I've put in the patch "by hand", thus:
if (start && (end =
strrchr(appl, ')'))) {
*start =
*end = '\0';
data = start
+ 1;
process_quotes_and_slashes(data,
',', '|');
} else if
(stringp!=NULL && *stringp=='"') {
stringp++;
data =
strsep(&stringp, "\"");
stringp++;
} else {
if (stringp)
data
= strsep(&stringp, ",");
else
data
= "";
}
#if 0
pbx_substitute_variables_helper(NULL,
ext, realext, sizeof(realext)-1);
cidmatch =
strchr(ext, '/');
if (cidmatch) {
*cidmatch =
'\0';
cidmatch++;
}
stringp=ext;
strsep(&stringp,
"/");
#endif
#if 1
pbx_substitute_variables_helper(NULL,
ext, realext, sizeof(realext)-1);
stringp = realext;
ext =
strsep(&stringp, "/");
cidmatch = stringp;
#endif
if (!data)
data="";
while(*appl &&
(*appl < 33)) appl++;
if (ipri) {
if
(!strcmp(realext, "_."))
ast_log(LOG_WARNING,
"The use of '_.' for an extension is strongly discouraged and c
if
(ast_add_extension2(con, 0, realext, ipri, cidmatch, appl, strdup(data),
FREE, registrar)
ast_log(LOG_WARNING,
"Unable to register extension at line %d\n", v->lineno);
}
}
free(tc);
which is what I think you intended and it still doesn't work for me (yes, I
did stop and restart Asterisk)... I'm in the UK using a cheap X100P clone
and V.23 Caller ID which used to work 100% under 1.0.7 in my extensions.conf
I have a context, thus:
;
; from-pstn : incoming calls from the FXO card from PSTN
;
[from-pstn]
exten => s,1,Answer
exten => s,2,NoOp("CallerIDnum=${CALLERIDNUM} CallerID=${CALLERID}")
exten => s/0,3,Goto(no-callerid,s,1)
; to dedicated lines on 7960s and the 7912s
exten =>
s,3,Dial(SIP/9001&SIP/9002&SIP/2003&SIP/2004&SIP/2005&SIP/2006&IAX2/thorcom/8102001,20,rt)
exten => s,4,Voicemail(u2001)
exten => s,5,HangUp
; okay, they withheld their caller id - play out a "we dont do withheld
callers" and dump them to voicemail
[no-callerid]
exten => s,1,Playback(withheld-callerid)
exten => s,2,Voicemail(su2001)
exten => s,3,Hangup
Now when I get a call from a withheld this happens:
Connected to Asterisk 1.0.8 currently running on gate (pid = 18645)
Verbosity is at least 3
-- Remote UNIX connection
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing NoOp("Zap/1-1", ""CallerIDnum=0 CallerID=Number Witheld
<0>"") in new stack
-- Executing Dial("Zap/1-1",
"SIP/9001&SIP/9002&SIP/2004&SIP/2005&SIP/2006&IAX2/thorcom/8102001|20|rt")
in new stack
-- Called 9001
-- Called 9002
-- Called ... etc. etc.
and all my phones ring :o(
Mike
----- Original Message -----
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Saturday, June 25, 2005 7:33 PM
Subject: Re: [Asterisk-Users] * 1.0.8: no more reacting to callerid?
Daryl Jones wrote:
It's not just you. Same thing happens here. I went back to 1.0.7.
There is definitely breakage in 1.0.8 in this area; please test the patch
below and report back the results here so we can get a new release made.
diff -u -r1.45.2.2 pbx_config.c
--- pbx/pbx_config.c 19 May 2005 02:51:00 -0000 1.45.2.2
+++ pbx/pbx_config.c 25 Jun 2005 17:32:47 -0000
@@ -1687,15 +1687,10 @@
else
data = "";
}
- pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);
- cidmatch =
strchr(ext, '/');
- if (cidmatch) {
- *cidmatch = '\0';
-
cidmatch++;
- }
- stringp=ext;
- strsep(&stringp,
"/");
-
+ pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);
+ stringp = realext;
+ ext =
strsep(&stringp, "/");
+ cidmatch =
stringp;
if (!data)
data="";
while(*appl &&
(*appl < 33)) appl++;
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