Re: [asterisk-users] sometimes extensions can't be called
Hi Noah, I mentioned last time i created a trunk in between my two asterisk servers. I have a dialplan that will detect if a user is registered on *1 or *2. e.g if user is in local server Dial(SIP/100) if on the other server Dial(SIP/[EMAIL PROTECTED]) my prob is if it dials to the other server, i get invite failed. so if Dial(SIP/100) it's ok if Dial([EMAIL PROTECTED]) i get this [Jul 26 02:42:33] NOTICE[14467]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '101 sip:[EMAIL PROTECTED];tag=as27bafd37' i have this on sip.conf [other-server] type=friend insecure=port,invite host=200,201,202,204 the user that is calling is registered, but i think the one being denied is the other asterisk. how can i allow it. TIA regards, nhadie Noah Miller wrote: Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL
Re: [asterisk-users] sometimes extensions can't be called
Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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] sometimes extensions can't be called
Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Wednesday, July 23, 2008 03:40 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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] sometimes extensions can't be called
Hi Sir, Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting ‘qualify=yes’ in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you’ll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be ‘UNREACHABLE’. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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
Re: [asterisk-users] sometimes extensions can't be called
Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones
Re: [asterisk-users] sometimes extensions can't be called
Hi Sir Thanks for your reply, since i don't know how to setup DUNDi, what i did for now is create a sip peer between the 2 servers and just use the regserver on the realtime db. but now with that setup i cant play the music on hold of the extension i'm calling to, e.g i'm 118102 i called 118103 1182102 has moh class moh-118102 and 118103 has class moh-118103. if the call is on the same server i have no issues moh plays the class of the user, but when the extension is on the other server and i put it on hold, it always plays the class default, anyway i will try to figure that one out also, thanks again to all your reply. regards, nhadie Noah Miller wrote: Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject
[asterisk-users] sometimes extensions can't be called
Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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] sometimes extensions can't be called
Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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] sometimes extensions can't be called
Hi, i see my extensions are there: 118103/118103 210.212.213.214 D N 5060 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118102/118102 210.212.213.214 D N 37743 Unmonitored 118101/118101 210.212.213.214 D N 5064 Unmonitored 118103/118103 210.212.213.214 D N 5060 Unmonitored and i have this on both servers: 17 sip peers [Monitored: 0 online, 0 offline Unmonitored: 15 online, 2 offline] regards, nhadie --- On Wed, 7/23/08, Darryl Dunkin [EMAIL PROTECTED] wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? ‘sip show peers’ in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nhadie Ramos Sent: Tuesday, July 22, 2008 21:52 To: asterisk-users@lists.digium.com Subject: [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are behind NAT and using stun server. sip keep-alive is enabled onxlite or ip phone. but it's just very inconsistent. i don't know where to look at to fix this. any idea? nhadie ___ -- 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
