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