[Asterisk-Users] NAT/Routing problem
Hi, all I have Asterisk here and SIP phone sitting at another location. Initially, I had problems registering the phone. Now I have added 'nat=yes' for this phone in sip.conf and phone registers. However, I can not make calls. SIP debug shows that phone registers with public IP address of the site, while calls somehow go to local address. Here is an example of SIP debug message: -- Registered SIP 'ext102' at 147.10.78.157 port 8103 expires 3600 -- Attempting native bridge of SIP/ext102-26a4 and SIP/ext101-1b49 Feb 27 17:02:31 WARNING[3160]: chan_sip.c:755 retrans_pkt: Maximum retries excee ded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Res ponse) As one can see, public IP 147.10.78.157 is used at registration time, while private IP 192.168.1.2 is used for communicating with phone. Remote site does not have firewall. My site does, but I could not see anything wrong there. I have turned on logging on firewall and no suspicios activity goes on. Any help is appreciated. Thanks, Rudolf P.S. Here is extract from my sip.conf file: [ext102] type=user nat=yes host=dynamic secret=ext102 context=default [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 ___ 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
Re: [Asterisk-Users] NAT/Routing problem
On 19:45, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote: As one can see, public IP 147.10.78.157 is used at registration time, while private IP 192.168.1.2 is used for communicating with phone. [ext102] type=user nat=yes host=dynamic secret=ext102 context=default [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 try adding: canreinvite=no -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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
Re: [Asterisk-Users] NAT/Routing problem
Thanks for suggestion. Unfortunately did not work. What does this option do anyway? Rudolf - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, February 27, 2005 8:18 PM Subject: Re: [Asterisk-Users] NAT/Routing problem On 19:45, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote: As one can see, public IP 147.10.78.157 is used at registration time, while private IP 192.168.1.2 is used for communicating with phone. [ext102] type=user nat=yes host=dynamic secret=ext102 context=default [ext102] type=peer nat=yes secret=ext102 host=dynamic context=default callerid=Ext 102 try adding: canreinvite=no -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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 ___ 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
Re: [Asterisk-Users] NAT/Routing problem
On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote: Thanks for suggestion. Unfortunately did not work. What does this option do anyway? I cannot explain it as clear as the wiki. have a look here: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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
Re: [Asterisk-Users] NAT/Routing problem
Hmm, is your asterisk server behind nat with port forwarded ports? If so, have you tried adding this to your sip.conf? [general] externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP localnet=192.168.0.0/255.255.255.0 ; your LAN ... ... [ext102] canreinvite=no host=dynamic nat=yes . . Julian J. M. On Sun, 27 Feb 2005 11:04:25 +0100, Michiel van Baak [EMAIL PROTECTED] wrote: On 20:52, Sun 27 Feb 05, Rudolf Ladyzhenskii wrote: Thanks for suggestion. Unfortunately did not work. What does this option do anyway? I cannot explain it as clear as the wiki. have a look here: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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 ___ 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
Re: [Asterisk-Users] NAT/Routing problem
Is your asterisk server behind nat with port forwarded ports? If so, have you tried adding this to your sip.conf? [general] externip=xxx.xxx.xxx.xxx; your external IP, provided by your ISP localnet=192.168.0.0/255.255.255.0 ; your LAN ... ... [ext102] canreinvite=no host=dynamic nat=yes . . Julian J. M. SIP debug shows that phone registers with public IP address of the site, while calls somehow go to local address. ___ 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
