[Asterisk-Users] Call terminated after 60 seconds

[Eric "ManxPower" Wieling]

For one thing, don't use the "r" option to dial.  It can hide major 
problems.  If you don't hear ringing without using "r" then you have 
massive problems.

Asterisk wrote:
> Nope,
> 
> It's not a firewall problem.
> I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.
> It replaces the inside IP adresses from the * server in the SIP frames 
> by the outside IP adress and creates pinholes for the udp streams.
> 
> I have several SIP connections (SIPphone, SIPGate, IPtel, Bugetphone 
> ...) and only this one is problematic. 
> 
> Andre
> 
> 
>     ----- Oorspronkelijk Bericht -----
>     *Onderwerp: *Re: [Asterisk-Users] Call terminated after 60 seconds
>     *Afzender:  *Franc esco Peeters (Asterisk) <francesco at fampeeters.com>
>     *Aan: *"Asterisk" <asterisk at vinkconsult.com>,"Asterisk Users Mailing
>     List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
>     *CC: *"asterisk-users at lists.digium.com."
>     <asterisk-users at lists.digium.com>
>     *Datum: *24-03-2006 12:18
> 
> 
>     On Fri, March 24, 2006 12:01, Asterisk said:
>      >
>      >
>      > Hello,
>      >
>      > I switched from my PSTN provider to a voip provider. (Voicedata in
>      > the Netherlands)
>      >>From the moment i switched all inbound calls are terminated after
>      > aproximatly 1 minute.
>      > The provider tells me it's not their issue since I have no other
>      > configuration than all their other users.
>      >
>      > What can I do.
>      >
>      > I removed all asterisk functionality by forwarding the inboud ca ll
>      > directly to a local phone
>      > ; Inbound voicedata context
>      > ;
>      > [from-voicedata]
>      > exten => ${VOICEDATACIDNUM},1,NoOp(From Voicedata)
>      > exten => ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr)
>      > ; end of context
>      > Regards,
>      >
>      > Andre Vink
>      >
> 
>     Check whether your firewall has a fixed UDP timeout set at 60 seconds...
>     That solved my problem... ;-)
>     (Together with activating SIP/VoIP support)
> 
>     -- 
>     F Peeters
>     PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
>     2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
>     AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
>     2 Sweex HFC-PCI cards
> 
> 
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> 
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