[Asterisk-Users] new here : logic of ser and asterisk allconfused---longish

[Iqbal]

Hi

will check the relay part, as for the loop not happening I pasted wrong
lune, basically where the EXTEN part is I had hardcoded a username which
existed in my DB, and that got now loop...but will check again.

What I am trying to set up, is to get SER to register all the calls, and
pass these all to asterisk, and do the same with pstn /voicemail etc
calls, and let asterisk take care of this.

According to your suggestion I should do the IP-->IP calls on SER itself
(am I reading it correct), and only pass the rest on.

If this is done, then how can I use asterisk billing software for the
IP--->IP calls, or will I have ot bill that separate, I am moving the
calls to asterisk for two reason one for scaling and two for easier
billling, since asterisk comes with a nice array of prepaid billing
options, however I dont want to run one with ser and one with asterisk.

I just re-read what u said,, u mean pass all the calls from ser to
asterisk, and the if its a extension, pass it back to ser and let it
dial, if they are not there etc, it should fall to asterisk for
voicemail, and pstn comes to asterisk.

Let me get the call flow correct

Xlite(1) --> ser ---> asterisk (if call for IP device) ---> ser --->
Xlite(2)
Xlite(1) --> ser ---> asterisk (if call for pstn) ---> pstn
Xlite(1) --> ser ---> asterisk (if call for voicemail) ---> Voicemail


So what would happen if call for pstn, and call not answered, and also
what happens if SIp end device (eh Xlite(2))  does not answer can I get
the call go back from ser to asterisk for voicemail

tks

iqbal

On 10/20/2004, "Asterisk ." <asterisk_in at yahoo.com> wrote:

>Hello,
>
>comments inline...
>
>--- Iqbal <iqbal at 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
>
>
>
>
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