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: >>> >>> 123/123 64.23.49.5 D N 15103 OK (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:* [email protected] >>> *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], [email protected] >>> 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:* [email protected] >>> *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 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
