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 |