similar to: Asterisk 10.12.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 14.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Hi, I only run a home-based asterisk (v1.4.18), and have never patched it, so I'm a unfamiliar with what time frame to expect for patches being implimented. I just downloaded (April 14) svn asterisk V1.6.1 r188415, on a "play" machine and noticed that when I stop asterisk, the astcanary module does not exit - when I restart asterisk, a new copy of astcanary also starts. In browsing

Try adding "server max protocol = nt1" in the [global] section? On Mon, Dec 24, 2018 at 1:59 PM Steven Hirsch via samba < samba at lists.samba.org> wrote: > Hi, all. > > I know this is ancient history, but I have a couple of DOS machines that > host older device programming hardware. I've been able to access an older > version of Samba for years without

On Mon, 24 Dec 2018, Luke Barone via samba wrote: Thanks, Luke. Unfortunately that makes no difference. Still getting: Error 5: Access has been denied at the MS_DOS console. > Try adding "server max protocol = nt1" in the [global] section? > > On Mon, Dec 24, 2018 at 1:59 PM Steven Hirsch via samba < > samba at lists.samba.org> wrote: > >> Hi, all.

Hi Sir It is requested to please tell the reason why the range of c(20945, 209547) is used in this function > npy <- length(events1[, "obs"])/(diff(range(ardieres[, "time"], + na.rm = TRUE)) - diff(ardieres[c(20945, 20947), "time"])) Please tell logic. Looking for quick response. Regards -- *Amina Shahzadi* [[alternative HTML version deleted]]

Hi, all. I know this is ancient history, but I have a couple of DOS machines that host older device programming hardware. I've been able to access an older version of Samba for years without incident. Last weekend I upgraded my server to Ubuntu 18.04, which provides Samba 4.7.6. Unfortunately, after hours of frustration I find I'm unable to connect from any of the older machines.

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

A week ago, I tried give realtime priority to asterisk proces using -p switch, asterisk was running inside astcanary, but yestarday asterisk probably starts eating all cpu and lock any access to computer, only ping was possible, so, anybody have experience, that ascanary process does really work to lower process priority in case of overloading? PJ

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

Hi LLVMers, there were a gcc ICE problem discussed in current mail list. Chris was right here: http://mail.cs.uiuc.edu/pipermail/llvmdev/2003-December/000693.html saying that the PR 12544 is not really the corresponding issue :) The correct one is PR 13392: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=13392 Interesting fact is that -O2 (or -O3) goes somehow around this

Hello, I was testing with sdp and something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine,

Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>