[asterisk-users] asterisk-users Digest, Vol 70, Issue 25

[Vardan]

Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you


Server1:

sip.conf

[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver2]
  exten =>  _X.,1,Noop(Call to server2)
  exten =>  _X.,2,Dial(SIP/interboxserver2/${EXTEN})
  exten =>  _X.,3,Hangup

[callfromserver2]

exten => _X.,1,Noop(Call from server2)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Server2:

sip.conf

[interboxserver1]
type=friend
host=192.168.250.111
context=callfromserver1
disallow=all
allow=ulaw
allow=alaw
allow=g729

extensions.conf

[calltoserver1]
  exten =>  _X.,1,Noop(Call to server1)
  exten =>  _X.,2,Dial(SIP/interboxserver1/${EXTEN})
  exten =>  _X.,3,Hangup

[callfromserver1]

exten => _X.,1,Noop(Call from server1)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup


Try so, I think it must work.
And also, look and delete any another records in both servers in 
sip.conf about this servers settings.

Vardan

Nasir Javaid wrote:
> Hi Vardan
>
> I did same as you told and deleted the SIP information in Astdb and
> restarted asterisk. but the result was same.
>
> as you said there might be mistake in sip.conf so i am pasting both
> servers configuration here..
>
> 1- nasir.server.com <http://nasir.server.com>
>
> [abc]
> username=abc
> type=friend
> secret=mysecret
> nat=yes
> mailbox=12234568
> incominglimit=2
> outgoinglimit=2
> host=dynamic
> dtmfmode=rfc2833
> context=payasyougo
> canreinvite=yes
> callerid="Nasir Qazi" <12234>
> accountcode=6:0:abc
> amaflags=default
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
>
>
> 2- 192.168.0.254 (client system)
>
> [abc]
> type=peer
> username=abc
> secret=mysecret
> host=nasir.server.com <http://nasir.server.com>
> context=default
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=very
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> ;qualify=yes
>
> [caller]
> type=friend
> secret=123456
> host=dynamic
> callerid="caller <12129887777>"
> context=out
> nat=yes
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> t38_udptl=yes
> qualify=yes
>
>
> I have registered [caller] on xlite at client system and dialing
> following context in local system that will dial [abc]
>
> [out]
> exten=> _X.,1,Dial(SIP/${EXTEN}@abc,30,1)
> exten=> _X.,n,Hangup
>
>
> as you can see above *highlighted that context of abc is payasyougo.*
> problem is that i want the call to land in that context on
> nasir.server.com <http://nasir.server.com>, which works if i use
> register string. but without register string call goes to default
> context on nasir.server.com <http://nasir.server.com>
>
> regards,
>
> Nasir Javaid
>
> Message: 19
> Date: Tue, 11 May 2010 20:54:30 +0500
> From: Vardan <hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
> To: asterisk-users at lists.digium.com <mailto:asterisk-users at lists.digium.com>
> Message-ID: <hsbujk$qk9$1 at dough.gmane.org <mailto:1 at dough.gmane.org>>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello Nasir
>
> I have some please.
> Do so, it help.
> Find all records about interexchange beetwen this two server and delete
> all records in sip.conf for this both server (first make backup
> sip.conf, or any another conf file that you use).
> restart asterisk.
> look in astdb about this old records, if any found, delete him
> Next, create new record in sip.conf on both servers, without
> registration string, reload sip.conf.
> give him right context from extensions.conf.
>
> Can you do this?
>
> I think is some mistake about configuration in sip.conf, you have I
> think two same records (peer or friend).
>
> Vardan
>