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Nov 2015
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asterisk users
79364 threads
Nov 2015
81 threads
Monday November 30 2015
Time
Replies
Subject
5:34PM
1
endwhile jumping out of macro
3:42PM
1
force sip URI call through PBX
3:40PM
0
endwhile jumping out of macro
Sunday November 29 2015
Time
Replies
Subject
7:33PM
1
Asterisk 13.6.0/How to set up default_outbound_endpoint
Saturday November 28 2015
Time
Replies
Subject
1:14PM
2
endwhile jumping out of macro
Thursday November 26 2015
Time
Replies
Subject
9:02AM
0
Patched Res_Musiconhold.So module
Wednesday November 25 2015
Time
Replies
Subject
7:16PM
0
[CFP] reminder! FOSDEM RTC dev-room talks: deadline Friday
7:02PM
2
Patched Res_Musiconhold.So module
4:45PM
0
Dialing a call back out on same SIP trunk as it came in
3:31PM
0
Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk
3:29PM
0
Patched Res_Musiconhold.So module
2:27PM
2
Dialing a call back out on same SIP trunk as it came in
1:30PM
0
תשובה: Dialing a call back out on same SIP trunk as it came in
1:14PM
2
Dialing a call back out on same SIP trunk as it came in
Tuesday November 24 2015
Time
Replies
Subject
12:39PM
0
subscriber state before dial
12:20PM
2
subscriber state before dial
Monday November 23 2015
Time
Replies
Subject
9:01PM
1
How exactly does asterisk know what IP to send RTP traffic to?
7:16PM
1
Which router/firewall would you use for a virtual-PBX Asterisk installation?
Saturday November 21 2015
Time
Replies
Subject
8:15PM
3
Patched Res_Musiconhold.So module
8:10PM
2
SIP calls dropping at 15 minutes
6:19PM
0
SIP calls dropping at 15 minutes
5:29PM
2
Looking for a patched res_musiconhold.so Module for 32-bit Linux/Astersk
11:48AM
1
very long line in extensions.conf
7:51AM
1
Error while compiling asterisk asterisk-1.8.32.3
6:52AM
1
How to custom the message on call busy or no answer in asterisk
Friday November 20 2015
Time
Replies
Subject
10:22PM
0
Which router/firewall would you use for a virtual-PBX Asterisk installation?
8:24PM
3
Which router/firewall would you use for a virtual-PBX Asterisk installation?
4:13PM
2
SIP calls dropping at 15 minutes
6:51AM
0
How to custom the message on call busy or no answer in asterisk
1:15AM
2
How to custom the message on call busy or no answer in asterisk
Tuesday November 17 2015
Time
Replies
Subject
7:29AM
0
SIP URI call to specific country code
7:03AM
0
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
2:58AM
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
2:44AM
0
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
2:34AM
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
2:16AM
0
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
1:35AM
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
Sunday November 15 2015
Time
Replies
Subject
11:22AM
0
Trying to compile DAHDI on Pidora 2014 (RPi)
Saturday November 14 2015
Time
Replies
Subject
7:03AM
0
Re-registration stops after temporal error
Friday November 13 2015
Time
Replies
Subject
10:01PM
3
Trying to compile DAHDI on Pidora 2014 (RPi)
Thursday November 12 2015
Time
Replies
Subject
10:00PM
0
Favorite this issue to Vote to enable Google Chromium to natively play WAV type 49 or 0x0031 (an asterisk default voice mail format).
4:46PM
0
No sound with internal calls depending on which phones
4:05PM
3
No sound with internal calls depending on which phones
3:25PM
0
No sound with internal calls depending on which phones
3:22PM
3
No sound with internal calls depending on which phones
Tuesday November 10 2015
Time
Replies
Subject
6:55PM
0
accept DMTF tone during ringing
Monday November 9 2015
Time
Replies
Subject
8:02PM
1
Asterisk unable to receive DTMF tone.
2:54PM
0
How to encode plus sign in REGEX function in dialplan?
2:06PM
2
How to encode plus sign in REGEX function in dialplan?
12:28PM
0
bad performance centos6 ->centos7
10:40AM
1
Update new IP address (move temporarily) for INVITE
Sunday November 8 2015
Time
Replies
Subject
9:50PM
2
accept DMTF tone during ringing
8:13AM
1
asterisk 13 systemd
Saturday November 7 2015
Time
Replies
Subject
8:34AM
0
asterisk 13 systemd
1:17AM
1
no ringing tone with Dial option r
Friday November 6 2015
Time
Replies
Subject
7:39PM
0
issue with bridgeConference
1:06PM
0
How to encode plus sign in REGEX function in dialplan?
9:54AM
0
Find me macro - calling multiple people to get a hold of one
9:18AM
2
bad performance centos6 ->centos7
8:34AM
0
Re-Invite to Native Bridge
3:08AM
0
Find me macro - calling multiple people, to get a hold of one
Thursday November 5 2015
Time
Replies
Subject
7:56PM
0
DAHDI driver question for custom card
5:36PM
2
DAHDI driver question for custom card
6:49AM
3
How to encode plus sign in REGEX function in dialplan?
Wednesday November 4 2015
Time
Replies
Subject
4:42PM
0
Find me macro - calling multiple people to get a hold of one
4:24PM
4
Find me macro - calling multiple people to get a hold of one
3:19PM
1
PJSIP with registratrion to DNS SRV records fail with PJLIB_UTIL_EDNSNOANSWERREC
8:43AM
0
no ringing tone with Dial option r
8:40AM
1
no ringing tone with Dial option r
Tuesday November 3 2015
Time
Replies
Subject
6:11PM
1
no ringing tone with Dial option r
6:07PM
0
no ringing tone with Dial option r
2:51PM
0
Demo lab for connectivity tests
6:54AM
0
1.8.32.3 - no timing indicated, tens of thousands of __sip_autodestruct error messages
Monday November 2 2015
Time
Replies
Subject
10:45PM
0
dahdi how to forward to another number
9:16PM
1
issue with bridgeConference
4:55PM
0
Asterisk Mobile Dialer
4:45PM
2
Asterisk Mobile Dialer
2:39PM
0
Using external RTP proxy for res_pjsip
2:14PM
0
We're one week away from OpenSIPS Week in Austin!
2:09PM
2
Using external RTP proxy for res_pjsip
Sunday November 1 2015
Time
Replies
Subject
5:38PM
5
no ringing tone with Dial option r