Summary: | ASTERISK-17054: Exchange AutoAttendant cannot make calls back to asterisk (worked in 1.6.2.2 + 1.6.2.6) | ||
Reporter: | Chris Baker (cmbaker82) | Labels: | |
Date Opened: | 2010-12-02 10:20:53.000-0600 | Date Closed: | 2011-01-19 10:09:53.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_dial |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) sipdebug.txt | |
Description: | When someone calls in to the exchange auto attendant they can say someones name and it will dial the correct extenstion. This worked fine in 1.6 but does not work in version 1.8.0. In the example below the call came in from 3174999999 (masked real number) was transferred to 888 (exchange um) and then exchange um tried to transfer to extension 501 which failed. ****** ADDITIONAL INFORMATION ****** Removed inline debug - pabelanger | ||
Comments: | By: rsw686 (rsw686) 2010-12-02 11:35:54.000-0600 This is probably related to issue 0018185. When using the auto attendant to call somebody, Exchange UM actually transfers the call. By: Paul Belanger (pabelanger) 2010-12-02 23:20:06.000-0600 Attach a debug log (see below) to the issue tracker, not within the notes field. --- We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Chris Baker (cmbaker82) 2010-12-07 15:37:16.000-0600 I should be able to upload a debug log in the next couple of days. By: Chris Baker (cmbaker82) 2011-01-06 14:41:51.000-0600 I've uploaded a sip debug log that shows what happens. This is from 1.8.1.1 By: Chris Baker (cmbaker82) 2011-01-06 15:53:31.000-0600 I installed 1.8.2-rc1 and it does not have the problem of transfering from the autoattendent to an extension; however it's now having problems transfering to the exchange voice mailbox. By: Chris Baker (cmbaker82) 2011-01-06 16:17:59.000-0600 Transferring directly to voicemail was fixed by changing my dial plan from: exten => _*11XXXX,n,Dial(SIP/EXUM/8800) To: exten => _*11XXXX,n,Dial(SIP/EXUM/8800,10,TtWw) |