SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226974 f38db490-d61c-443f-a65b-d21fe96a405b
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@ -929,6 +929,8 @@ static const struct cfsip_methods {
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{ SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
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};
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static unsigned int chan_idx;
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/*! Define SIP option tags, used in Require: and Supported: headers
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We need to be aware of these properties in the phones to use
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the replace: header. We should not do that without knowing
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@ -6938,8 +6940,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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sip_pvt_unlock(i);
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/* Don't hold a sip pvt lock while we allocate a channel */
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tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, linkedid, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i);
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tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, linkedid, i->amaflags, "SIP/%s-%08x", my_name, ast_atomic_fetchadd_int((int *)&chan_idx, +1));
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}
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if (!tmp) {
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ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
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