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 afteraproximatly 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 call 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060324/4b7ca379/attachment.htm
Francesco Peeters (Asterisk)
2006-Mar-24 04:18 UTC
[Asterisk-Users] Call terminated after 60 seconds
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 call > 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
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: Francesco Peeters (Asterisk) Aan: "Asterisk" ,"Asterisk Users Mailing List - Non-Commercial Discussion" CC: "asterisk-users@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 call > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060324/9790ac77/attachment-0001.htm
Eric "ManxPower" Wieling
2006-Mar-24 11:03 UTC
[Asterisk-Users] Call terminated after 60 seconds
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@fampeeters.com> > *Aan: *"Asterisk" <asterisk@vinkconsult.com>,"Asterisk Users Mailing > List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> > *CC: *"asterisk-users@lists.digium.com." > <asterisk-users@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 > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users