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: > > 123/123 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
