[asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP

[tracinet]

Basically, Server 1 is the main customer PBX where we have multiple
customers running (each on their own virtual PBX separated by their
contexts).  Each customer has their own accountcode that we use to track
calls for billing purposes, etc.  The customer uses a SIP phone to register
to Server 1 and sends calls to it.  Server 1 in turn, passes the calls to
Server 2 which is connected to various SIP providers and T-1's, etc. for
termination to the PSTN.   In the following sip configuration, calls work
perfectly, except that the caller ID gets passed as the value from
"fromuser" instead of the numeric value we set via the
Set(CALLERID(num)=5555555555) command.  In other words, the fromuser
overrides the caller ID value.  If we remove the "fromuser" in the sip
configuration, calls work great and caller ID is passed, BUT all calls land
in the customerb context on Server 2 since that is the last SIP entry in
sip.conf that has a host entry set to "192.168.0.11" which is the IP of
Server 1.

Server 1 (192.168.0.11)

sip.conf

[general]
disallow = all
allow = ulaw
port = 5060
context = incoming
maxexpirey=3600
defaultexpirey=300
canreinvite=no
dtmfmode=auto
nat=yes

; Customer A Outbound SIP
[customera-out]
context=customera
type=friend
username=customera-out
fromuser=customera-out
secret=aaaa
host=192.168.0.12
canreinvite=no
accountcode=customera
amaflags=billing
dtmfmode=auto

; Customer A SIP Phone Account
[customera101]
context=customera
type=friend
username=customera101
secret=1234
host=dynamic
canreinvite=no
mailbox=101 at customera
nat=yes
qualify=yes
callerid="John Smith" <101>
accountcode=customera
amaflags=billing
dtmfmode=rfc2833

; Customer B Outbound SIP
[customerb-out]
context=customerb
type=friend
username=customerb-out
fromuser=customerb-out
secret=bbbb
host=192.168.0.12
canreinvite=no
accountcode=customerb
amaflags=billing
dtmfmode=auto

; Customer B SIP Phone Account
[customerb101]
context=customerb
type=friend
username=customerb101
secret=1234
host=dynamic
canreinvite=no
mailbox=101 at customerb
nat=yes
qualify=yes
callerid="Jane Jones" <101>
accountcode=customerb
amaflags=billing
dtmfmode=rfc2833




Server 2 (192.168.0.12)

sip.conf:

[general]
disallow = all
allow=ulaw
port = 5060
context = incoming
canreinvite=no
nat=no
dtmfmode=auto

[customera-out]
context=customera
type=friend
username=customera-out
secret=aaaa
host=192.168.0.11
accountcode=customera
amaflags=billing
dtmfmode=auto

[customerb-out]
context=customerb
type=friend
username=customerb-out
secret=bbbb
host=192.168.0.11
accountcode=customerb
amaflags=billing
dtmfmode=auto








On Fri, Mar 6, 2009 at 2:48 PM, Steve Howes <steve at geekinter.net> wrote:

>
> On 6 Mar 2009, at 19:29, tracinet wrote:
>
> > That stinks... We are migrating to SIP from IAX2 at the moment and
> > running into the same exact problem.  No way to control the
> > destination context unless you use the "fromuser".  Of course that
> > is rendering Caller ID useless as you pointed out.
>
> Give me the exact sip.conf you have both ends. Might be able to get it
> working.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090306/20be28f4/attachment.htm