similar to: T.38 not working - help needed with log interpretation

Hi, On Tue, Dec 2, 2014 at 9:24 AM, Recursive <lists at binarus.de> wrote: > - Packets 14313, 14314: The provider re-invites asterisk for T.38 (confirmed by viewing the packet's details), asterisk answers "Trying ..." to the provider > - Packets 14315, 14321, 14322: Asterisk re-invites the local endpoint (again confirmed by looking into the packet's details), the

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

Just a thought regarding testing. Create a suitable TIFF file with more than 30 seconds worth of data and send it from Asterisk using SendFAX() to convince yourself whether Asterisk will work with your ITSP, you may still need to enable session timers. Have you considered setting up an extension on your Asterisk server which will receive the fax e.g. using ReceiveFAX() and see if your

Dear list, I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. 1) Ports and IP addresses which PJSIP bind to I have configured one transport like that: [tr_wZCMk5MvC2ATNzAr] type = transport protocol = udp bind = 192.168.20.48 Nevertheless, PJSIP

On Sun, Dec 21, 2014 at 4:54 AM, Recursive <lists at binarus.de> wrote: > Dear list, > > I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know. > > 1) Ports and IP addresses which PJSIP bind to > > I have configured one transport

https://bugzilla.samba.org/show_bug.cgi?id=14315 Bug ID: 14315 Summary: rsync hangs when many errors Product: rsync Version: 3.1.3 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: core Assignee: wayne at opencoder.net Reporter: mvitale at

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

https://bugzilla.samba.org/show_bug.cgi?id=14323 Bug ID: 14323 Summary: Defaults for --skip-compress are not working, everything is being compressed Product: rsync Version: 3.1.3 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: core

Hi, i have atwo node HA-cluster with pacemaker, corosync, libvirt and KVM. Recently i configured a new VirtualDomain which runs fine, but live Migration does not succeed. This is the error: VirtualDomain(vm_snipanalysis)[14322]: 2019/09/18_16:56:54 ERROR: snipanalysis: live migration to ha-idg-2 failed: 1 Sep 18 16:56:54 [6970] ha-idg-1 lrmd: notice: operation_finished:

Hi all, new to the list and this is probably a basic question and couldn't find anything clear googling around but I don't know how to handle calls to sip extensions not defined on sip.conf while using pattern matching. On my example I have sip extensions 10, 11, 12, and 13 on sip.conf. On a basic extension.conf I set up a pattern starting with "1" and a second digit should dial

Hi, I have 2 date vectors d1 and d2. d1 <- structure(c(14526, 14495, 14464, 14433, 14402, 14371, 14340, 14309, 14278, 14247, 14216, 14185), class = "Date") d2 <- structure(c(14526, 14509, 14488, 14466, 14453, 14441, 14396, 14388, 14343, 14333, 14310, 14281), class = "Date") I would like to create another dataframe with columns d1 and d2, where d1 is the original d1

On 10/13/16 10:23 AM, Konstantin Khomoutov wrote: > On Thu, 13 Oct 2016 09:53:19 -0500 > Bryan Holloway <bryan at shout.net> wrote: > > [...] >> Is there a way to see the IMAP commands coming from the client? I've >> tried looking at PCAPs, but of course they're encrypted so I can't >> see the actual dialog going on between the server and client. I

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call

I have a project I'm investigating. The goal is to basically add lossless extensions to Opus. You have an Opus stream with standard packets, but interwoven in there are extension packets that contain the residuals. Ideally, compliant decoders play the stream back and ignore the extension packets. This (hopefully) makes the "lossless" stream compatible with existing players.

> On October 13, 2016 at 6:52 PM Konstantin Khomoutov <flatworm at users.sourceforge.net> wrote: > > > On Thu, 13 Oct 2016 10:35:14 -0500 > Bryan Holloway <bryan at shout.net> wrote: > > > > [...] > > >> Is there a way to see the IMAP commands coming from the client? > > >> I've tried looking at PCAPs, but of course they're

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry

L'octidi 18 pluviôse, an CCXXV, Miscellaneous a écrit : > > The RLP draft mentioned on the libvorbis docs page, are there sample > > implementations (in C) of this yet anywhere?  > Sorry that should say RTP, not RLP I am not sure I understand exactly what you are asking, but libavformat has a RTP packetizer that works for both Vorbis and Theora, implemented in rtpenc_xiph.c.

Thanks for testing - are you able to see if there's anything in the server logs? I've just committed some extra verbosity in the client's log messages that might clarify where it is exiting (patch attached). -d On Fri, 29 Mar 2019, Adam Eijdenberg wrote: > On Wed, Mar 27, 2019 at 10:04 PM Damien Miller <djm at mindrot.org> wrote: > > > > OpenSSH 8.0p1 is almost

L'octidi 18 pluviôse, an CCXXV, Miscellaneous a écrit : > I've built some simple code on the excellent libvorbis API overview on > the xiph.org site, but the example relies on the ogg_packet struct for > final output and input to decoder, and shows now examples on how to do > without it. Why do you not want it? It is just a simple C struct to carry the packet's metadata.

On Sat, Nov 10, 2012 at 3:21 AM, Patrik Hägglund H <patrik.h.hagglund at ericsson.com> wrote: > From r167602 and onwards I get a fail in 'make check-all' for llvm+clang, > built with gcc-4.7.1 (but not with clang-3.1) on Linux x86_64: > > Failing Tests (1): > Clang :: CodeGenCXX/mangle-ms-templates.cpp > > clang: