Actually,
I believe that something is wrong with the way asterisk implements the
whole rfc2833 in rtp.c , moreover, the default value of 100ms in
dtmf_tones[] in do_senddigit() inchannel.c is to short to be detected
for lots of commercially available fxo gateways.
This was reported several times but as of today the issue is there.
I ended up with the following ugly hack for rtp.c & channel.c:
--- work/asterisk-1.2.4/rtp.c Wed Mar 1 20:25:03 2006
+++ /root/cvs/rtp.c Tue Feb 21 00:05:55 2006
@@ -1150,7 +1150,7 @@
rtpheader[1] = htonl(rtp->lastdigitts);
rtpheader[2] = htonl(rtp->ssrc);
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
- for (x = 0; x < 6; x++) {
+ for (x = 0; x < 15; x++) {
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
res = sendto(rtp->s, (void *) rtpheader, hdrlen
+ 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
if (res < 0)
@@ -1163,12 +1163,12 @@
ntohs(rtp->them.sin_port),
payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
}
/* Sequence number of last two end packets does not get
incremented */
- if (x < 3)
+ if (x < 12)
rtp->seqno++;
/* Clear marker bit and set seqno */
rtpheader[0] = htonl((2 << 30) | (payload << 16) |
(rtp->seqno));
/* For the last three packets, set the duration and the
end bit */
- if (x == 2) {
+ if (x == 11) {
#if 0
/* No, this is wrong... Do not increment
lastdigitts, that's not according
to the RFC, as best we can determine */
--- work/asterisk-1.2.4/channel.c Wed Mar 1 20:25:01 2006
+++ /root/cvs/channel.c Tue Feb 21 00:05:50 2006
@@ -2111,22 +2111,22 @@
* it by doing our own generation. (PM2002)
*/
static const char* dtmf_tones[] = {
- "!941+1336/100,!0/100", /* 0 */
- "!697+1209/100,!0/100", /* 1 */
- "!697+1336/100,!0/100", /* 2 */
- "!697+1477/100,!0/100", /* 3 */
- "!770+1209/100,!0/100", /* 4 */
- "!770+1336/100,!0/100", /* 5 */
- "!770+1477/100,!0/100", /* 6 */
- "!852+1209/100,!0/100", /* 7 */
- "!852+1336/100,!0/100", /* 8 */
- "!852+1477/100,!0/100", /* 9 */
- "!697+1633/100,!0/100", /* A */
- "!770+1633/100,!0/100", /* B */
- "!852+1633/100,!0/100", /* C */
- "!941+1633/100,!0/100", /* D */
- "!941+1209/100,!0/100", /* * */
- "!941+1477/100,!0/100" }; /* # */
+ "!941+1336/250,!0/100", /* 0 */
+ "!697+1209/250,!0/100", /* 1 */
+ "!697+1336/250,!0/100", /* 2 */
+ "!697+1477/250,!0/100", /* 3 */
+ "!770+1209/250,!0/100", /* 4 */
+ "!770+1336/250,!0/100", /* 5 */
+ "!770+1477/250,!0/100", /* 6 */
+ "!852+1209/250,!0/100", /* 7 */
+ "!852+1336/250,!0/100", /* 8 */
+ "!852+1477/250,!0/100", /* 9 */
+ "!697+1633/250,!0/100", /* A */
+ "!770+1633/250,!0/100", /* B */
+ "!852+1633/250,!0/100", /* C */
+ "!941+1633/250,!0/100", /* D */
+ "!941+1209/250,!0/100", /* * */
+ "!941+1477/250,!0/100" }; /* # */
if (digit >= '0' && digit <='9')
ast_playtones_start(chan, 0,
dtmf_tones[digit-'0'], 0);
else if (digit >= 'A' && digit <= 'D')
Arsen Chaloyan wrote:
Vahan,
see my comments below.
[snip]
>
Actually SPA-3000 should receive rfc2833 DTMF events
from IP side and put inband DTMFs instead to PSTN
side.
SPA-3000 fails to correctly detect rfc2833 events,
which poorly constructed by asterisk.
So asterisk isn't fully compliant to rfc2833, from the
other hand other VoIP gateways (but not SPA-3000)
manage to detect rfc2833 DTMFs from asterisk.
SPA-3000 serves well in this scenario if source of RTP
stream is another VoIP device or phone (cisco, snom,
aastra...), not asterisk itself.
Here come the solution:
try to exclude asterisk from media path
canreinvite=yes
[111]
;SPA-3000 FXO port
type=friend
username=111
secret=xxx
host=dynamic
canreinvite=yes
dtmfmode=rfc2833
This is the only way I manage to setup
asterisk/SPA-3000 in described scenario.
Hope this will help,
Arsen.
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research & Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
end:vcard
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