asterisk users - Oct 2014

Friday October 31 2014
TimeRepliesSubject
3:26PM 0 PlayTones while in call
4:26AM 0 Re: asterisk-users Digest, Vol 123, Issue 38
1:32AM 0 Paul Albrecht
 
Thursday October 30 2014
TimeRepliesSubject
10:01PM 0 MWI publish VIA pjsip for non sip channels
7:40PM 0 PlayTones not working
7:18PM 0 Register multiple phones to a single AOR with PJSIP
1:52PM 0 ${HASH(SIP_CAUSE,<channel-name>)}
9:35AM 0 Asterisk registration with Dialogic HMP.
 
Wednesday October 29 2014
TimeRepliesSubject
10:16PM 0 Asterisk 13 : SILK codec ?
6:35PM 0 OT: script to remove leading and trailing silence
4:10PM 0 My Asterisk can not send fax via T.38
10:50AM 0 Astricom 2014 presentations
 
Tuesday October 28 2014
TimeRepliesSubject
3:22PM 0 dialplan reload context
12:30PM 0 Asterisk 13 stable?
12:21PM 0 Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.
9:58AM 0 Asterisk 12 - zombie processes
 
Monday October 27 2014
TimeRepliesSubject
11:35PM 0 sip.conf to pjsip.conf conversion script
7:04PM 0 AppKonference 2.6
12:56PM 0 authentication time for asterisk server
8:48AM 0 Detect hangup due to RTP timeout
3:42AM 0 Setting Music on Hold with the Manager Interface
 
Sunday October 26 2014
TimeRepliesSubject
5:34PM 0 Asterisk and Kamailio Load Balancing
8:55AM 0 Port number in From URI on Asterisk 12 PJSIP
8:22AM 0 DTMF behavior in asterisk 12 with PJSIP
1:16AM 0 Asterisk 13.0.0 Now Available!
 
Saturday October 25 2014
TimeRepliesSubject
10:43PM 0 make asterisk do something when an outgoing call is picked up
9:09PM 0 Voicemail ODBC Storage
 
Friday October 24 2014
TimeRepliesSubject
10:09PM 0 Questions on musiconhold.conf custom mode
3:51PM 0 Call forwarding from Phones and getting the referrer IP
12:47PM 0 ConfBridge / internal_sample_rate=auto / warning
11:39AM 0 Debugging issues with setup
 
Thursday October 23 2014
TimeRepliesSubject
8:32PM 0 Ast 13 beta 3 - Segfault when calling on pjsip trunk with directmedia=yes
3:26PM 0 11.13.1: unable to load sip.conf (or iax )
3:07PM 0 logger.conf
2:57PM 0 Auto video call hangup
2:13AM 0 PJSIP and NAT behind a dynamic IP address
 
Wednesday October 22 2014
TimeRepliesSubject
9:43PM 0 SPA504G auto answer
7:06PM 0 Re: AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)
6:40PM 0 Video call
4:14PM 0 Re: [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)
3:33PM 0 SIP dialing with authentication with dialstring and wothout sip; conf
2:31PM 0 AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)
3:23AM 0 res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
 
Tuesday October 21 2014
TimeRepliesSubject
6:54AM 0 Asterisk 11.9.0 crash and restart
6:19AM 0 [asterisk-user] Confbridge Kick Action
12:29AM 0 TLS on SIP trunk
 
Monday October 20 2014
TimeRepliesSubject
9:58PM 0 bristuff-0.4.0-RC4-xr7
4:41PM 0 AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability
4:40PM 0 Asterisk 1.8.28-cert2, 1.8.31.1, 11.6-cert7, 11.13.1, 12.6.1, 13.0.0-beta3 Now Available (Security Release)
 
Saturday October 18 2014
TimeRepliesSubject
9:47PM 0 Asterisk Crashes Randomly with Cepstral Swift TTS
9:03AM 0 Asterisk 12.6 and MWI, no more working
 
Thursday October 16 2014
TimeRepliesSubject
1:07PM 0 Queue: Passing params to macros and gosubs
7:48AM 0 Asterisk GOIP Outgoing Callerid not working
 
Wednesday October 15 2014
TimeRepliesSubject
11:36PM 0 OpenSIPS Summit Oct 21st before Astricon
10:59AM 0 allo.com gsm card with AsteriskNOW
6:50AM 0 Asterisk 12 CDR dst field empty
 
Tuesday October 14 2014
TimeRepliesSubject
3:47PM 0 Issue playing high quality white noise
2:57PM 0 Do subroutines need their own h extension?
12:59PM 0 debugging T.38 issues
 
Monday October 13 2014
TimeRepliesSubject
6:50AM 0 asterisk stun setup , not using public ip returned by stun server
 
Saturday October 11 2014
TimeRepliesSubject
1:09AM 0 Reset calls processed counter
 
Friday October 10 2014
TimeRepliesSubject
1:14PM 0 howto cancel simultaneous calls - dial(sip/phone1&sip/phone2)
10:47AM 0 Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages
 
Thursday October 9 2014
TimeRepliesSubject
1:20PM 0 SIP over 3G Mobile Network using NAT
12:06PM 0 sdp_crypto_process: Crypto life time unsupported: crypto
 
Wednesday October 8 2014
TimeRepliesSubject
1:35PM 0 Asterisk LTS segment faults
 
Tuesday October 7 2014
TimeRepliesSubject
10:32AM 0 Grandstream GXP2160 + SRTP
8:11AM 0 Asterisk Phone ( Telecom feature )
5:21AM 0 new app_swift is live
 
Sunday October 5 2014
TimeRepliesSubject
11:40PM 0 Setting channel musicclass from AGI
3:45PM 0 Voicemail message number off by one when using ODBC storage
 
Saturday October 4 2014
TimeRepliesSubject
2:48PM 0 Pjsip and regcontext (for DUNDi)
1:43PM 0 No chan_sip in compiled asterisk-11.13.0
 
Friday October 3 2014
TimeRepliesSubject
7:12PM 0 Lost audio on forwarded calls
9:33AM 0 SPA112: one analog phone works, not the other
 
Thursday October 2 2014
TimeRepliesSubject
5:52PM 0 how to strip +1 out of incoming number
5:32PM 0 Voice Mail Questions
3:04PM 0 AstLinux 1.2.0 Released
9:02AM 0 Sent ami event from AGI?
 
Wednesday October 1 2014
TimeRepliesSubject
2:09PM 0 JABBER_STATUS CODE 7
1:00PM 0 CALLERID(num) and CDR(clid) - originate
9:40AM 0 PBX hacked: why hundred of calls to the same number ?