Re: [asterisk-users] Asterisk NAT
Dear Mr. Newton Thank you for your response. I red the wiki and sip.conf sample file and I know about directmedia option. Actually these options are for times that you know about your connected networks (you know which clients are behind NAT and which are not). But my configuration is different. I have an A2Billing server + Asterisk (these two share a database using ARA); I also wrote a web service that allows users to automatically register and get a username and password. After registration users can connect to Asterisk to call other users. Here I want Asterisk to automatically detect when two users are behind the same NAT and redirect their traffic inside that NAT; this way the load of RTP traffic on Asterisk server will be reduced. On Thu, Feb 20, 2014 at 11:13 PM, Rusty Newton rnew...@digium.com wrote: On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from ignorant tourist to seasoned traveller -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) It is worth nothing that the official Asterisk wiki is at http://wiki.asterisk.org. If there is something missing from there, feel free to let me or someone in #asterisk-dev know and we'll make sure things get updated. One thing I do have on my to-do list is a NAT guide. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 NAT
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com wrote: Dear Mr. Newton Thank you for your response. I red the wiki and sip.conf sample file and I know about directmedia option. Actually these options are for times that you know about your connected networks (you know which clients are behind NAT and which are not). But my configuration is different. I have an A2Billing From my understanding and the documentation, the intent with directmedia=nonat is that it will act like directmedia=yes if the peer is detected as *not* being behind NAT, and directmedia=no if the peer is detected as being behind NAT. This implies that the administrator would not know ahead of time what is needed, otherwise seemingly you would just use yes or no. However I'm still not sure that will be helpful for your particular scenario. users. Here I want Asterisk to automatically detect when two users are behind the same NAT and redirect their traffic inside that NAT; this way the load of RTP traffic on Asterisk server will be reduced. I don't know that this is possible with any simple Asterisk configuration. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 NAT
Anyway, thank you so much. ;-) On Fri, Feb 21, 2014 at 9:32 PM, Rusty Newton rnew...@digium.com wrote: On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com wrote: Dear Mr. Newton Thank you for your response. I red the wiki and sip.conf sample file and I know about directmedia option. Actually these options are for times that you know about your connected networks (you know which clients are behind NAT and which are not). But my configuration is different. I have an A2Billing From my understanding and the documentation, the intent with directmedia=nonat is that it will act like directmedia=yes if the peer is detected as *not* being behind NAT, and directmedia=no if the peer is detected as being behind NAT. This implies that the administrator would not know ahead of time what is needed, otherwise seemingly you would just use yes or no. However I'm still not sure that will be helpful for your particular scenario. users. Here I want Asterisk to automatically detect when two users are behind the same NAT and redirect their traffic inside that NAT; this way the load of RTP traffic on Asterisk server will be reduced. I don't know that this is possible with any simple Asterisk configuration. Good luck! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 NAT
On Tue, Feb 18, 2014 at 10:53 PM, Gholamreza Sabery gr.sab...@gmail.com wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? Hi! As many others mentioned, if you don't get an answer, first go googling then try the #asterisk IRC channel, or maybe the forums at forums.asterisk.org. I noticed your first post today and was going to answer it there, before I saw this new post as well... To attempt answering your question... I believe so. The NAT section of the sip.conf sample contains a lot of helpful options, including: ;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). That is for chan_sip in Asterisk 11, and should also be available in Asterisk 1.8 I've not used a config with this option before, but it sounds like the intent is what you may need. A link to the sample file (that is also included with your source files) http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=markup -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 NAT
On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from ignorant tourist to seasoned traveller -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) It is worth nothing that the official Asterisk wiki is at http://wiki.asterisk.org. If there is something missing from there, feel free to let me or someone in #asterisk-dev know and we'll make sure things get updated. One thing I do have on my to-do list is a NAT guide. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 NAT
On 19/2/14 4:53 am, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? I can't help on the can Asterisk detect they're behind the same NAT part of the question, but I would caution you that an assumption that 'each NAT box has a single external IP' is risky - especially if you have to deal with the possibility of double-NAT and other such evilness (and it's hard to avoid sometimes - how many non-tech people go and buy a wireless router to 'extend their WiFi' rather than an access point, or don't know how to switch said router into AP-only mode). You also have to consider users who have multiple LANs (which might not necessarily be able to route between themselves) behind a single external IP: this one's quite common in shared/managed office environments - one external IP and several RFC1918 VLANs internally, with no routing between them. So in summary, unless you have a considerable level of control over your endpoints such that you can be sure these (and no doubt other) scenarios don't apply, it's probably safest to send RTP traffic through Asterisk regardless, otherwise you're potentially opening up a support nightmare for yourself. 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] Asterisk NAT
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? There is a bit of a tendency on this list to ignore questions that have been answered before. It's disconcerting at first, but remember: *you* are the stereotype tourist here, and *not repeating oneself* is a part of the natives' culture. It is not exactly rudeness, per se, even though it might look that way; just an aversion to saying the same thing twice. No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from ignorant tourist to seasoned traveller -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) -- 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 NAT
Anyway Thank you guys. ;-) On Wed, Feb 19, 2014 at 12:25 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? There is a bit of a tendency on this list to ignore questions that have been answered before. It's disconcerting at first, but remember: *you* are the stereotype tourist here, and *not repeating oneself* is a part of the natives' culture. It is not exactly rudeness, per se, even though it might look that way; just an aversion to saying the same thing twice. No answer on the list probably just means the question was answered before; so your best bet is to search the mailing list archives and the wiki at http://voip-info.org Eventually, you will have been yomping around in Tech Land for long enough to graduate from ignorant tourist to seasoned traveller -- and then you get to ignore noob questions yourself. Or set yourself up as a tour guide, if you feel that way inclined :) -- 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 -- _ -- 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] Asterisk NAT
Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? -- _ -- 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 NAT
On Wed, 19 Feb 2014, Gholamreza Sabery wrote: Hello, a few days ago I sent a question: http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html but no one answered me! I just want to know is it possible or not? If it were only so easy... Participation in these lists is purely voluntary. You only get a reply if you managed to pique somebody's interest and they feel they have something to offer -- which may be commiseration rather than an answer. Having said all that, there are some incredibly knowledgeable and generous participants who have helped me out of some sticky situations. Think of it like a message in a bottle. You cast it out to sea and you may make an incredible contact. You may not. Something to keep in mind. These lists is largely 'US centric' by which I mean that if you post after the US work day ends (even accounting for 'programmer hours') you are limiting your potential audience. Posting late on a Friday afternoon can be an exercise in futility. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Asterisk NAT
I have an Asterisk box with a public IP address and two SIP clients behind the same NAT device(I also have SIP clients behind different NATs). I want to know is it possible for Asterisk to detect if both clients are behind the same NAT and use direct media between them and use other options for clients that are behind different NATs? By detection I mean is it possible for Asterisk to take a look at the public IP address of packets and if both packets have the same IP address it tells the clients to send RTP traffic directly to each other. Is there a module or piece of code for this behavior in Asterisk?? PS:I assumed each NAT box has a single external IP address, and this assumption is good for me. -- _ -- 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 NAT friendly settings
On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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 NAT friendly settings
On 1/8/2014 4:17 AM, Ishfaq Malik wrote: On 7 January 2014 22:55, Adam Moffett adamli...@plexicomm.net mailto:adamli...@plexicomm.net wrote: I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia Have you added directmedia=no? Nope, I'll look into that. -- _ -- 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] Asterisk NAT friendly settings
I'm asking about this scenario: Asterisk(public IP) -- Internet -- Router (public IP) -- SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia -- _ -- 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] Asterisk, NAT, and RTP?
Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _ -- 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] Asterisk, NAT, and RTP?
You'd need RTP ports open for asterisk then. Transfers and parking can be done at the SIP level, asterisk doesn't have to be in the RTP path, as it can reinvite itself into the callpath as necessary. On Wed, Jan 27, 2010 at 5:23 AM, Vincent codecompl...@free.fr wrote: Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _ -- 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 -- _ -- 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
[asterisk-users] Asterisk, nat, gizmo and fwd
Hi there everyone! I use asterisk as a home pbx. My internet connection is a DSL one, and I have a Linksys WRT54G that nat things for me in a 192.168.X.X style network. I've installed asterisk on my mac, and tried several examples I've found on the net (voip-info, gizmo, etc.) about how to create a Gizmo and a FWD trunk. However, all my attempts failed. The FWD thing kinda worked, but it wouldn't receive any calls. The Gizmo one worked ONLY to call their echo test - If I tried to call a real number I got some serious noise, sometimes the phone ringing, but never getting audio through. It also brings my asterisk down (1.4.0) - it just hangs, without the peers managing to authenticate on the server again. Has anyone succesfully managed to connect to Gizmo and FWD in a similar setup and would be so kind to share the configs? []'s Francis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- NAT -- Internet -- NAT -- Sipura-3K (No Asterisk)
Joseph: I think that was under the sip tab you have to configure the sip proxy, just put the address of the firewall where you are doing port forwarding to the asterisk box. Also in the nat field at the sipura device put yes. And finally you have to open and forward the UDP ports of the firewall in the sipura LAN and forward it to the sipura device. Let me know if this works. Lucas Alvarez Joseph wrote: So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 1 - 2 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk -- NAT -- Internet -- NAT -- Sipura-3K (No Asterisk)
So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 1 - 2 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk NAT behaviour
Hello, Does it make a difference for the NAT traversal capabilities of Asterisk if the users are registered directly to Asterisk or if they are registered to a SIP proxy which just relays the calls to Asterisk? Are there any cases where Asterisk would be able to traverse the NAT if the user is registered directly which won't work if the user is registered on a proxy? bye, /gst signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk+Nat+sipura (Help)
Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to "Nat", Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk+Nat+sipura (Help)
I don't think the problem is NAT-related. Looks like To header in SIP INVITE message do not match to User ID in sipura settings. On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote: Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 Not Found back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to Nat, Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk+Nat+Sipura/Linksys
Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to "Nat", Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk Nat solution for such scenario?
Hi, everyone. I have such a setup: Asterisk---FireWall---internet--FireWall |--Cisco7960 |--Xlite And I followed the instructions I got from the internet, change the sip.conf in asterisk like this: [general] nat=yes extexternip=203.92.75.87 .. [cisco] . [xlite] ... Finally I manage to get Xlite works perfectly, while for Cisco7960 I have to disabled the sip password, then I can make outgoing call, but still cannot receive incoming call. Is there anyone encouter the same problem before? any ideas how to solve it? Thanks, Liangliang ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk NAT spa-2000
Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 [EMAIL PROTECTED] context=sip callerid=James 2001 secret=hidden canreinvite=no allow=ulaw nat=yes qualify=yes I added the nat and qualify entries after hunting round google but still get this error, spot the no nat bit. to 81.178.77.67:34504 Retransmitting #2 (no NAT): OPTIONS sip:81.178.77.67:34504 SIP/2.0 Via: SIP/2.0/UDP 62.188.201.123:5060;branch=z9hG4bK68af34fa From: asterisk sip:[EMAIL PROTECTED];tag=as5582cfae To: sip:81.178.77.67:34504 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 18 Jul 2004 12:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 any ideas anyone thanks in advance Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk NAT spa-2000
I would comment out these lines in sip.conf ;externip=111.222.333.444 ;localnet=192.168.1.0 ;localmask=255.255.255.0 Then set nat=no -Original Message- From: Simon Chappell [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004 4:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk NAT spa-2000 Hi All, I have a asterisk box that is now on its own static address on the net.it was originally behind a nat firewall. The problem I have is that the remote SPA-2000's that are behind nat firewalls now fail. here is relevent sip.con entry [2001] type=friend username=2001 host=dynamic defaultip=81.178.77.67 allow=ulaw dtmfmode=rfc2833 [EMAIL PROTECTED] context=sip callerid=James 2001 secret=hidden canreinvite=no allow=ulaw nat=yes qualify=yes I added the nat and qualify entries after hunting round google but still get this error, spot the no nat bit. to 81.178.77.67:34504 Retransmitting #2 (no NAT): OPTIONS sip:81.178.77.67:34504 SIP/2.0 Via: SIP/2.0/UDP 62.188.201.123:5060;branch=z9hG4bK68af34fa From: asterisk sip:[EMAIL PROTECTED];tag=as5582cfae To: sip:81.178.77.67:34504 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 18 Jul 2004 12:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 any ideas anyone thanks in advance Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk NAT Gateway Setup
I am currently using Asterisk behind Belkin NAT router. With what ever NAT router I have used, I have had difficulties in registration and audio problems with my SIP provider (Iconnect and Nikotel) It was suggested that I try to connect the asterisk box directly to the internet avoiding the NAT transition. As I will still need internet connectivity, I am trying to make the asterisk box the NAT gateway. I have an additional NIC for my Asterisk box. As I am no Linux or Asterisk expert, can anyone make suggestions as to this approach and any recommended steps to accomplish this? Also, how would Asterisk know which interface to bind to? I know there is a bindaddress= parameter in the SIP config, but the address to the internet is dynamic via DHCP from my cable provider. Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Nat Issue
Here's the problem my sipura 2000 is setup on Nat Network in my office and my Asterisk Server is setup also on Nat Network at home the sipura can register and get calls but no audio comes in and out of the sipura and when i dial local extensions on the sipura i get this error message. any suggestions on what i can try as work around. *CLI NOTICE[1158921008]: File chan_sip.c, Line 5394 (handle_request): Unknown SIP command 'NOTIFY' from '205.158.181.200'WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)NOTICE[1158921008]: File chan_sip.c, Line 5394 (handle_request): Unknown SIP command 'NOTIFY' from '205.158.181.200'WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)WARNING[1158921008]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)
