[asterisk-users] asterisk-users Digest, Vol 70, Issue 24

[Nasir Javaid]

Yes this scenario works on my 2 systems which are at LAN. I made one system
as server (192.168.0.20)  and registered from other system... it is fine but
now there is a different scene.

actually there is a registered user named abc at system1 (192.168.0.20)
having context [payasyougo] which is used to do outbound calls. we want to
use this user's context and account so that when we register this account on
our other system (192.168.0.254) call should be dropped into the context of
abc which is [payasyougo].

now this works fine on above mentioned systems, and calling from system
(192.168.0.254) like this

SIP/${EXTEN}@abc

sends call to the abc's context [payasyougo] and from there system1
(192.168.0.20) takes charge of dialing out the number in ${EXTEN}.

but when i change system 1 (192.168.0.20) to my real server (e.g.
nasir.server.com) which has abc as user configured same as on system1
(192.168.0.20), call goes to [default] instead of going to [payasyougo]
context and is treated as incoming call...

when we use register string calls works ok on real server too. I also tried

SIP/abc:mysecret/${EXTEN} and
SIP/${EXTEN}@abc:mysecret

but nothing seems to work.

there is another problem that sometime my real server (nasir.server.com)
becomes unreachable and this error is returned

 NOTICE[3898]: chan_sip.c:11489 sip_reg_timeout:    -- Registration for '
abc at nasir.server.com' timed out, trying again (Attempt #38)

It may be a simple problem but is driving my crazy... please help me out

thanks in advance

Nasir Javaid


> Message: 6
> Date: Tue, 11 May 2010 13:57:23 +0500
> From: Nasir Javaid <nasirjavaidnasir at gmail.com>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23
> To: asterisk-users at lists.digium.com
> Message-ID:
>        <AANLkTim3qsPCY0sY3hanDxtTg8HPbXJEmj7HWZSknO1W at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanks Vardan,
>
> I will like to know if this scenario can work when peer is not having fixed
> ip and we use
> host = nasir.server.com
> ?
> also I have set insecure=invite,port
>
> what if i use
> insecure=no
>
> thanks again.
>
> Message: 24
> Date: Tue, 11 May 2010 10:52:14 +0500
> From: Vardan <hvardan71 at gmail.com>
> Subject: Re: [asterisk-users] Dialing a SIP Peer without using
>       register strin
> To: asterisk-users at lists.digium.com
> Message-ID: <hsarab$ok7$1 at dough.gmane.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Remove username and secret and use IP authentication on both side
>
> [server1_abc]
> type=peer
> host=192.168.0.20
> context=default
> dtmfmode=rfc2833
> canreinvite=yes - canreinvite with nat=yes is not working
> insecure=invite,port
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> qualify=yes
>
>
>
> [server2_abc]
> type=peer
> host=192.168.0.21
> context=default
> dtmfmode=rfc2833
> canreinvite=yes
> insecure=invite,port
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> qualify=yes
>
>
>
> Nasir Javaid wrote:
> > Hi,
> >
> > I am new to this list and this is first time i m posting here. please
> > help me out
> >
> > currently I am working on dialing a sip peer on an asterisk server from
> > 2nd asterisk server. scenario is like this.
> >
> > on my system i am using this peer in sip.conf.
> >
> > [abc]
> > type=peer
> > username=abc
> > secret=mysecret
> > host=192.168.0.20
> > context=default
> > dtmfmode=rfc2833
> > ;restrictcid=no
> > canreinvite=yes
> > insecure=invite,port
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g729
> > allow=gsm
> > nat=yes
> > qualify=yes
> >
> > and using following register string
> >
> > register  => abc:mysecret at 192.168.0.20 <abc%3Amysecret at 192.168.0.20> <
> abc%3Amysecret at 192.168.0.20 <abc%253Amysecret at 192.168.0.20>><mailto:
> abc%3Amysecret at 192.168.0.20 <abc%253Amysecret at 192.168.0.20> <
> abc%253Amysecret at 192.168.0.20 <abc%25253Amysecret at 192.168.0.20>>>
> >
> >
> > now problem is that when i use register string everything goes ok. but
> > when i remove register string call doesn't go as expected.
> >
> > I would like to know if there is any feature that i can use to call sip
> > peer and authenticate is in dial command or some feature in sip.conf
> >
> > i dont wanna use register string. please help.
> >
> > regards,
> >
> > Nasir Javaid
> >
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