similar to: nat=yes

I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT,

-----Original Message----- From: Dirk Eddelbuettel [mailto:dirk.eddelbuettel at gmail.com] On Behalf Of Dirk Eddelbuettel Sent: Thursday, 19 January 2017 11:21 AM To: Klint Gore Cc: r-sig-debian at r-project.org Subject: Re: [R-sig-Debian] Taking determinant of a matrix of NAs results in intermittent memory corruption >So this converges towards 'old versions bad, new versions fine' ?

-----Original Message----- From: Dirk Eddelbuettel [mailto:dirk.eddelbuettel at gmail.com] On Behalf Of Dirk Eddelbuettel Sent: Thursday, 19 January 2017 12:41 PM To: Klint Gore Cc: Dirk Eddelbuettel; r-sig-debian at r-project.org Subject: RE: [R-sig-Debian] Taking determinant of a matrix of NAs results in intermittent memory corruption On 19 January 2017 at 01:26, Klint Gore wrote: | >So

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there

Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of

I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL:

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

One of my buddies signed up with GV yesterday with a view to using them on his * server. Problem is that the settings they gave him don't work with asterisk. They do however work with X-Lite. Any ideas? He's using the settings outlined on my web page. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

I got a call from GV on Monday evening telling me they wanted me to move my Asterisk server over to a new IP address (216.229.127.40) by this saturday. Why the couldn't tell me this in an email is beyond me but anyways .. So I done changed the number and so far its all ok but whilst testing I noticed that I could no longer accept incoming phone calls. I swapped back and still no inbound

Hi Folks, OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back to V1.09. Other than downloading the code, how do I do it? I thought someone once said that I have to delete all my modules or something? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hello list, I am getting the list with days of delay, take for example this message: Received: from unknown (HELO lists.digium.com) (216.207.245.17) by mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 -0000 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from <asterisk-users-bounces at lists.digium.com>) id

So I've been fighting to get the X100P working. A battle which I've kinda won but not without a cost. Before I won the Zaptel battle I was able to hear all of the messages that asterisk plays. For example, when I'm accessing VoiceMail I would have been requested to input my password. This did work but now it doesn't. Asterisk does show that it is playing the file but no audio is

Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

-----Original Message----- >From: Charles Lepple [mailto:clepple at gmail.com] >Sent: Thursday, 13 November 2014 12:12 AM > >I am surprised that the blazer_usb configuration didn't work. If you leave off >the matching options, does this find the UPS? > >[upsonicRT1000] > driver=blazer_usb ## or nutdrv_qx > port=auto > desc="UPSonic RT1000"