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 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 from an iax trunk to sip phone
SIP phone places call on hold.
Unplugged SIP phone
Music on hold did not disconnect even after the expiry.
Doing a sip show [exten]
Shows the peer is unreachable.
You may want to file a bug report.
Doug
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