<html><head><meta http-equiv=Content-Type content="text/html;charset=UTF-8"><meta content="Group-Office 2.15" name="GENERATOR"></head><body>Nope,<br /><br />It's not a firewall problem.<br />I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.<br />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.<br /><br />I have several SIP connections (SIPphone, SIPGate, IPtel, Bugetphone ...) and only this one is problematic. <br /><br />Andre<br /><br /><br /><blockquote style="BORDER-TOP-WIDTH: 0px; PADDING-RIGHT: 0px; PADDING-LEFT: 5px; BORDER-BOTTOM-WIDTH: 0px; PADDING-BOTTOM: 0px; MARGIN: 0px 0px 0px 5px; BORDER-LEFT: #22437f 2px solid; PADDING-TOP: 0px; BORDER-RIGHT-WIDTH: 0px"><font face="verdana" size="2">----- Oorspronkelijk Bericht -----<br /><strong>Onderwerp: </strong>Re: [Asterisk-Users] Call terminated after 60 seconds<br /><strong>Afzender: </strong>Francesco
Peeters (Asterisk) <francesco@fampeeters.com><br /><strong>Aan: </strong>"Asterisk" <asterisk@vinkconsult.com>,"Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br /><strong>CC: </strong>"asterisk-users@lists.digium.com." <asterisk-users@lists.digium.com><br /><strong>Datum: </strong>24-03-2006 12:18<br /></font><br /><br />On Fri, March 24, 2006 12:01, Asterisk said:<br />><br />><br />> Hello,<br />><br />> I switched from my PSTN provider to a voip provider. (Voicedata in<br />> the Netherlands)<br />>>From the moment i switched all inbound calls are terminated after<br />> aproximatly 1 minute.<br />> The provider tells me it's not their issue since I have no other<br />> configuration than all their other users.<br />><br />> What can I do.<br />><br />> I removed all asterisk functionality by forwarding the inboud call<br
/>> directly to a local phone<br />> ; Inbound voicedata context<br />> ;<br />> [from-voicedata]<br />> exten => ${VOICEDATACIDNUM},1,NoOp(From Voicedata)<br />> exten => ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr)<br />> ; end of context<br />> Regards,<br />><br />> Andre Vink<br />><br /><br />Check whether your firewall has a fixed UDP timeout set at 60 seconds...<br />That solved my problem... ;-)<br />(Together with activating SIP/VoIP support)<br /><br />-- <br />F Peeters<br />PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch<br />2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0<br />AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN<br />2 Sweex HFC-PCI cards<br /><br /></blockquote></body></html>