Re: [asterisk-users] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
On 6/13/13 16:20 , Matthew J. Roth wrote: It's hard to be certain without seeing a full SIP trace, but I think the INVITE with the internal IP is actually a re-INVITE that Asterisk is sending to establish a native bridge between the SIP friend and the SIP gateway to PSTN converter. It's actually pretty easy. If an INVITE message has a tag parameter in both To and From headers, it's a re-INVITE. If the To header doesn't have a tag parameter, it's an initial INVITE. -- Andreas Sikkema -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Andreas Sikkema wrote: On 6/13/13 16:20 , Matthew J. Roth wrote: It's hard to be certain without seeing a full SIP trace, but I think the INVITE with the internal IP is actually a re-INVITE that Asterisk is sending to establish a native bridge between the SIP friend and the SIP gateway to PSTN converter. It's actually pretty easy. If an INVITE message has a tag parameter in both To and From headers, it's a re-INVITE. If the To header doesn't have a tag parameter, it's an initial INVITE. Andreas, Thanks for the tip. That's a very useful bit of information to know. It also confirms that the INVITE in Mickael's original post [1] is a re-INVITE: From: sip: at 109.69.217.6;tag=as15b47581 To: test sip: at 109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Consequently, the advice in my response [2] about initially reconfiguring Asterisk with directmedia=no and then fine-tuning the NAT SUPPORT and MEDIA HANDLING settings is valid. [1] http://lists.digium.com/pipermail/asterisk-users/2013-June/279435.html [2] http://lists.digium.com/pipermail/asterisk-users/2013-June/279450.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Hello Matthew, My version is Asterisk 1.6.2.9. Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends) - 1 interface with internal ip (10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP/PSTN gateway. It could be a SIP trunk to a SIP provider Internet, the user does not have to know... Best regards, Mickael 2013/6/13 Matthew J. Roth mr...@imminc.com Mickael MONSIEUR wrote: I have a standard Asterisk configuration: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6 ;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
On Thursday 13 June 2013, Mickael MONSIEUR wrote: Hello Matthew, My version is Asterisk 1.6.2.9. Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. And it's obviously doing NAT, if anything plugged into one interface can see anything plugged into the other. The important question is: Does it work? Because if so, leave it alone. IP addresses are not secret. If anything in your network depends on someone on the outside not knowing one or more of your inside IP addresses, then you are doing it wrong. -- AJS Answers come *after* questions. -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Mickael MONSIEUR wrote: My version is Asterisk 1.6.2.9. Or have you seen NAT ? I have no NAT on my network . Have you seen my little diagram above ? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends ) - 1 interface with internal ip ( 10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP /PSTN gateway . It could be a SIP trunk to a SIP provider Internet , the user does not have to know. .. Mickael, It's hard to be certain without seeing a full SIP trace, but I think the INVITE with the internal IP is actually a re-INVITE that Asterisk is sending to establish a native bridge between the SIP friend and the SIP gateway to PSTN converter. This would allow the endpoints to send their media directly to one another, but in your case I'd expect it to cause one-way audio because the SIP friend shouldn't be able to send RTP packets to the internal IP. If it's a re-INVITE, start by reconfiguring Asterisk with directmedia=no in the [general] section of sip.conf and for all of the endpoints involved in the calls. That should completely eliminate the re-INVITEs at the expense of relaying all RTP through Asterisk, even for calls between two phones on the internal network. After you've confirmed that internal IPs are no longer being sent to external endpoints you can start fine-tuning the NAT SUPPORT and MEDIA HANDLING settings in sip.conf to only allow re-INVITEs when appropriate for your environment. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
You mean the SDP payload? You kind of need that c= is used for RTP transmission. o= always confuses me so I will just say it's important at well. You can put a proxy in the middle and do topology hiding I guess however, that is beyond the scope of this list? Kind Regards, Nick. On 6/12/13, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones)-Asterisk-SIP gateway to PSTN converter 80.236.215.61 109.69.217.6internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Thank you, Mickael -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Mickael MONSIEUR wrote: I have a standard Asterisk configuration: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6 ;tag=as15b47581 To: test sip:@109.69.217.6 ;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
