Re: [asterisk-users] External sip phones register with the servers IP...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 8/1/13 9:17 PM, Michael L. Young wrote: - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013 8:41:19 PM Subject: [asterisk-users] External sip phones register with the servers IP... We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up having something like this: Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003' at 192.168.2.50:58545 Reliably Transmitting (no NAT) to 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172 To: sip:2003@192.168.2.50:58545;ob Contact: sip:asterisk@192.168.2.50:5060 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. I have tried with all nat variations and I get the same result. I upgraded to 11.5 yesterday but same problem. External phones still register using the servers internal IP. IAX works fine. - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlH73YUACgkQqmNh+MyHzx7vwACfeh+97zTm4RnFqpUTfQfZupZA 6YEAoKL3k5au2/I+VY3Sdp/Uyq0Ly7uU =j+t/ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External sip phones register with the servers IP...
Please post one of your sip.conf phone configs, so we can have a look. Alyed 2013/8/2 Carlos Chavez cur...@telecomabmex.com -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 8/1/13 9:17 PM, Michael L. Young wrote: - Original Message - From: Carlos Chavez cur...@telecomabmex.com To: asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013 8:41:19 PM Subject: [asterisk-users] External sip phones register with the servers IP... We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up having something like this: Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003' at 192.168.2.50:58545 Reliably Transmitting (no NAT) to 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172 To: sip:2003@192.168.2.50:58545;ob Contact: sip:asterisk@192.168.2.50:5060 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. I have tried with all nat variations and I get the same result. I upgraded to 11.5 yesterday but same problem. External phones still register using the servers internal IP. IAX works fine. - -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.18 (Darwin) Comment: GPGTools - http://gpgtools.org Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/ iEYEARECAAYFAlH73YUACgkQqmNh+MyHzx7vwACfeh+97zTm4RnFqpUTfQfZupZA 6YEAoKL3k5au2/I+VY3Sdp/Uyq0Ly7uU =j+t/ -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] External sip phones register with the servers IP...
- Original Message - From: Carlos Chavez cur...@telecomabmex.com To: asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013 8:41:19 PM Subject: [asterisk-users] External sip phones register with the servers IP... We have just updated our office server to Asterisk 11.4.0 from 1.8.15 and internally everything is working fine. The problem we are having is that we cannot use any external phone connected through the Internet. This used to work fine with 1.8 but since the upgrade whenever you register any phone from an outside network the phone tries to register using the servers internal IP. I endo up having something like this: Sending to 187.163.93.235:58545 (no NAT) -- Registered SIP '2003' at 192.168.2.50:58545 Reliably Transmitting (no NAT) to 192.168.2.50:58545: OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 Max-Forwards: 70 From: asterisk sip:asterisk@192.168.2.50;tag=as4ed13172 To: sip:2003@192.168.2.50:58545;ob Contact: sip:asterisk@192.168.2.50:5060 Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.4.0 Date: Fri, 02 Aug 2013 00:27:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 I really cannot understand what is wrong, I have checked my sip.conf configuration and it is the same as in past versions. externaddr and localnet are set to the proper values. Any ideas? Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. -- Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
