Always set cause_sip and reason_sip in channel params when processing the sip transaction.
git-svn-id: http://voip.null.ro/svn/yate@5663 acf43c95-373e-0410-b603-e72c3f656dc1
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@ -6167,10 +6167,19 @@ bool YateSIPConnection::process(SIPEvent* ev)
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m_byebye = false;
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m_byebye = false;
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paramMutex().lock();
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paramMutex().lock();
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parameters().setParam("cause_sip","408");
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parameters().setParam("cause_sip","408");
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parameters().clearParam("reason_sip");
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paramMutex().unlock();
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paramMutex().unlock();
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setReason("Request Timeout",code);
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setReason("Request Timeout",code);
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hangup();
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hangup();
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}
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}
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else if (!m_hungup && code >= 100) {
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Lock lck(paramMutex());
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parameters().setParam("cause_sip",String(code));
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if (msg && msg->reason)
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parameters().setParam("reason_sip",msg->reason);
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else
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parameters().clearParam("reason_sip");
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}
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// Only update channels' callid if dialog tags change
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// Only update channels' callid if dialog tags change
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if (updateTags) {
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if (updateTags) {
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