[asterisk-users] sometimes extensions can't be called

[Nhadie]

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:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Nhadie Ramos
> *Sent:* Wednesday, July 23, 2008 03:40
> *To:* asterisk-users at 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 /<ddunkin at netos.net>/* wrote:
> 
> From: Darryl Dunkin <ddunkin at netos.net>
> Subject: RE: [asterisk-users] sometimes extensions can't be called
> To: nhadie.ramos at yahoo.com, asterisk-users at 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:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Nhadie Ramos
> *Sent:* Tuesday, July 22, 2008 21:52
> *To:* asterisk-users at 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
> 
>  
> 
>  
> 
> 
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