[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)

[Nestor A. Diaz]

ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp 
traffic is not passing thought asterisk, or i have to put canreinvite=no ?

slds.
> rtp*timeout for sip peers is not a fix but a
> workaround.
> Try to set both values and reload sip.
> Then when you witness what you posted try doing a
> "core show channels". You can then try to "soft
> hangup" a stuck channel or wait for the rtp*timeouts.
>
>
>
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-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia