From 175dd0ebf628e7ac4770fc2b2fe3a0c5a130753c Mon Sep 17 00:00:00 2001
From: Andrew Latham <lathama@gmail.com>
Date: Wed, 2 Feb 2011 15:25:12 +0000
Subject: [PATCH] Replace link to old doc with new wiki page.
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@305759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a33a0ddd1c..760c2ce818 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3417,7 +3417,7 @@ static int retrans_pkt(const void *data)
if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
if (pkt->is_fatal || sipdebug) { /* Tell us if it's critical or if we're debugging */
- ast_log(LOG_WARNING, "Retransmission timeout reached on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n"
+ ast_log(LOG_WARNING, "Retransmission timeout reached on transmission %s for seqno %d (%s %s) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n"
"Packet timed out after %dms with no response\n",
pkt->owner->callid,
pkt->seqno,
@@ -3426,7 +3426,7 @@ static int retrans_pkt(const void *data)
(int) ast_tvdiff_ms(ast_tvnow(), pkt->time_sent));
}
} else if (pkt->method == SIP_OPTIONS && sipdebug) {
- ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid);
+ ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n", pkt->owner->callid);
}
if (xmitres == XMIT_ERROR) {
@@ -3446,7 +3446,7 @@ static int retrans_pkt(const void *data)
pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE;
}
if (pkt->owner->owner) {
- ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid);
+ ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).\n", pkt->owner->callid);
if (pkt->is_resp &&
(pkt->response_code >= 200) &&
--
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