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

Wed Jul 23 17:07:35 CDT 2008

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:* 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
>>
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> -- 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
>