[Asterisk-Users] Ringinbacktone even without 'r', and inexistant codec

[Manuel Wenger]

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 at inalp|90") in new stack
    -- Called 070 at 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


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