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) <[EMAIL PROTECTED]>
*Aan: *"Asterisk" <[EMAIL PROTECTED]>,"Asterisk Users Mailing
List - Non-Commercial Discussion" <[email protected]>
*CC: *"[EMAIL PROTECTED]"
<[email protected]>
*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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