[Asterisk-Dev] chan_sip

Jamie Carl asterisk at jazz-inc.net
Tue Mar 11 19:54:24 MST 2003

Your 415 No Media, (or Unsupported Media Type) comes from there  not being a
common voice codec available for the two endpoints.  As far as I can tell,
asterisk only supports PCM (alaw, mulaw) and GSM.  Now most devices won't
support GSM and maybe your provider will not allow PCM cause it uses too
much bandwidth.  Which leaves you with nothing pretty much.

I can't see a solution for this, apart from getting either your provider to
allow PCM, which is a bandwidth hog, or GSM, which would be ideal.  Or
alternative whinge to Mark about getting other codecs available in Asterisk.


Jamie Carl
Email:    me at jazz-inc.net
Web:    www.jazz-inc.net
PH:        +61-414-365-466

----- Original Message -----
From: "Nickolay Shestakov" <npshe at mail.ru>
To: <asterisk-dev at lists.digium.com>
Sent: Wednesday, March 12, 2003 12:40 PM
Subject: [Asterisk-Dev] chan_sip

> Hi, all
> I'm the beginner in Asterisk. Sorry, if my questions are already
> The first questions about chan_sip:
> 1. When * registers with SIP provider, I see the following messages:
> 401 Unauthorized
> 200 OK
> ACK (From * ) - It seems to me, that this ACK is superfluous. So, may be,
> just delete  transmit_request(p, "ACK", 0) after receiving 200 OK ?
> And I noticed, that the reactions of SIP providers are different (after
> ACK). FWD sends 200 OK, but my SIP provider (www.voipexchange.ru) returns
> another ACK to me.
> 2. Registration with SIP provider is ok. But, when * sends INVITE, my SIP
> privider returns 401 Unauthorized. And, of course, after that * crashes
> segmentation fault, because, in function handle_response
> case 401:
> do_register_auth...
> May be, the matter is, that in RFC 3261:
> "The branch parameter value MUST be unique across space and time for
>    all requests sent by the UA"
> But, I noticed, that branch is different when * registers and when it
> INVITE. I didn't dig deeper, just deleted parameter branch from via
> works without it, and now it works (excluding, that now my provider
> 415 No media - I don't know  why.. :)
> Regards,
> Nickolay
> npshe at mail.ru
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev

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