Re: [Asterisk-Users] siemens optipoint 400 SIP

[Wilhelm Wimmreuter]

Thomas,

On Fri, 2003-05-30 at 08:22, Tomaz Izanc wrote:
> hi!
> 
> anyone try  siemens optipoint 400 economy SIP phone with * ?
> 
Yes, it works pretty well and has message waiting indication.
and has rfc2833 if you apply a workaround.


But you need:

- Patch to replay contact address as is
  * usually replies his own address in contact header
... this shall be corrected by Asterisk
  

- Patch that provides a workaround for message waiting
  indication in rtp.c. This patch is just copied from a
  previous workaround for Cisco.

  This seems to be a problem of the OptiPoint 400.
  ... Siemens currently looks at the issue and
  may has a solution soon.
  

in sip.conf it looks like:

[3999]
type=friend
username=3999
host=dynamic
dtmfmode=rfc2833
callerid="3999" <3999>
mailbox=3999
context=SIPorig

BTWY:
 What version has your Optipoint SIP-SW?
...mine is 1.1.4; I'm waiting for 2.0

you may call me at:

SIP:[EMAIL PROTECTED]
SIP:[EMAIL PROTECTED]

Willi

--- ./a-cvs/asterisk/channels/chan_sip.c2003-05-23 17:12:08.0 +0200
+++ ./a-wrk/asterisk/channels/chan_sip.c2003-05-24 13:41:21.0 +0200
@@ -1901,7 +1901,11 @@
snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, 
p->expiry);
snprintf(tmp, sizeof(tmp), "%d", p->expiry);
add_header(resp, "Expires", tmp);
+   /*ww lwc change header to copy
add_header(resp, "Contact", contact);
+   */
+   copy_header(resp, req, "Contact");
+
} else {
add_header(resp, "Contact", p->our_contact);
}
@@ -2964,6 +2968,8 @@
char resp_hash[256];
char tmp[256] = "";
char *c;
+   /*ww need to check authorization headers for 'space' and ',' *z */
+   char *z; 
char *response ="";
char *resp_uri ="";
 
@@ -3001,7 +3007,10 @@
}
 
} else
+   /*ww we need to check for spaces as well   
 
c = strchr(c, ',');
+   */
+   if ((z = strchr(c,' ')) || (z = strchr(c,','))) c=z;
if (c)
c++;
}
--- ./a-cvs/asterisk/rtp.c  2003-05-16 04:50:46.0 +0200
+++ ./a-wrk/asterisk/rtp.c  2003-05-24 18:23:36.0 +0200
@@ -326,6 +326,12 @@
/* Comfort Noise */
f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res 
- hdrlen);
if (f) return f; else return &null_frame;
+ /* OP400 payloadtype 100 */
+ } else if (payloadtype == 100) {
+   /* Comfort Noise */
+   f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res 
- hdrlen);
+   if (f) return f; else return &null_frame;
+  /* end OP400 */
  } else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
return &null_frame;

Re: [Asterisk-Users] Opportunistic VoIP

[Wilhelm Wimmreuter]

John,

On Wed, 2003-06-11 at 21:58, John Todd wrote:
> >This is slightly off-topic I suppose, but:
> 
> 2) The ENUM system is centralized.  TRIP can be established between 
> two telephone systems, independently of any third party's cooperation 
> or assistance.  Routes can be exchanged in any way that is acceptable 
> to those two systems.
> 

 I belive, that his is exactly the problem of TRIP.
TRIP needs peering agreements between all providers.
Compared to ENUM that can be queried without having
a pre-established contract between all operators.
This applies for queries as well as for providioning
of the repository. Besides this, TRIP is way 
overloaded with attributes no one needs at the end.

 My conclusion:
TRIP breaks because it is not open to all parties.


and this goes well along with Florian Overkamp's
statement:

>> TRIP only makes it harder for widespread use to deal with
>> such things as number portability (can't ever do that with
>> IP, remember). As far as I can tell from the TRIP docs
>> this looks a lot like some big telco tries to make it more
>> difficult for customers to move to another telco and still
>> use their old number...


Willi


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