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automerge commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@13274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -5856,7 +5856,7 @@ static void *ss_thread(void *data)
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number = 0;
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/* If set to use V23 Signalling, launch our FSK gubbins and listen for it */
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} else if (p->cid_signalling == CID_SIG_V23) {
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cs = callerid_new(cid_signalling);
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cs = callerid_new(p->cid_signalling);
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if (cs) {
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samples = 0;
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#if 1
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@@ -6027,7 +6027,7 @@ static void *ss_thread(void *data)
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}
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} else if (p->use_callerid && p->cid_start == CID_START_RING) {
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/* FSK Bell202 callerID */
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cs = callerid_new(cid_signalling);
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cs = callerid_new(p->cid_signalling);
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if (cs) {
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#if 1
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bump_gains(p);
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@@ -86,6 +86,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;progressinband=never ; If we should generate in-band ringing always
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; use 'never' to never use in-band signalling, even in cases
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; where some buggy devices might not render it
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; Valid values: yes, no, never Default: never
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;useragent=Asterisk PBX ; Allows you to change the user agent string
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;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
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; Note that promiscredir when redirects are made to the
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