Re: [asterisk-users] Nat Issue - I think
On Sat, 2010-11-13 at 13:43 -0500, Dan Journo wrote: Hi, I'm using qualify= on my asterisk server that provides outgoing pstn calls to a few companies. I've got one client in particular that has their own asterisk server which is connected to my server. This client seems to be having a nat issue. It's not a connectivity issue as i've tried constant pings and the line is up constantly. I'm getting the following: [2010-11-13 17:56:27] NOTICE[26082] chan_sip.c: Peer 'client_201' is now UNREACHABLE! Last qualify: 29 [2010-11-13 17:56:37] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (26ms / 5000ms) [2010-11-13 17:58:12] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 28 [2010-11-13 17:58:22] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (27ms / 5000ms) [2010-11-13 17:59:58] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 27 [2010-11-13 18:00:08] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (27ms / 5000ms) [2010-11-13 18:01:43] NOTICE[26082] chan_sip.c: Peer 'client _201' is now UNREACHABLE! Last qualify: 29 [2010-11-13 18:01:53] NOTICE[26082] chan_sip.c: Peer 'client _201' is now Reachable. (28ms / 5000ms) Looking at the sip log, on the client's server, the sip OPTIONS packets arent being received. Then suddenly the sip packets start being received again. (without sending out a new packet to open up the nat mapping). I've tried replacing the router because I thought it was faulty. Here is the SIP log from my server: http://pastebin.com/ZbDYGG9R Finally, I tried mapping port 5060 so to avoid NAT issues, but that didnt help. Could there be an ISP problem? Any assistance would be appreciated. Many thanks Dan These are always loads of fun to try and fix... Check if the router has a sip ALG than needs disabling such as on XyZel routers. Try specifying a port on the phone if possible and forwarding it through the router. If you have the resources you could try creating a proxy server that handles the natting Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] NAT issue (i think?)
sorry for the late reply. got tied with something else. please see attached ngrep result. my setup is simple, i add the user on the realtime database. i configure a phone to set my domain (i have DNS SRV enabled on my domain) my clients are usually using ADSL links and have those SOHO ADSL router (usually linksys). IP Phones are on DHCP, i set NAT Keep-Alive to Yes and NAT Mapping to Yes, i also set the ffg, to Yes, Handle VIA received, Handle VIA rport,Insert VIA received and Insert VIA rport. also if there are more than 1 phone behind the same NAT, i set different SIP port and each phone, i'm just wondering why it only happens on linksys phones, using yealink and grandstream it's ok. Thanks again. Regards Ron On 9/29/10 7:39 AM, Danny Dias wrote: Hello Ron.. The answer that i see here is only a trying to a Register...means the REGISTRATION procedures are taking a significant amount of time. You should get a 200 OK Can you lease make a simple draw of your architecture? seems to be a NAT problem, that's for sure REgards! 2010/9/28 Ronnha...@gmail.com Hi Danny On the pap2 by default it is set to 3600 and i have not change that. by the way, is the NAT keep-alive same with the NOTIFY message? coz i am seeing my asterisk respond to those as bad event could that be causing it to loose the registration? here's the registration from ngrep: U 78.65.34.12:5094 - 12.34.56.78:5060 REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. From: Kristinesip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristinesip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. Max-Forwards: 70. Contact: Kristinesip:456...@78.65.34.12:5094;expires=3600. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . U 12.34.56.78:5060 - 78.65.34.12:5094 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. From: Kristinesip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristinesip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias wrote: You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ronnha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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 # U 40.30.20.10:5060 - 10.20.30.40:5060 REGISTER sip:sip.mydomain.com SIP/2.0..Via: SIP/2.0/UDP 40.30.20.10:5060;branch=z9hG4bK-8890bff6;rport..From: 123456 sip:123...@sip.mydomain.com;tag=e9 cb2ff4bca8b074o0..To: 123456 sip:123...@sip.mydomain.com..Call-ID: 1da9d5f4-2eea4...@40.30.20.10..cseq: 1 REGISTER..Max-Forwards: 70..Contact: 123456 sip:43 9...@40.30.20.10:5070;expires=86400..User-Agent:
Re: [asterisk-users] NAT issue (i think?)
On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote: got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. Sounds like NAT problems, do you have qualify enabled for these devices? Also the Linksys devices have a keep alive option (NAT Keep Alive Enable). But even with both these setting enabled NAT gateways sometimes seem to lose track of SIP sessions (I have more trouble with Cisco devices than Linux routers), setting the UDP session timeout to 10m seems to help (default is something like 3m). -- Daniel Tryba -- _ -- 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] NAT issue (i think?)
hi daniel thank you for your reply, i have enabled NAT keep-alive and NAT-Mapping on the Linksys devices. i have enabled qualify before but it seems it's worst as the linksys devices keeps on rebooting and there was one issue that when i set qualify to all users my server bogged down and asterisk crashed. where do i enable the UDP session timeout? on the linksys devices or the asterisk? TIA Regards Ron On 9/28/10 6:14 PM, Daniel Tryba wrote: On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote: got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. Sounds like NAT problems, do you have qualify enabled for these devices? Also the Linksys devices have a keep alive option (NAT Keep Alive Enable). But even with both these setting enabled NAT gateways sometimes seem to lose track of SIP sessions (I have more trouble with Cisco devices than Linux routers), setting the UDP session timeout to 10m seems to help (default is something like 3m). -- _ -- 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] NAT issue (i think?)
You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060 register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ron nha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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] NAT issue (i think?)
On Tue, Sep 28, 2010 at 07:08:36PM +0800, Ron wrote: i have enabled qualify before but it seems it's worst as the linksys devices keeps on rebooting and there was one issue that when i set qualify to all users my server bogged down and asterisk crashed. Never seen these problems before related to qualify, how many devices are there? where do i enable the UDP session timeout? on the linksys devices or the asterisk? Neither, that is a property of the NAT device. You should figure out when the problem happens wether the NAT device has any knowledge of the UDP session between the SIP device and Asterisk. -- Daniel Tryba -- _ -- 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] NAT issue (i think?)
Hi Danny On the pap2 by default it is set to 3600 and i have not change that. by the way, is the NAT keep-alive same with the NOTIFY message? coz i am seeing my asterisk respond to those as bad event could that be causing it to loose the registration? here's the registration from ngrep: U 78.65.34.12:5094 - 12.34.56.78:5060 REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. From: Kristine sip:456...@sip.mydomain.com;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com. Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. Max-Forwards: 70. Contact: Kristine sip:456...@78.65.34.12:5094;expires=3600. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . U 12.34.56.78:5060 - 78.65.34.12:5094 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. From: Kristine sip:456...@sip.mydomain.com;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com. Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias wrote: You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ronnha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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] NAT issue (i think?)
Hello Ron.. The answer that i see here is only a trying to a Register...means the REGISTRATION procedures are taking a significant amount of time. You should get a 200 OK Can you lease make a simple draw of your architecture? seems to be a NAT problem, that's for sure REgards! 2010/9/28 Ron nha...@gmail.com Hi Danny On the pap2 by default it is set to 3600 and i have not change that. by the way, is the NAT keep-alive same with the NOTIFY message? coz i am seeing my asterisk respond to those as bad event could that be causing it to loose the registration? here's the registration from ngrep: U 78.65.34.12:5094 - 12.34.56.78:5060 REGISTER sip:sip.mydomain.com SIP/2.0. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;rport. From: Kristine sip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. Max-Forwards: 70. Contact: Kristine sip:456...@78.65.34.12:5094;expires=3600. User-Agent: Linksys/PAP2T-3.1.15(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. . U 12.34.56.78:5060 - 78.65.34.12:5094 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 78.65.34.12:5094;branch=z9hG4bK-25a5eec4;received=78.65.34.12;rport=5094. From: Kristine sip:456...@sip.mydomain.comsip%3a456...@sip.mydomain.com ;tag=68fc368d164925e0o0. To: Kristine sip:456...@sip.mydomain.com sip%3a456...@sip.mydomain.com . Call-ID: c9bd8b57-f7bdc...@192.168.1.52. CSeq: 116228 REGISTER. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. On 9/28/10 7:24 PM, Danny Dias wrote: You have to increase the time of expiration for the Register...on linksys devices is located on Proxy and Registration section under the EXTN: (Where N is the extension number) Try putting this to: 3600 To check wheter or not is loosing Register, try with ngrep-sip and check it: ngrep -p -q -W byline port 5060register.pkt Then post here the content of register.pkt; but please, after issuing the change explained above! Regards! 2010/9/28 Ronnha...@gmail.com Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is troublesome. i'm using Asterisk 1.4.31 with the following realtime config: rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=no one thing i noticed is that it only seems to happen on linksys devices e.g. PAP2 and SPA's. another phone i'm using is yealink and so far no client has complain about it. hope anyone can help. thank you. regards Ron -- _ -- 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 -- _ -- 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
