
hints" also shows that it's InUse. But of course it
isn't and noone can contact this extension since I
disabled call waiting for it (I also rebooted the ATA
and forced re-registration of the ATA SIP client).
So how can I "kill" this channel without restarting
the asterisk daemon?
|
Strangely enough the command in the CLI is:
soft hangup <channel>
Just type soft hangup then push the TAB key and it will auto complete for
you ...
|
I think the OP's point is that "soft hangup ..." does not hang up one
of the channels successfully. I have seen this before, usually from
soft-phones that do not disconnect cleanly, or WiFi phones that lose
signal during a call.
Using rtptimeout and rtpholdtimeout settings in sip.conf seems to
improve matters, as does using more recent versions of asterisk - You
do not say what version you are running.
Regards,
Steve |
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rentorbuy at yahoo.com
Guest
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| Posted: Tue Mar 25, 2008 9:52 am Post subject: [asterisk-users] force soft hangup |
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--- Steve Davies <davies147 at gmail.com> wrote:
| Quote: | On 25/03/2008, Gordon Henderson
<gordon+asterisk at drogon.net> wrote:
| Quote: | On Tue, 25 Mar 2008, Vieri wrote:
| Quote: | How can I "force" soft hangup (if that makes
|
|
sense)?
| Quote: | | Quote: |
"show channels" reveals a stale sip channel.
|
|
It's of
| Quote: | | Quote: | an analog phone behind a Grandstream ATA which
|
|
was
| Quote: | | Quote: | communicating with another SIP softphone. The
|
|
latter
| Quote: | | Quote: | crashed. A soft hangup of the softphone seems
|
|
to have
| Quote: | | Quote: | worked but it doesn't for the analog/ATA phone.
|
|
"show
| Quote: | | Quote: | hints" also shows that it's InUse. But of
|
|
course it
| Quote: | | Quote: | isn't and noone can contact this extension
|
|
since I
| Quote: | | Quote: | disabled call waiting for it (I also rebooted
|
|
the ATA
| Quote: | | Quote: | and forced re-registration of the ATA SIP
|
|
client).
| Quote: | | Quote: |
So how can I "kill" this channel without
|
|
restarting
| Quote: | | Quote: | the asterisk daemon?
|
Strangely enough the command in the CLI is:
soft hangup <channel>
Just type soft hangup then push the TAB key and
|
it will auto complete for
I think the OP's point is that "soft hangup ..."
does not hang up one
of the channels successfully. I have seen this
before, usually from
soft-phones that do not disconnect cleanly, or WiFi
phones that lose
signal during a call.
Using rtptimeout and rtpholdtimeout settings in
sip.conf seems to
improve matters, as does using more recent versions
of asterisk - You
do not say what version you are running.
|
Thanks Steve,
I'm using 1.2.26.2 and upgrading to 1.2.27 today.
rtp*timeout sounds like it could do the job (just hope
it won't do false hangups) and I think that these
settings are available for 1.2.
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rentorbuy at yahoo.com
Guest
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| Posted: Tue Mar 25, 2008 10:17 am Post subject: [asterisk-users] force soft hangup |
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--- Steve Davies <davies147 at gmail.com> wrote:
| Quote: | Using rtptimeout and rtpholdtimeout settings in
sip.conf
|
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI> sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0 s at macro-dial:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
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tilghman at mail.jeffa...
Guest
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| Posted: Tue Mar 25, 2008 10:48 am Post subject: [asterisk-users] force soft hangup |
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On Tuesday 25 March 2008 10:17:54 Vieri wrote:
| Quote: | --- Steve Davies <davies147 at gmail.com> wrote:
| Quote: | Using rtptimeout and rtpholdtimeout settings in
sip.conf
|
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI> sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0 s at macro-dial:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
|
Given that you didn't give Dial a timeout, yes, it will try
forever, until it receives a response. Note that this has
nothing to do with rtptimeout, as that takes effect when
the call is established, and the RTP packets stop flowing.
Without using a firewall rule between the two hosts, it is
somewhat difficult to mock up that situation (as the RTP
is still flowing, even if the audio is silent).
--
Tilghman |
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davies147 at gmail.com
Guest
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| Posted: Tue Mar 25, 2008 11:12 am Post subject: [asterisk-users] force soft hangup |
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On 25/03/2008, Tilghman Lesher <tilghman at mail.jeffandtilghman.com> wrote:
| Quote: | On Tuesday 25 March 2008 10:17:54 Vieri wrote:
| Quote: | --- Steve Davies <davies147 at gmail.com> wrote:
| Quote: | Using rtptimeout and rtpholdtimeout settings in
sip.conf
|
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI> sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0 s at macro-dial:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
|
Given that you didn't give Dial a timeout, yes, it will try
forever, until it receives a response. Note that this has
nothing to do with rtptimeout, as that takes effect when
the call is established, and the RTP packets stop flowing.
Without using a firewall rule between the two hosts, it is
somewhat difficult to mock up that situation (as the RTP
is still flowing, even if the audio is silent).
|
I would have assumed that "Up" indicates that it thinks the call is
already in-progress, and that RTP should be flowing.
You will not be able to change the rtptimeout values of an established
call as AFAIK these settings are copied into the channel object when
it is created, and not updated if SIP is reloaded.
So... I think that channel is stuck until a restart, but you may be
able to reduce its occurrence in future by using rtptimeout, and
possibly also by using Dial timeouts as Tilghman pointed out.
I use rtptimeout as standard these days, and have not seen a channel
stuck-hard for quite a while (Nor have I seen false hang-ups).
Regards,
Steve |
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rentorbuy at yahoo.com
Guest
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| Posted: Tue Mar 25, 2008 11:21 am Post subject: [asterisk-users] force soft hangup |
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--- Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:
| Quote: | On Tuesday 25 March 2008 10:17:54 Vieri wrote:
| Quote: | --- Steve Davies <davies147 at gmail.com> wrote:
| Quote: | Using rtptimeout and rtpholdtimeout settings in
sip.conf
|
I set
rtptimeout=10
rtpholdtimeout=30
(just for testing; I know these values are way too
low)
then did a
CLI> sip reload
and waited more than 30 seconds.
The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0 s at macro-dial:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an
|
infinite
| Quote: | loop trying to dial/reach SIP/4053?
|
Given that you didn't give Dial a timeout, yes, it
will try
forever, until it receives a response.
|
Must put timeouts on all Dials then. Just in case.
Thanks!
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rentorbuy at yahoo.com
Guest
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| Posted: Tue Mar 25, 2008 12:58 pm Post subject: [asterisk-users] force soft hangup |
|
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--- Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:
| Quote: | | Quote: | The SIP channel is still there (InUse).
Channel Location State
Application(Data)
SIP/6010-b38d53e0 s at macro-dial:8 Up
Dial(SIP/4053||tTwW)
Should I interpret the above that it's in an
|
infinite
| Quote: | loop trying to dial/reach SIP/4053?
|
Given that you didn't give Dial a timeout, yes, it
will try
forever, until it receives a response.
|
I've just run into another similar case.
This time a softphone (4012) which isn't even
registered anymore is still locking up a voice channel
through SIP/205 which is an FXO gateway connected to
PSTN. If I chanspy then I hear nothing/ total silence
(no tone plays).
Shouldn't rtptimeout do its job here and disconnect?
How can I make sure there really is no RTP flow?
Channel Location State
Application(Data)
SIP/205-0a778a58 (None) Up
Bridged Call(SIP/4012-b3291db8
SIP/4012-b3291db8 s at macro-dialout-trun Up
Dial(SIP/205/0666558844|300|TW
The FXO Grandstream gateway does not have Silence
Suppression enabled.
Thanks for any advice.
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benny+usenet at amorse...
Guest
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| Posted: Tue Mar 25, 2008 3:21 pm Post subject: [asterisk-users] force soft hangup |
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Vieri <rentorbuy at yahoo.com> writes:
| Quote: | Should I interpret the above that it's in an infinite
loop trying to dial/reach SIP/4053?
|
They are just stuck channels. It's simply a bug in 1.2.x. Fortunately
it's fixed in 1.4.x. We upgrade our customers to 1.4.x when they hit
that bug.
/Benny |
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