similar to: Networking in KVM

On Fri, 2016-03-18 at 13:02 -0400, Mike - st257 wrote: > Paul, > > > On Thu, Mar 17, 2016 at 5:47 PM, Digimer <lists at alteeve.ca> wrote: > On 17/03/16 04:47 PM, paul.greene.va at verizon.net wrote: > > Thanks, I followed the 2nd article, and it got the existing > virtual > > machines communicating with each other. > >

Paul, On Thu, Mar 17, 2016 at 5:47 PM, Digimer <lists at alteeve.ca> wrote: > On 17/03/16 04:47 PM, paul.greene.va at verizon.net wrote: > > Thanks, I followed the 2nd article, and it got the existing virtual > > machines communicating with each other. > Right, so your VMs are on the same bridge group now (at Layer2 of OSI). > > > > However, any new

<div style="FONT-FAMILY: Arial; COLOR: rgb(0, 0, 0); FONT-SIZE: 12px"><div>Thanks, I followed the 2nd article, and it got the existing virtual machines communicating with each other.<br /><br />However, any new virtual machines I created after making the changes can't communicate with anything, they can't even get out to the internet, even if I manually

../libtool: line 1231: cygpath: command not found You need to put cygpath in your PATH. This might also be why configure is failing. Best, Tristan --20cf307f35267e30eb0505f59648 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <div dir=3D"ltr">Hi,<br><div class=3D"gmail_extra"><br><div class=3D"gmail_=

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

<div dir="ltr"><br> </div><div dir="ltr">Actually I want to run some analysis pass only on the user defined functions but not on the library functions. Is there any boolean method that can tell which is a library function and which is not? </div><div class="wps_quotion">On 5 Feb 2020 5:23 a.m., David Blaikie

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

If you don't use 'CERTVERIFY 1', then this will at least make sure that nobody can sniff your sessions without a large effort (...) > So, do I misunderstand CERTVERIFY directive ? Or is there a bug ? >> Can you reproduce such behaviour ? >> > > I'm not sure what is going on. Can you try running 'upsmon' with debugging > enabled? The following are

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys