Summary: | ASTERISK-05952: Call-limit does not function while using queue | ||
Reporter: | Kenneth Holm (saitech) | Labels: | |
Date Opened: | 2006-01-02 11:54:26.000-0600 | Date Closed: | 2006-01-02 12:37:39.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I am having some problems with limiting my sip calls. I want to eliminate the call waiting for X-Lite/Eyebeam, so im using the call-limit function to ensure that only 1 sip call is allowed pr user. If the asterisk receive a sip call and put it into a queue. The first attempt to call the user is failed, if he already have a call. But the after the first announce in the queue, the call will not fail, and starts as a call waiting for the user. I have also tried the deprecated funtions incominglimit and outgoinglimit both sat to 1. I have used thes setup before with asterisk 1.0.7 where it did function pretty good. | ||
Comments: | By: Jason Parker (jparker) 2006-01-02 12:37:21.000-0600 6111 is a duplicate of this, however, per request, I'll close this one instead of 6111. Please make any changes/updates to that issue instead. |