[asterisk-users] sometimes extensions can't be called
Nhadie
nhadie at tbgi.net.ph
Fri Jul 25 13:55:28 CDT 2008
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/100 at other-server)
my prob is if it dials to the other server, i get invite failed.
so if Dial(SIP/100) it's ok
if Dial(SIP.100 at other-server) 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:101 at 200.201.202.203>;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:
>>>
>>> 123/123 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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>
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