I've been asking about this problem in Asterisk channel... I didn't report it has a bug...Probably it is recommended... On 4/24/06, Thomas Winter
[EMAIL PROTECTED] wrote:Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta:
How do I report a Bug to Digium? or asterisk project?Did you report this
Am Wednesday 19 April 2006 16:37 schrieb Marco Mouta:
How do I report a Bug to Digium? or asterisk project?
Did you report this bug?
I checked and have seen only an timeout in the channel will kill the dead
channels.
Iam using GROUP_COUNT, so it is easy to kill my Asterisk if somebody is make
Marco Mouta wrote:
Hi all,
I've asterisk 1.2.5 , and what is happening is this:
Sip user agent A calls a pstn phone B
Sip User agent Activates MOH.
B starts listening.
A doesn't hangup and just Disconnect Sipoftphone XLITE (exit)
Sounds like an XLITE bug. On exit it should send some
Asterisk shouldn't see that the specific SIP user agent isn't there any more?On 4/19/06, Doug Lytle
[EMAIL PROTECTED] wrote:Marco Mouta wrote: Hi all, I've asterisk
1.2.5 , and what is happening is this: Sip user agent A calls a pstn phone B SipUser agent Activates MOH. B starts listening.
A
Marco Mouta wrote:
Asterisk shouldn't see that the specific SIP user agent isn't there
any more?
Eventually, yes. After the registration expires.
Doug
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I've tested maxexpirey=120 and even with this, asterisk didn't stop the call:Scenario: SIP user agent has left without telling to asterisk it was leaving...There was a call to pstn world with MOH running...
Any tip to solve this?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote:
Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualify
On Wed, 2006-04-19 at 14:51, Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call:
Scenario: SIP user agent has left without telling to asterisk it was
leaving...
There was a
qualify=yes may overload my network .. no?On 4/19/06, Gareth Blades [EMAIL PROTECTED]
wrote:Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualifyOn Wed, 2006-04-19 at 14:51, Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop the call:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call:
Scenario: SIP user agent has left without telling to asterisk it was
leaving...
There was a call to pstn world with MOH running...
Any tip to solve this?
None.
I just confirmed this:
Dial
Gareth Blades wrote:
Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualify
My phones are already set to qualify=500
Doug
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How do I report a Bug to Digium? or asterisk project?On 4/19/06, Doug Lytle [EMAIL PROTECTED] wrote:
Marco Mouta wrote: I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call: Scenario: SIP user agent has left without telling to asterisk it was leaving... There was a call to
Http://bugs.digium.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
MoutaSent: Wednesday, April 19, 2006 10:38 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Music on Hold bug? User disconnect Sip user
Did you try rtpholdtimeout in sip.conf ?
Hans
Marco Mouta schrieb:
How do I report a Bug to Digium? or asterisk project?
On 4/19/06, *Doug Lytle* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't
Johann Steinwendtner wrote:
Did you try rtpholdtimeout in sip.conf ?
Just tried it with rtpholdtimeout=60
did a reload from the console, and tried again.
Unplugging the phone and sitting on hold for 3 minutes. Never disconnected.
Just a reminder, I'm doing this over an IAX trunk to a SIP
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