similar to: Use of Re-INVITE and t,T (transfer) options

Kevin P. Fleming wrote: > This is because hook-flash support is in chan_zap, not in the core of > Asterisk. There is no need to support hook-flash on any other channel > type, because every other channel type supported by Asterisk has its own > (more reliable) methods of signaling. I'm seeing many discussions on how to provision Blind Transfer, Attendant Transfer and Call

2014-11-09 9:59 GMT-03:00 Charles Lepple <clepple at gmail.com>: > On Nov 9, 2014, at 6:58 AM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: > >> I'm attaching a new debug log with this modification, just in case, >> but I'm still seeing the lines you've pointed at: >> >> 0.062308 Path: UPS.PowerSummary.Voltage, Type:

2014-11-08 20:46 GMT-03:00 Charles Lepple <clepple at gmail.com>: > On Nov 7, 2014, at 7:30 AM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: > >> 2014-11-07 0:16 GMT-03:00 Charles Lepple <clepple at gmail.com>: >>> On Nov 6, 2014, at 5:56 PM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: >>> >>>>

On Thu, Jan 31, 2013 at 12:11 AM, Marcelo Tosatti <mtosatti at redhat.com> wrote: > On Wed, Jan 30, 2013 at 11:21:08AM +0300, Andrey Korolyov wrote: >> On Wed, Jan 30, 2013 at 3:15 AM, Marcelo Tosatti <mtosatti at redhat.com> wrote: >> > On Tue, Jan 29, 2013 at 02:35:02AM +0300, Andrey Korolyov wrote: >> >> On Mon, Jan 28, 2013 at 5:56 PM, Andrey Korolyov

Hey folks! I need to allow smartcard authentication of a third party certificate generated with an UPN that has a suffix that is not my domain name. From AD literature, it's possible. I followed these guidelines to make an additional UPN available for login: https://technet.microsoft.com/en-us/library/cc772007.aspx But I'm missing something. Kerberos does a part of the job, but then

On Monday 10 September 2018 at 21:54:33, Marcelo Terres wrote: > I have think it should be > > context=0705680837 > > Not > > context=[0705680837] Ha! You're right... so simple :) Antony. > On Mon, 10 Sep 2018, 20:43 , <asterisk at a-domani.nl> wrote: > > On 2018-09-09 10:27, Antony Stone wrote: > > > > <snip > > > > >

On Wed, Apr 18, 2018 at 9:33 AM, Marcelo Ricardo Leitner <marcelo.leitner at gmail.com> wrote: > On Tue, Apr 17, 2018 at 04:35:18PM -0400, Vlad Yasevich wrote: >> On 04/02/2018 10:47 AM, Marcelo Ricardo Leitner wrote: >> > On Mon, Apr 02, 2018 at 09:40:01AM -0400, Vladislav Yasevich wrote: >> >> Now that we have SCTP offload capabilities in the kernel, we can

Asterisk Project Security Advisory - Product Asterisk Summary Re-invite with T.38 and malformed SDP causes crash. Nature of Advisory Remote Crash Susceptibility Remote Authenticated Sessions Severity Minor

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

2014-11-07 0:16 GMT-03:00 Charles Lepple <clepple at gmail.com>: > On Nov 6, 2014, at 5:56 PM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: > >> battery.voltage: 10000000. >> battery.voltage.nominal: 0.0 > > Well, these are certainly interesting values ;-) > > I suspect the correction for earlier units has failed. Can you please start

On Nov 7, 2014, at 7:30 AM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: > 2014-11-07 0:16 GMT-03:00 Charles Lepple <clepple at gmail.com>: >> On Nov 6, 2014, at 5:56 PM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: >> >>> battery.voltage: 10000000. >>> battery.voltage.nominal: 0.0 >> >> Well,

The VNC server just listens to the appropriate TCP/IP port and then runs Xvnc which does the actual VNC communication. Ideally I'd be able to do the same thing for RDP then the daemon doesn't get any more complicated, and a bug in the RDP layer can't crash the server. I don't know enough about how NX works but I suspect we could do the same thing as for VNC and RDP. I'm

On 04/12/2015 19:17, Marcelo Ricardo Leitner wrote: > Em 03-12-2015 14:46, Duncan Brown escreveu: >> Here is a couple of pictures, >> >> http://i.imgur.com/Vqvqn1H.jpg >> http://i.imgur.com/WQaz1j9.png >> >> Any use? > > Of some. It's failing on ftrace initialization while allocating > memory, but can't know the real reason. It can be just

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their

2014-10-30 0:49 GMT-03:00 Charles Lepple <clepple at gmail.com>: > On Oct 29, 2014, at 5:08 PM, Marcelo Fernandez <marcelo.fidel.fernandez at gmail.com> wrote: > >> Hi, >> >> I've bought a Liebert GXT3 UPS and I'm trying to use it with nut. I'm >> using Ubuntu 14.04, but I'm having troubles (similar to this thread >> [1]) using the

Hello everyone, I'm using the very good imaptest [0] tool to test my little imap server implementation. I've tried to use the dovecot-crlf [1] file, but it looks like there are some major issues : $ grep -n "In-Reply-To.*;" tests/data/dovecot-crlf 479:In-Reply-To: <20020806175441.GA7148 at linux.taugt.net>; from rueckert at informatik.uni-rostock.de on Tue, Aug 06, 2002

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

Em 05-12-2015 05:35, Duncan Brown escreveu: > On 04/12/2015 19:17, Marcelo Ricardo Leitner wrote: >> Em 03-12-2015 14:46, Duncan Brown escreveu: >>> Here is a couple of pictures, >>> >>> http://i.imgur.com/Vqvqn1H.jpg >>> http://i.imgur.com/WQaz1j9.png >>> >>> Any use? >> >> Of some. It's failing on ftrace