Sat Feb 15 07:00:01 CET 2003

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@615 f38db490-d61c-443f-a65b-d21fe96a405b

3 files changed, 16 insertions, 3 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 2d49f1365..b9f235c3c 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1512,7 +1512,7 @@ static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_
 {
 	struct sip_request resp;
 	char tmp[256];
-	snprintf(tmp, sizeof(tmp), "DIGEST realm=\"asterisk\", nonce=\"%s\"", randdata);
+	snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", randdata);
 	respprep(&resp, p, msg, req);
 	add_header(&resp, "Proxy-Authenticate", tmp);
 	add_header(&resp, "Content-Length", "0");
@@ -2517,7 +2517,7 @@ static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header
 
 	strncpy(tmp, get_header(req, header),sizeof(tmp) - 1);
 	c = tmp;
-	c+=strlen("DIGEST ");
+	c+=strlen("Digest ");
 	while (c) {
 		while (*c && (*c < 33)) c++;
 		if (!*c)
@@ -2561,7 +2561,7 @@ static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header
 	snprintf(resp,sizeof(resp),"%s:%s:%s",a1_hash,nonce,a2_hash);
 	md5_hash(resp_hash,resp);
 
-	snprintf(digest,digest_len,"DIGEST username=\"%s\", realm=\"%s\", algorithm=\"MD5\", uri=\"%s\", nonce=\"%s\", response=\"%s\"",p->peername,realm,uri,nonce,resp_hash);
+	snprintf(digest,digest_len,"Digest username=\"%s\", realm=\"%s\", algorithm=\"MD5\", uri=\"%s\", nonce=\"%s\", response=\"%s\"",p->peername,realm,uri,nonce,resp_hash);
 
 	return 0;
 }
diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample
index 8c73d0b6f..e32548795 100755
--- a/configs/zapata.conf.sample
+++ b/configs/zapata.conf.sample
@@ -139,6 +139,15 @@ txgain=0.0
 ;
 group=1
 ;
+; Ring groups (a.k.a. call groups) and pickup groups.  If a phone is ringing
+; and it is a member of a group which is one of your pickup groups, then
+; you can answer it by picking up and dialing *8#.  For simple offices, just
+; make these both the same
+;
+callgroup=1
+pickupgroup=1
+
+;
 ; Specify whether the channel should be answered immediately or
 ; if the simple switch should provide dialtone, read digits, etc.
 ;
diff --git a/sounds.txt b/sounds.txt
index 4d248d540..bb08559bf 100755
--- a/sounds.txt
+++ b/sounds.txt
@@ -32,6 +32,10 @@
 
 %demo-echotest.gsm%You are about to enter an echo test.  In this mode everything you say will be repeated back to you just as soon as it is received.  The purpose of this test is to give you an audible sense of the latency between you and the machine that is running the echo test application.  You may end the test by hanging up or by pressing the pound key.
 
+%demo-enterkeywords.gsm%Please enter one or more keywords separated by * and then press the pound key.
+
+%demo-nomatch.gsm%I'm sorry there are no matches for those keywords
+
 %demo-instruct.gsm%If you would like to learn more technical information about Asterisk dial 2 now.  If you'd like to test out the voice over IP capabilities of Asterisk you can dial 500 to attempt an Inter-Asterisk Exchange or IAX connection to a demonstration server at Digium.  In order for this test to work you will need to be connected to the Internet and have at least a 28.8 kilobit modem.  To execute an echo test dial 600.  This test is most useful when you have connected to this Asterisk server from a remote location.  The sample configuration also has a single user with extension 1234 and password 4242.  That user is configured to ring the console when their extension is dialed.  If you dial 1234 you can try to ring the console.  If the console is busy or unavailable you will be given the option to leave voicemail.  To check voice mail for the user dial extension 8500 to enter the voicemail system.  Finally,  you can press the pound key to disconnect from the PBX.
 
 %demo-moreinfo.gsm%This message has not yet been written.