Re: [asterisk-users] Sending calls from behind NAT
On 12-11-13 11:38 PM, bilal ghayyad wrote: Dears; What Jian said is the right and it worked. But I have the following questions: Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to set the localnet or it is enough to set the externip? The IP information is called IP subnetting which is the basic rule about IP address. It seems you are using a Class C private subnet so 192.168.10.2 will never be a correct network address. (Depend on your subnet mask, the network address could be like something else. But the last octet 2 is obviously wrong!) If your Asterisk is behind a NAT router/firewall, you need: localnet= externip= nat=yes Regards, Jian From the other side, I am using Asterisk 1.8.12.0 and when I was searching in the sip.conf, I did not find externip (so I added by my hand) and I remember very well that before I was able to find the externip in the sip.conf, although I am finding externadd. So why this? One more thing, what is the difference between externadd and externip? Regards Bilal --- Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 I think these setting are all wrong: 1. local network should be something like: 192.168.10.0 2. Subnetmask cant' be 255.255.255.254 ! 3. externip=x.x.x.x (Not externadd) Jian But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To: sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- _ -- 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
[asterisk-users] Sending calls from behind NAT
Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To: sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- _ -- 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] Sending calls from behind NAT
Dear Bilal, I understood correctly that the problem is that calls drops? What router are you using? Eliezer On 11/13/2012 11:16 PM, bilal ghayyad wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisksip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To:sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID:6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- Eliezer Croitoru https://www1.ngtech.co.il IT consulting for Nonprofit organizations eliezer at ngtech.co.il -- _ -- 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] Sending calls from behind NAT
On 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 This should be externip not externadd You are still sending them your local address Cheers Duncan -- _ -- 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] Sending calls from behind NAT
It looks like you need to enable the sip application layer gateway or ALG on your router. The problem is not exactly a Nat issue. The problem is most likely with the sip header keeping the private IP address, the ALG when enabled will change this to your public On 13 Nov 2012, at 21:17, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To: sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- _ -- 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] Sending calls from behind NAT
On 13/11/12 9:31 pm, Leighton Brennan wrote: It looks like you need to enable the sip application layer gateway or ALG on your router Quite often the reverse is true. Most routers (at least those I've used) seem to have such a lousy implementation of a SIP ALG it's often far better to just disable it and do your own NAT fixups in Asterisk (as others have indicated in previous posts). In fact, it's now the first thing we advise clients to do when they report call problems or one-way audio: disable the SIP ALG in your router. Sadly, there are also quite a few routers out there now that have ALGs that can't be disabled (or that make it extremely difficult to disable them). Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Sending calls from behind NAT
I'm with Duncan, you need a public IP address, not private. Chris BudinickNetwork Technician RAINIER CONNECTFrom: "Duncan Turnbull" dun...@e-simple.co.nzTo: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comSent: Tuesday, November 13, 2012 1:29:28 PMSubject: Re: [asterisk-users] Sending calls from behind NATOn 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference". Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 This should be externip not externaddYou are still sending them your local addressCheers Duncan--_-- 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/helloasterisk-users mailing listTo 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] Sending calls from behind NAT
Check reinvite and NAT settings on the line as well as the SIP peers. You can use a stun client from inside your network to see what’s going on with the NAT From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Budinick Sent: Tuesday, November 13, 2012 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending calls from behind NAT I'm with Duncan, you need a public IP address, not private. Chris Budinick Network Technician RAINIER CONNECT _ From: Duncan Turnbull dun...@e-simple.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 13, 2012 1:29:28 PM Subject: Re: [asterisk-users] Sending calls from behind NAT On 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 This should be externip not externadd You are still sending them your local address Cheers Duncan -- _ -- 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 attachment: image001.jpg-- _ -- 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] Sending calls from behind NAT
On 12-11-13 01:16 PM, bilal ghayyad wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 I think these setting are all wrong: 1. local network should be something like: 192.168.10.0 2. Subnetmask cant' be 255.255.255.254 ! 3. externip=x.x.x.x (Not externadd) Jian But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To: sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- _ -- 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] Sending calls from behind NAT
Dears; What Jian said is the right and it worked. But I have the following questions: Why 192.168.10.2 is wrong and I have to use 192.168.10.0? Also, do I have to set the localnet or it is enough to set the externip? From the other side, I am using Asterisk 1.8.12.0 and when I was searching in the sip.conf, I did not find externip (so I added by my hand) and I remember very well that before I was able to find the externip in the sip.conf, although I am finding externadd. So why this? One more thing, what is the difference between externadd and externip? Regards Bilal --- Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 I think these setting are all wrong: 1. local network should be something like: 192.168.10.0 2. Subnetmask cant' be 255.255.255.254 ! 3. externip=x.x.x.x (Not externadd) Jian But even, the calls are drop .. so what I have to do? The following what I get when I enabled the sip debug: --- SIP read from UDP:194.0.220.220:5060 --- SIP/2.0 403 UA behind NAT not accepted here Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK123c8781;rport=5060;received=196.40.164.239 From: asterisk sip:gwbilalkwpbx@192.168.10.2;tag=as45d7c63b To: sip:9617565...@outbound.exxs.com:5060;tag=84f31a80f7bc633204d0bd8d76e9cb24.53a5 Call-ID: 6f68f414109ba1ff596726c74a0cb5ac@192.168.10.2:5060 CSeq: 102 INVITE P-Behind-NAT: source Server: Service Provider Global Proxy v2 Content-Length: 0 So what could resolve my problem? Regards Bilal -- _ -- 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
