[asterisk-ss7] My experiences with chan_ss7, some questions and a solution for the ringback tone

Jacob Tinning tinning at sifira.dk
Wed Mar 15 08:27:34 MST 2006


On Wed, 15 Mar 2006, Kai Militzer wrote:

> I tought I could share my experience with chan_ss7 with you and maybe
> get some answers/opinions from the rest of you.

> So I added a timer that waits for a SAM after an IAM and starts
> again if a SAM is received. In my opinion this is the only way to use
> chan_ss7 as a gateway without knowledge of the numberingplan on the
> final destination. Sifira didn't see it this way and wouldn't take my
> patch into the main chan_ss7 ;( , maybe some of you could convince them
> to do so. ;)

We didn't like the timer-solution because we think its wrong to delay all calls
X seconds just because the SS7-asterisk doesn't know another Asterisk's dialplan.

My suggestions is
 1. Use identical dialplans on the SS7-gateway and the SIP server
 2. Store the dialplan in a shared database.
 3. I think it is (maybe) posible to 'share' the dialplan through IAX (anybody ?)

> Another problem I had was with the handling of the hangupcause which
> weren't translated correctly from SS7 to SIP and other way round. In my
> opinion the error was in ast_softhangup_nolock in asterisk, but seems
> not to be the case (see http://bugs.digium.com/view.php?id=6550).
> @sifira, if you are reading this: would it be possible to fix this in
> chan_ss7?

We well look into this. Thank you for noticing.

> Now my question to the comunity: Is anyone running asterisk with
> chan_ss7 as PSTN-to-SIP Gateway anf if yes what are your experiences?
> Does it work reliable, what call volumes do you handle with it?

We at Sifira are also very interested in hearing any experiences :)

> And last but not least, I also had the problem that no ringback tones
> were generated by asterisk. The following two lines in the dialplan
> inserted before the Dial statement do the trick:

> exten => _X.,n,SetLanguage(de)
> exten => _X.,n,Playtones(ring)

We actually tried this, but we had to insert a ,1,Answer before the Playtones command.
...but the Answer before Playtones, breaks most telcos billing system,
since a call is 'from the Answer to a hangup'.

Anyway, we are working at a solution which will get chan_ss7 & Asterisk to
generate indication-tones before the ANM has been sent to the remote switch.

Mvh. Jacob

-- 
Jacob Tinning
System Developer                                           SIFIRA A/S



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