[asterisk-users] asterisk-users Digest, Vol 70, Issue 24
Vardan
hvardan71 at gmail.com
Tue May 11 10:54:30 CDT 2010
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
Nasir Javaid wrote:
> 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 ${EXTEN}.
>
> but when i change system 1 (192.168.0.20) to my real server (e.g.
> nasir.server.com <http://nasir.server.com>) which has abc as user
> configured same as on system1 (192.168.0.20), call goes to [default]
> instead of going to [payasyougo] context and is treated as incoming call...
>
> when we use register string calls works ok on real server too. I also tried
>
> SIP/abc:mysecret/${EXTEN} and
> SIP/${EXTEN}@abc:mysecret
>
> but nothing seems to work.
>
> there is another problem that sometime my real server (nasir.server.com
> <http://nasir.server.com>) becomes unreachable and this error is returned
>
> NOTICE[3898]: chan_sip.c:11489 sip_reg_timeout: -- Registration for
> 'abc at nasir.server.com <mailto:abc at nasir.server.com>' timed out, trying
> again (Attempt #38)
>
> It may be a simple problem but is driving my crazy... please help me out
>
> thanks in advance
>
> Nasir Javaid
>
>
> Message: 6
> Date: Tue, 11 May 2010 13:57:23 +0500
> From: Nasir Javaid <nasirjavaidnasir at gmail.com
> <mailto:nasirjavaidnasir at gmail.com>>
> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23
> To: asterisk-users at lists.digium.com
> <mailto:asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTim3qsPCY0sY3hanDxtTg8HPbXJEmj7HWZSknO1W at mail.gmail.com
> <mailto:AANLkTim3qsPCY0sY3hanDxtTg8HPbXJEmj7HWZSknO1W at mail.gmail.com>>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanks Vardan,
>
> I will like to know if this scenario can work when peer is not
> having fixed
> ip and we use
> host = nasir.server.com <http://nasir.server.com>
> ?
> also I have set insecure=invite,port
>
> what if i use
> insecure=no
>
> thanks again.
>
> Message: 24
> Date: Tue, 11 May 2010 10:52:14 +0500
> From: Vardan <hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>>
> Subject: Re: [asterisk-users] Dialing a SIP Peer without using
> register strin
> To: asterisk-users at lists.digium.com
> <mailto:asterisk-users at lists.digium.com>
> Message-ID: <hsarab$ok7$1 at dough.gmane.org <mailto: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
> <mailto:abc%3Amysecret at 192.168.0.20> <abc%3Amysecret at 192.168.0.20
> <mailto:abc%253Amysecret at 192.168.0.20>><mailto:
> abc%3Amysecret at 192.168.0.20 <mailto:abc%253Amysecret at 192.168.0.20>
> <abc%253Amysecret at 192.168.0.20
> <mailto: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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