I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk server and STUN server are outside the firewall on a public network. I would like the Sipuras to be able to reinvite, so I set canreinvite=yes in my sip.conf, and set the STUN server under the SIP tab in the Sipuras. However, I am not able to hear the other caller (the Sipura is not recieving RTP packets, it is sending just fine). Am I missing something on the Sipura config? I am not sure what all of the VIA options mean, and which ones I should use. Cant find any good info out there, can someone hrer help me out? Thank you.
AJ Grinnell wrote:>I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk >server and STUN server are outside the firewall on a public network. I would >like the Sipuras to be able to reinvite, so I set canreinvite=yes in my >sip.conf, and set the STUN server under the SIP tab in the Sipuras. However, >I am not able to hear the other caller (the Sipura is not recieving RTP >packets, it is sending just fine). Am I missing something on the Sipura >config? I am not sure what all of the VIA options mean, and which ones I >should use. Cant find any good info out there, can someone hrer help me out? >Thank you. > > > > >You need these settings: Substitute_VIA_Addr "Yes" ; STUN_Enable "Yes" ; NAT_Mapping_Enable[1] "Yes" ; NAT_Keep_Alive_Enable[1] "Yes" ; STUN_Test_Enable "Yes"; and of course define your STUN Server. -- Andres Network Admin http://www.telesip.net
That works perfect, man, I owe you a beer! -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Andres Sent: Wednesday, May 26, 2004 5:28 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sipura stun settings AJ Grinnell wrote:>I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk >server and STUN server are outside the firewall on a public network. Iwould>like the Sipuras to be able to reinvite, so I set canreinvite=yes in my >sip.conf, and set the STUN server under the SIP tab in the Sipuras.However,>I am not able to hear the other caller (the Sipura is not recieving RTP >packets, it is sending just fine). Am I missing something on the Sipura >config? I am not sure what all of the VIA options mean, and which ones I >should use. Cant find any good info out there, can someone hrer help meout?>Thank you. > > > > >You need these settings: Substitute_VIA_Addr "Yes" ; STUN_Enable "Yes" ; NAT_Mapping_Enable[1] "Yes" ; NAT_Keep_Alive_Enable[1] "Yes" ; STUN_Test_Enable "Yes"; and of course define your STUN Server. -- Andres Network Admin http://www.telesip.net _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned by Arialink for dangerous content and is believed to be clean. For more information please email support@arialink.com
Well, it worked for 1 call, but now I am back to getting half a ring from the ATA and then nothing. I am only seeing one rtp packet recieved per call. Any other ideas? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Andres Sent: Wednesday, May 26, 2004 5:28 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Sipura stun settings AJ Grinnell wrote:>I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk >server and STUN server are outside the firewall on a public network. Iwould>like the Sipuras to be able to reinvite, so I set canreinvite=yes in my >sip.conf, and set the STUN server under the SIP tab in the Sipuras.However,>I am not able to hear the other caller (the Sipura is not recieving RTP >packets, it is sending just fine). Am I missing something on the Sipura >config? I am not sure what all of the VIA options mean, and which ones I >should use. Cant find any good info out there, can someone hrer help meout?>Thank you. > > > > >You need these settings: Substitute_VIA_Addr "Yes" ; STUN_Enable "Yes" ; NAT_Mapping_Enable[1] "Yes" ; NAT_Keep_Alive_Enable[1] "Yes" ; STUN_Test_Enable "Yes"; and of course define your STUN Server. -- Andres Network Admin http://www.telesip.net _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned by Arialink for dangerous content and is believed to be clean. For more information please email support@arialink.com
AJ Grinnell wrote:>Well, it worked for 1 call, but now I am back to getting half a ring from >the ATA and then nothing. I am only seeing one rtp packet recieved per call. >Any other ideas? > > >Does it work ok if canreinvite=no ?? To be honest I have not experimented much with canreinvite. All our Sipura subs register with SER and not Asterisk. STUN works fine for us with the settings I sent you.>-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Andres >Sent: Wednesday, May 26, 2004 5:28 PM >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] Sipura stun settings > > >AJ Grinnell wrote: > > > >>I am using sipura spa-1000s and spa-2000s behind a firewall. My asterisk >>server and STUN server are outside the firewall on a public network. I >> >> >would > > >>like the Sipuras to be able to reinvite, so I set canreinvite=yes in my >>sip.conf, and set the STUN server under the SIP tab in the Sipuras. >> >> >However, > > >>I am not able to hear the other caller (the Sipura is not recieving RTP >>packets, it is sending just fine). Am I missing something on the Sipura >>config? I am not sure what all of the VIA options mean, and which ones I >>should use. Cant find any good info out there, can someone hrer help me >> >> >out? > > >>Thank you. >> >> >> >> >> >> >> >You need these settings: >Substitute_VIA_Addr "Yes" ; >STUN_Enable "Yes" ; >NAT_Mapping_Enable[1] "Yes" ; >NAT_Keep_Alive_Enable[1] "Yes" ; >STUN_Test_Enable "Yes"; > >and of course define your STUN Server. > >-- >Andres >Network Admin >http://www.telesip.net > > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- > >This message has been scanned by Arialink for dangerous content and is >believed to be clean. For more information please email support@arialink.com > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- Andres Network Admin http://www.telesip.net "Providing Wholesale Florida SIP/IAX2 Termination for US$0.01/minute"