asterisk users - Oct 2004 - new here : logic of ser and asterisk all confused---longish

Hi

I have managed to get SER up and running fine, and got my UA (Xlite and
IPphone) talking to each other, all authenticating nicely from the
DB....and then I started to look at asterisk, I figured I may as well
split the proxy (ser) and do the call features like voicemail and all
using asterisk.

This is where my logic went a bit pear shaped.

I installed asterisk, its on the same box as ser, but on a different port

SER port = 5060
Asterisk port =5090

before I tested ser and asterisk together I tested asterisk on its own,
dialed 1000 got the voicemail, so I figured all is good, only after that
did I change the ports to 5090, and fored up SER again.

Now from my understanding SER is a proxy, hence will allow calls just to
pass through, and the media stream doesnt neccessarily have to go
through it..

What I wanted to do, was to initiate a call using Xlite get SER to
register/authenticate it via my DB, and then pass the call out to
destination via asterisk,the call destination could be another SIP UA
(Xlite), or pstn , the latter is not implemented as yet (on a side note
can I instead of installing cards, just pass this pstn traffic to a
global terminator, if so how, beit H323 or SIP, especially interested if
anyone has done this with quintum A800)

However what i have managed to somehow cobble together is this, and it
seems to work....

Xlite ---> Asterisk(5090)--->sip.conf ----> extensions --->SER
---->
forward statement in SER ----> Xlite

what I need is

Xlite ----> SER ----use forward statement in ser.cfg ----> Asterisk
---->sip.conf---->extensions.conf ---> PSTN or another xlite

Now is the second one possible or does it have to go back via SER, also
what about the billing, is their any problems if ser is not passing the
stream (confused at this point :-))

ser.cfg
-----
if (uri =~ "sip:2[0-9]*@sip.ipclouds.co.uk"){

        log(1, "Forwarding to Asterisk\n");
        rewritehostport("193.218.160.25:5090");

        break;
    }

which I think means any number starting with a 2 send to asterisk
server..now when I dial this , in the SER logs it shows the message
Forwarding to Asterisk, and then waits, but in asterisk sip debug there
is nothing, not a sausage

sip.conf
-----
[general]

context=OUTGOING
autocreatepeer=yes

this to me suggests to lookup contect OUTGOING in extensions file so....

extensions.conf
-------

[globals]
(snipped out the common ones)

SERADDRESS=sip.ipclouds.co.uk:5060

[OUTGOING]
; Line below added for ser --- iqbal
exten => 1000,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)

seeing all this it would seems that asterisk and ser go into a loop,
cause extensions simply sends it back to SER, which is what seems to
happen, and the ser.cfg sends it back to extensions.

-------
Ser tries to send to asterisk from my logs, but alas asterisk does not
want to take the call and make an outbound.

hope it wasnt too looong

tks

iqbal

Hello,

comments inline...

--- Iqbal <iqbal@gigo.co.uk> wrote:
> if (uri =~ "sip:2[0-9]*@sip.ipclouds.co.uk"){
> 
>         log(1, "Forwarding to Asterisk\n");
>         rewritehostport("193.218.160.25:5090");
> 
>         break;
>     }
> 
> which I think means any number starting with a 2 send to asterisk
> server..now when I dial this , in the SER logs it shows the message
> Forwarding to Asterisk, and then waits, but in asterisk sip debug there
> is nothing, not a sausage
I am afraid you are not sending the calls to Asterisk, but just rewriting the
host and port. After
rewriting, forward/relay the calls to Asterisk.
> SERADDRESS=sip.ipclouds.co.uk:5060
> 
> [OUTGOING]
> ; Line below added for ser --- iqbal
> exten => 1000,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r)
> 
> seeing all this it would seems that asterisk and ser go into a loop,
> cause extensions simply sends it back to SER, which is what seems to
> happen, and the ser.cfg sends it back to extensions.
Dont Asterisk complain about a '482 Loop Detected' error? The Dial
statement will create a new
INVITE and will be relayed to SER, which will send it back to Asterisk, thus
resulting in a loop.
Asterisk will drop this call. For dialing extensions use either Asterisk or SER.
IMO, use ser for
all extension dialing, and have appropriate forwarding and failure routing in
the ser.cfg to send
calls to Asterisk for the PBX features and voicemail.

Regards, Girish



		
_______________________________
Do you Yahoo!?
Declare Yourself - Register online to vote today!
http://vote.yahoo.com