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tnp20 Brekeke Member
Joined: 21 Apr 2007 Posts: 11
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Posted: Thu Nov 15, 2007 7:39 am Post subject: Call rolls into voicemail without rining |
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1. Brekeke Product Name and version: 2.0.7.2
2. Java version:1.6.0_03-b05
3. OS type and the version: Windows XP
4. UA (phone), gateway or other hardware/software involved: Grandstream BT102 and GXP2000
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : Pattern 4 applies
6. Your problem:
I have 2 phones registered with PBX, GrandstreamBT102 and GXP2000. I can call each phone from the other and it works fine. I have set the voicemail rollover to 20 second ring so as expected after 20 seconds, the call rolls into voice mail.
However, when I have another call coming in from outside the PBX(i.e. an unregistered phone to the PBX) from another SIP server (in this case a phone registered to another SIP server) and I have a rule that says $auth=false, and I try to call the registered BT102 or GXP2000, it rolls straight into voicemail without even ringing.
I tried setting the authentication for register and invite off, but it made no difference...I tried changed the order of some of the rules but it did not makemuch difference...the only thing that happened was instead of going to voicemail, it just kept ringing and this time, after 20 seconds it did not roll into voice mail..so for some reason call coming from outside either keep riniging or go straight into voicemail. |
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voipwell.com Partner PBX
Joined: 20 Sep 2005 Posts: 528
Location: Tannersville, Pennsylvania
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Posted: Thu Nov 15, 2007 10:36 pm Post subject: |
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downgrade java to 1.5 and upgrade to 2.1.5.6 |
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tnp20 Brekeke Member
Joined: 21 Apr 2007 Posts: 11
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Posted: Fri Nov 16, 2007 9:08 am Post subject: |
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I tried downloading the beta 2.1.5.6... it blew away my previous license and now I am on a 5 day license.
The problem is still there with 2.1.5.6...though I need to test it with downlevel version of Java 1.5...
do you think Java 1.5 is going to make much difference ? |
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tnp20 Brekeke Member
Joined: 21 Apr 2007 Posts: 11
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Posted: Fri Nov 16, 2007 10:24 am Post subject: My dial plan for reference |
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$port=15062 To=%1%2
$localhost=true $target=localhost:15060
$request=^INVITE $auth=false
To=(^.*sip:)pbx(.+$)
$port=15062 $auth=false
$localhost=true $target=%1
$request=^INVITE X-PBX-Target=
X-PBX-Target=(.+) &net.sip.hide.loopback=true
$port=15062 $auth=false
$localhost=true &net.sip.hide.loopback=true
$request=^INVITE
$request=^INVITE $auth=false
$registered=false $target=localhost:15060
$request=^INVITE $auth=false
$target=localhost:15060
$request=^SUBSCRIBE $target=localhost:15062
Event=^x-tthd.*|^message-summary$
$port=15060 $action=register
$localhost=true $auth=false
$request=^REGISTER &net.registrar.thru.catchhere=false
&net.registrar.onlyglobal=true
Brekeke SIP Server for Brekeke PBX, Version 2.1.5.6
Copyright © 2002-2007, Brekeke Software, Inc. |
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