Re: [asterisk-users] Help with remote users
Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
I have just confirmed that they may be having a problem with double NAT. They have two ATAs, and they have two different DSL connections. One set-up goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems) to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has the ATA plugged into it. The other ATA is configured from a DSL Modem (again, I was told NAT Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in there. I have the same issues on both ATAs. I have no idea why their network is as poorly designed as it is, the bad part is I have to make sure the phones work there and try to troubleshoot from 3000 miles away. Any work arounds for a problem because of double NAT? A quick and dirty solution for them to get their phones working right? Steve Anness On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Help with remote users
Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. The fact that they work initially is probably a clear indication that the NAT bindings are closing up after a few minutes. In some cases it does not matter that you have qualify=yes, since the router only keeps bindings open if the traffic is being generated from the inside-outside. Your solution would be to enable the keep-alive settings on the PAP2 and set it low to something like 15 seconds. The setting is under the tab of line 1 and line 2 and its called NAT Keep Alive Enable. Andres http://www.neuroredes.com do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
THANK YOU!!! This appears to have worked. I am assuming we can do the same thing on our SPA-962s that we send to make sure they work with no problems. Thank you to everyone here for your help. This is an excellent group to have access to for questions. I hope to learn and be able to help others. Steve Anness On 10/7/08 9:53 AM, Andres [EMAIL PROTECTED] wrote: Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. The fact that they work initially is probably a clear indication that the NAT bindings are closing up after a few minutes. In some cases it does not matter that you have qualify=yes, since the router only keeps bindings open if the traffic is being generated from the inside-outside. Your solution would be to enable the keep-alive settings on the PAP2 and set it low to something like 15 seconds. The setting is under the tab of line 1 and line 2 and its called NAT Keep Alive Enable. Andres http://www.neuroredes.com do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
Load the firmware of www.dd-wrt.com on that WRT54G and then put all the VoIP devices directly behind it. It MIGHT work to set the first NAT router to have the 2nd NAT router in the 1st's DMZ... but I prefer to do things The Right Way. On Tue, Oct 7, 2008 at 7:24 AM, Steve Anness [EMAIL PROTECTED] wrote: I have just confirmed that they may be having a problem with double NAT. They have two ATAs, and they have two different DSL connections. One set-up goes from the first DSL Modem (NAT Wirless are disabled on the DSL Modems) to a Linksys WRT110 and then there is a WRT54G hooked in to the 110 that has the ATA plugged into it. The other ATA is configured from a DSL Modem (again, I was told NAT Wireless were disabled on the modem) to a WRT600N and the ATA is plugged in there. I have the same issues on both ATAs. I have no idea why their network is as poorly designed as it is, the bad part is I have to make sure the phones work there and try to troubleshoot from 3000 miles away. Any work arounds for a problem because of double NAT? A quick and dirty solution for them to get their phones working right? Steve Anness On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: Make sure they are not using double NAT. Many ISPs these days send their subscribers a modem that in reality is a router. Also if you can post the PAP2 configuration. I hope you are using provisioning.. too bad Linksys makes it possible to obtain that information. On Mon, Oct 6, 2008 at 12:40 PM, Steve Anness [EMAIL PROTECTED] wrote: I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the no-service message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn't actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn't blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T's to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can't understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be unreachable and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T’s to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can’t understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be “unreachable” and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with remote users
I am using NAT so the ATAs are configured with a proxy server. Qualify is set to yes. Here is what is happening. After they plug in the ATA on the otherside, and things register and I can call and they can call. After several minutes I try to call and then get the ³no-service² message. This is with Qualify=yes. -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0, CDR(accountcode)=Hiramine) in new stack -- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0, CALLERID(all)=(Hiramine) 2545239280) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0, SIP/17110-1SIP/17112-1|20| w) in new stack [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Oct 6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (2:0/0/2) -- Executing [EMAIL PROTECTED]:4] Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack If qualify is equal to no, then it just trys to ring, I get no errors it just keeps trying (except the phone doesn¹t actually ring). I just wrote an email to find out more about their network settings there. To see if the ATAs are actually getting a private or public address. If they are getting a public address I suppose I can just set NAT=no and as long as I can ping the public address and port 5060 isn¹t blocked by a firewall than I should be able to resolve these issues. Thanks for your time. Steve Anness On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote: On Oct 6, 2008, at 1:53 PM, Steve Anness wrote: I know I have asked about this before, but I thought that I would ask again with some more detail and maybe someone will have an idea. This is my first time to be setting up an asterisk server and I have a server running. I sent Linksys PAP2T¹s to several remote users. Only one out of the four users actually work like they should. One of the other users I am assuming is behind a firewall on his wireless router and needs to open up the proper ports. However, I have two users in New York on a DSL connection and I can¹t understand why things are happening like they are. Here Is the situation. Both users can plug in their ATAs and I can watch the server output, they register and then they can make calls and I can call them. Some time later (usually within minutes) the ATAs show to be ³unreachable² and I can no longer call; however, they can still make calls. do you have qualify=yes ?? Is asterisk on a public IP? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
