asterisk users - Nov 2015

Monday November 30 2015
TimeRepliesSubject
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
TimeRepliesSubject
7:33PM 1 Asterisk 13.6.0/How to set up default_outbound_endpoint
 
Saturday November 28 2015
TimeRepliesSubject
1:14PM 2 endwhile jumping out of macro
 
Thursday November 26 2015
TimeRepliesSubject
9:02AM 0 Patched Res_Musiconhold.So module
 
Wednesday November 25 2015
TimeRepliesSubject
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
TimeRepliesSubject
12:39PM 0 subscriber state before dial
12:20PM 2 subscriber state before dial
 
Monday November 23 2015
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
11:22AM 0 Trying to compile DAHDI on Pidora 2014 (RPi)
 
Saturday November 14 2015
TimeRepliesSubject
7:03AM 0 Re-registration stops after temporal error
 
Friday November 13 2015
TimeRepliesSubject
10:01PM 3 Trying to compile DAHDI on Pidora 2014 (RPi)
 
Thursday November 12 2015
TimeRepliesSubject
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
TimeRepliesSubject
6:55PM 0 accept DMTF tone during ringing
 
Monday November 9 2015
TimeRepliesSubject
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
TimeRepliesSubject
9:50PM 2 accept DMTF tone during ringing
8:13AM 1 asterisk 13 systemd
 
Saturday November 7 2015
TimeRepliesSubject
8:34AM 0 asterisk 13 systemd
1:17AM 1 no ringing tone with Dial option r
 
Friday November 6 2015
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
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
TimeRepliesSubject
5:38PM 5 no ringing tone with Dial option r