Re: [asterisk-users] Registrations problems to SIP-provider.
Actually it was my Firewall (Endian). By rebooting my firewall, all problems were solved and till this moment every communication succeeds. I do expect them back... I don't want to hijack this Asterisk-mailinglist, but I think that firewall-issues also are related to Asterisk-support. So my question : how could my Endian firewall cut off registration and thus disable VoIP-communications ?? Jonas. On Wed, 2009-07-01 at 22:11 +0200, hh174 wrote: Reliably Transmitting (no NAT) and you are natted I presume ( Port 5060 is forwarded to the internal IP-address of my Asterisk-server). Another Belgian user :) Olivier ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registrations problems to SIP-provider.
Just by blocking the packets coming from 3stars. Your asterisk, by receiving no responses think thet the host is not reachable. They are mayny good routers/firewall permitting to avoid these kind of problems. Olivier jonas kellens a écrit : Actually it was my Firewall (Endian). By rebooting my firewall, all problems were solved and till this moment every communication succeeds. I do expect them back... I don't want to hijack this Asterisk-mailinglist, but I think that firewall-issues also are related to Asterisk-support. So my question : how could my Endian firewall cut off registration and thus disable VoIP-communications ?? Jonas. On Wed, 2009-07-01 at 22:11 +0200, hh174 wrote: Reliably Transmitting (no NAT) and you are natted I presume ( Port 5060 is forwarded to the internal IP-address of my Asterisk-server). Another Belgian user :) Olivier ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registrations problems to SIP-provider.
How about putting nat=yes for everything even if it isn't NAT, use qualify, and get rid of your firewall rules if you are registering. On Thu, Jul 2, 2009 at 8:18 AM, hh174oliv...@hh174.be wrote: Just by blocking the packets coming from 3stars. Your asterisk, by receiving no responses think thet the host is not reachable. They are mayny good routers/firewall permitting to avoid these kind of problems. Olivier jonas kellens a écrit : Actually it was my Firewall (Endian). By rebooting my firewall, all problems were solved and till this moment every communication succeeds. I do expect them back... I don't want to hijack this Asterisk-mailinglist, but I think that firewall-issues also are related to Asterisk-support. So my question : how could my Endian firewall cut off registration and thus disable VoIP-communications ?? Jonas. On Wed, 2009-07-01 at 22:11 +0200, hh174 wrote: Reliably Transmitting (no NAT) and you are natted I presume ( Port 5060 is forwarded to the internal IP-address of my Asterisk-server). Another Belgian user :) Olivier ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registrations problems to SIP-provider.
jonas kellens wrote: Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register = 092779077:x...@85.119.188.3 mailto:df6...@85.119.188.3 This the output of SIP show peers : asterisk*CLI sip show peers Name/username HostDyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified)D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified)D 0 UNKNOWN grandstream/grandstream192.168.1.13 D 5060 OK (35 ms) 3starsnet/09277907785.119.188.3 N 5060 UNREACHABLE I'd say that your server is no longer able to access the SIP-provider. Confirm that you have network access first, then verify that you didn't make any changes to your configuration. If everything is good there, I would work with your provider to make sure the settings are correct. -- Dean Hoover Network Administrator Centurion, Inc. 262-317-5622 Phone dhoo...@centonline.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registrations problems to SIP-provider.
Reliably Transmitting (no NAT) and you are natted I presume ( Port 5060 is forwarded to the internal IP-address of my Asterisk-server). Another Belgian user :) Olivier jonas kellens a écrit : Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register = 092779077:x...@85.119.188.3 This the output of SIP show peers : asterisk*CLI sip show peers Name/username Host Dyn Nat ACL Port Status twinkle-candy/twinkle-can (Unspecified) D 0 UNKNOWN twinkle-jonas/twinkle-jon (Unspecified) D 0 UNKNOWN grandstream/grandstream 192.168.1.13 D 5060 OK (35 ms) 3starsnet/092779077 85.119.188.3 N 5060 UNREACHABLE This is the output of SIP debug : --- [Jul 1 21:08:37] NOTICE[15920]: chan_sip.c:7683 sip_reg_timeout: -- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #2) [Jul 1 21:08:37] REGISTER 12 headers, 0 lines [Jul 1 21:08:37] Reliably Transmitting (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:37] Really destroying SIP dialog '5d983c167b08b76b6211954c63c2a...@127.0.0.1' Method: REGISTER [Jul 1 21:08:38] Retransmitting #1 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Retransmitting #2 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:39] Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:40] Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #3 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK0fa45d19;rport From: sip:092779...@85.119.188.3;tag=as3306590c To: sip:092779...@85.119.188.3 Call-ID: 5d983c167b08b76b6211954c63c2a...@127.0.0.1 CSeq: 104 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@78.22.164.52 Event: registration Content-Length: 0 --- [Jul 1 21:08:41] Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 78.22.164.52:5060;branch=z9hG4bK671f78b3;rport From: "asterisk" sip:aster...@78.22.164.52;tag=as6e1c81b5 To: sip:85.119.188.3 Contact: sip:aster...@78.22.164.52 Call-ID: 070856d002110e3b6422a38747e54...@78.22.164.52 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Wed, 01 Jul 2009 19:08:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 1 21:08:42]
