Posted: Wed Mar 12, 2003 1:40 am Post subject: [Asterisk-Dev] chan_sip
Hi, all
I'm the beginner in Asterisk. Sorry, if my questions are already discussed.
The first questions about chan_sip:
1. When * registers with SIP provider, I see the following messages:
REGISTER
401 Unauthorized
REGISTER
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 this
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 with
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 sends
INVITE. I didn't dig deeper, just deleted parameter branch from via (ata-186
works without it, and now it works (excluding, that now my provider returns
415 No media - I don't know why.. :)
Posted: Wed Mar 12, 2003 2:16 am Post subject: [Asterisk-Dev] chan_sip
Quote:
1. When * registers with SIP provider, I see the following messages:
REGISTER 401 Unauthorized REGISTER 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 ?
Yes, I believe the ACK is superfluous and I've removed it from CVS.
Quote:
2. Registration with SIP provider is ok. But, when * sends INVITE, my SIP
privider returns 401 Unauthorized. And, of course, after that * crashes with
segmentation fault, because, in function handle_response
case 401:
do_register_auth...
I believe the use of &digest was a programming error on the proxy auth
patch for registration. I think it should work properly now.
Quote:
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 sends
INVITE. I didn't dig deeper, just deleted parameter branch from via (ata-186
works without it, and now it works (excluding, that now my provider returns
415 No media - I don't know why.. :)
I don't think I see your complaint. If it has to be unique for all
requests, does that mean that all requests should have the *same* branch
or that all requests should have *different* branches?
Posted: Wed Mar 12, 2003 2:54 am Post subject: [Asterisk-Dev] chan_sip
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.
----- Original Message -----
From: "Nickolay Shestakov" <npshe@mail.ru>
To: <asterisk-dev@lists.digium.com>
Sent: Wednesday, March 12, 2003 12:40 PM
Subject: [Asterisk-Dev] chan_sip
Quote:
Hi, all
I'm the beginner in Asterisk. Sorry, if my questions are already
discussed.
Quote:
The first questions about chan_sip:
1. When * registers with SIP provider, I see the following messages:
REGISTER
401 Unauthorized
REGISTER
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
this
Quote:
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
with
Quote:
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
sends
Quote:
INVITE. I didn't dig deeper, just deleted parameter branch from via
(ata-186
Quote:
works without it, and now it works (excluding, that now my provider
returns
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