Remco Barendse wrote:
Hi all!!
I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the
other box keeps repeating :
[Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
When i set debug on, it seems to come from that SIP DID.
<--- SIP read from 82.101.62.99:5060 --->
Cirpack KeepAlive Packet
<------------->
[Feb 11 23:37:59] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---
What i don't understand is why i get this message on one box only?
Ideas anyone?
Not sure. Difference in "core set verbose"? Afaik it's harmless but
quite annoying. Attached are two patches. One should "fix" the keep
alive stuff by silently dropping it and the other is a dtmf fix cause
Cirpack has a bug with dtmf. At least it did a while ago. Not sure if
it's still needed.
Regards,
Patrick
diff -uNr asterisk-1.4.13.org/channels/chan_sip.c asterisk-1.4.13/channels/chan_sip.c
--- asterisk-1.4.13.org/channels/chan_sip.c 2007-10-10 16:42:00.000000000 +0200
+++ asterisk-1.4.13/channels/chan_sip.c 2007-11-14 04:33:05.000000000 +0100
@@ -6620,6 +6620,12 @@
if (*e)
*e++ = '\0';
e = ast_skip_blanks(e);
+ if (!strcasecmp(req->rlPart1, "Cirpack") &&
+ !strcasecmp(req->rlPart2, "KeepAlive") &&
+ !strcasecmp(e, "Packet")) {
+ /* Silently drop bogus Cirpack keepalive packets */
+ return -1;
+ }
if (strcasecmp(e, "SIP/2.0") ) {
ast_log(LOG_WARNING, "Bad request protocol %s\n", e);
return -1;
diff -uNr asterisk-1.4.13.org/main/rtp.c asterisk-1.4.13/main/rtp.c
--- asterisk-1.4.13.org/main/rtp.c 2007-10-08 22:06:33.000000000 +0200
+++ asterisk-1.4.13/main/rtp.c 2007-11-11 13:12:28.000000000 +0100
@@ -1383,6 +1383,7 @@
[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
+ [96] = {0, AST_RTP_DTMF},
[99] = {1, AST_FORMAT_H264},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},
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