asterisk users - Jul 2004 - Ringinbacktone even without 'r', and inexistant codec

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with
Asterisk. It works ... Partially.

We are using the Inalp to connect ISDN phones, it basically acts like an ISDN
ATA.

First of all, when I make a SIP call to the unit with a simple Dial() command
(no "r", so Asterisk shouldn't generate its ringback tone) I hear
Asterisk's ringback tone anyway (I'm sure it's Asterisk generating
it because by changing the country it indications.conf, the ringing changes).
That's what I see in the CLI:

    -- Executing Dial("SIP/2017-71be", "SIP/070@inalp|90")
in new stack
    -- Called 070@inalp
    -- SIP/inalp-eaf3 is making progress passing it to SIP/2017-71be
    -- SIP/inalp-eaf3 is ringing

Now comes the fun part: if the ISDN extension answers the phone, the call is
dropped, and I get the following message:

    -- SIP/inalp-eaf3 answered SIP/2017-71be
    -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3
    -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3
Jul  7 20:36:20 WARNING[112708528]: chan_sip.c:1800 sip_write: Asked to transmit
frame type 64, while native formats is 4 (read/write = 8/4)

Now, frame type 64 is "16 bit signed linear PCM", which IMHO has
nothing to do with SIP and its RTP stream. I have the usual "disallow=all;
allow=ulaw; allow=alaw" sequence in sip.conf, and the Inalp unit is
configured to allow alaw and ulaw, nothing else (it doesn't even support
that 16 bit PCM thing).

But we're not through yet. If I add the "r" paramenter to the
Dial() command, the call completes successfully. But unfortunately, now Asterisk
doesn't (!) generate the ringback tone anymore. I get no ringing at all,
just silence, until the other party answers.

Isn't * supposed to generate a ringback tone when "r" is appended
in the Dial command? Isn't * supposed *not* to generate a ringback tone when
there is *no* "r"? What in the world is codec 64?

By the way, outgoing (ISDN-to-SIP) calls from the Inalp unit work perfectly.
Other SIP clients (ATAs, softphones) work perfectly on our setup (and also the
ringback tone behaviour is correct with those). Only that single unit (the most
expensive one, by the way :-)) doesn't want to cooperate.

I'm clueless here... Anyone?

Thanks
-Manuel


___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com

This was fixed today.. update
bkw
----- Original Message ----- 
From: "Manuel Wenger" <manuel.wenger@ticinocom.com>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, July 07, 2004 1:44 PM
Subject: [Asterisk-Users] Ringinbacktone even without 'r', and
inexistant
codec

> I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work
with Asterisk. It works ... Partially.
>
> We are using the Inalp to connect ISDN phones, it basically acts like an
ISDN ATA.
>
> First of all, when I make a SIP call to the unit with a simple Dial()
command (no "r", so Asterisk shouldn't generate its ringback tone)
I hear
Asterisk's ringback tone anyway (I'm sure it's Asterisk generating
it
because by changing the country it indications.conf, the ringing changes).
That's what I see in the CLI:
>
>     -- Executing Dial("SIP/2017-71be",
"SIP/070@inalp|90") in new stack
>     -- Called 070@inalp
>     -- SIP/inalp-eaf3 is making progress passing it to SIP/2017-71be
>     -- SIP/inalp-eaf3 is ringing
>
> Now comes the fun part: if the ISDN extension answers the phone, the call
is dropped, and I get the following message:
>
>     -- SIP/inalp-eaf3 answered SIP/2017-71be
>     -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3
>     -- Attempting native bridge of SIP/2017-71be and SIP/inalp-eaf3
> Jul  7 20:36:20 WARNING[112708528]: chan_sip.c:1800 sip_write: Asked to
transmit frame type 64, while native formats is 4 (read/write =
8/4)
>
> Now, frame type 64 is "16 bit signed linear PCM", which IMHO has
nothing
to do with SIP and its RTP stream. I have the usual "disallow=all;
allow=ulaw; allow=alaw" sequence in sip.conf, and the Inalp unit is
configured to allow alaw and ulaw, nothing else (it doesn't even support
that 16 bit PCM thing).
>
> But we're not through yet. If I add the "r" paramenter to the
Dial()
command, the call completes successfully. But unfortunately, now Asterisk
doesn't (!) generate the ringback tone anymore. I get no ringing at all,
just silence, until the other party answers.
>
> Isn't * supposed to generate a ringback tone when "r" is
appended in the
Dial command? Isn't * supposed *not* to generate a ringback tone when there
is *no* "r"? What in the world is codec 64?
>
> By the way, outgoing (ISDN-to-SIP) calls from the Inalp unit work
perfectly. Other SIP clients (ATAs, softphones) work perfectly on our setup
(and also the ringback tone behaviour is correct with those). Only that
single unit (the most expensive one, by the way :-)) doesn't want to
cooperate.
>
> I'm clueless here... Anyone?
>
> Thanks
> -Manuel
>
>
> ___________________________________________________
> Ticinocom SA - Via Stazione 5 - 6600 Muralto
> Tel 0844 007070 - Fax 0844 007071
> http://www.ticinocom.com
>
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