similar to: NAT router and off-premise SIP audio problem

Rich, thank you for your informative reply. I checked with our admin and he replied: "I setup from the start "nat=yes" and "canreinvite=no" on sip phones from Internet and modified the rtp channels (voice ports) and the rtp port on the phones. Still have the same problem, no sound." Perhaps the VPN solution is something we should try but this is more limiting than

I''m looking at an example where a stub seems to work sometimes, and sometimes appears to become "unstubbed". I haven''t boiled it down to a minimal example, but it goes something like this: ---- the model: class Premise << ActiveRecord::Base def lookup_stuff_on_the_web $stderr.puts("entering lookup_stuff_on_the_web with #{self}") ... end end

I have two models with a straightforward has_many / belongs_to relationship: class Premise < ActiveRecord::Base has_many :metered_services, :dependent => :destroy ... end class MeteredService < ActiveRecord::Base belongs_to :premise ... end and nested routes to match: Demo::Application.routes.draw do devise_for :users resources :premises do resources :metered_services

I''m doing RSpec controller testing with CanCan authorization, and I''m seeing something I''ve never seen in RSpec before: the same test run twice fails on the second one. I am NOT doing before(:all) or other things that should cause state to persist between tests: Here''s the relevant code: context "POST create" do context "with user logged

On Dec 14, 2013, at 11:27, David Blaikie <dblaikie at gmail.com> wrote: > (+llvmdev) > > > On Sat, Dec 14, 2013 at 11:02 AM, Adrian Prantl <aprantl at apple.com> wrote: > Hi David, > > I just bisected a debug info problem we were experiencing down to your legendary commit: > Revert "Revert "Revert "Revert "DebugInfo: Omit debug info for

(+llvmdev) On Sat, Dec 14, 2013 at 11:02 AM, Adrian Prantl <aprantl at apple.com> wrote: > Hi David, > > I just bisected a debug info problem we were experiencing down to your > legendary commit: > Revert "Revert "Revert "Revert "DebugInfo: Omit debug info for dynamic > classes in TUs that do not have the vtable for that class"”""

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

How does one generate a form within a view using javascript? Specifically, assume views/premise/new.html.erb looks like this: <%= form_for(Premise.new) do |f| %> <%= f.hidden_field :full_address, :value => $FULL_ADDRESS %> <%= f.hidden_field :geocoding, :value => $GEOCODING %> <%= f.submit :value => "use this address" %> <%= f.label

Can anybody give me a reason why this would be a bad idea. So the premise for this question is that I setup an exclude=*.i368,*.i686 in my yum.conf. While doing a yum update I come across missing package dependencies for instance mkinitrd for the i386 package. I noticed there is already one for x86_64. I realized during the kickstart install that some of these *.i386 got installed before I

Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP service provider in the middle. We had slightly different parameters, e. g. that we would

Ned, Thank you very much for the response. Great example following through on the premise. It sounds like I need to have a better understanding of the traffic patterns on my network to know the optimal order for iptables filtering rules. My brief example - Premise: I want to limit outsiders from interfering with LAN client machines. So, I have the following rules regarding forwarding traffic:

Hi, I need to implement trouble tracking system, we have 250 users in one premise & 3 desktop support technicians. I need to implement trouble ticket system, where user will enter their application / other issues. Mail will be sent to technician available on duty. trouble ticket will be provided to user & will be given close stat once resolved. Kindly suggest me one such application

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called

> From: George Lin [mailto:glin@cosini.com] > > I want to deploy multiple SIPs phone in our office. And we have shutdown > the > firewall at our office router(with ip 211.x.x.x). we have deployed the > asterisk with IP 218.x.x.x. > > All SIP phones have 192.x.x.x. We have something similar George, * sits outside the firewall with a registered IP address, the SIP phones

Hi, I'm looking for a few pointers on using ATA to connect Asterisk to the PSTN. Basically, I'm running a Hosted PBX service, and in urban centers I can usually get SIP or PRIs. Since I sell my customers SIP hardphones, the data flow is like this: Customer's SIP Hardphone ---- My own Asterisk ----- Outside lines But when it comes to smaller villages (I deal with people in tiny

On 2/21/23 20:32, Eric Blake wrote: > On Tue, Feb 21, 2023 at 07:07:38PM +0100, Laszlo Ersek wrote: >> Well, given Daniel's comments meanwhile, it seems like the original >> execvp() is something we shouldn't fret about. :/ > > glibc marks execvp() and exevpe() as 'MT-Safe env', which means it > does not modify 'environ' and presumably does not use

All, These discussions seem to be based on the premise that there is a need for the compiler to exploit undefined behavior for performance optimization reasons. So far the only beneficial optimization I am aware of that relies on some form of “undefined” is Dan Gohman’s original project for LP64 targets of promoting i32 induction variables to i64 and hoisting sign-extension out of the

I wanted to check that I'm headed in the right direction before I work more on LLVM debug info. What I'd like to do is update DIBuilder to expose exactly the facilities represented in modern DWARF, and to store that in the bitcode. In LLVM we would have a DwarfOpts that specifies the major version of DWARF we're targetting and some additional compatibility flags to work around debugger

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ?

Does your *shared* hosting account work without errors? I''m interested in both positive and negative reports. If you are willing, please contribute - name of the hosting service - how long you have used them - your experience ("flawless!", "it''s crap!") You *do not* need to reply saying your VPS service works great. This is about *shared* hosting