From 175dd0ebf628e7ac4770fc2b2fe3a0c5a130753c Mon Sep 17 00:00:00 2001 From: Andrew Latham Date: Wed, 2 Feb 2011 15:25:12 +0000 Subject: 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(-) (limited to 'channels') diff --git a/channels/chan_sip.c b/channels/chan_sip.c index a33a0ddd1..760c2ce81 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) && -- cgit v1.2.3