similar to: One way sound

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

I'm trying to have the following small architecture: PSTN | Linux box with an public IP and Asterisk | Internet | Firewall/NAT/Router | SIP Proxy | IP Phones (or soft phone like X-Lite)

I'm running Asterisk on an external static IP address, siproxd on a different server with its own external static IP address, and communicating using a Grandstream behind a NAT firewall configured to register with Asterisk using siproxd as the outbound proxy. Now I'm aware that siproxd is not intended to be used as an outbound proxy but rather as a SIP relay when installed on the same box

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

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I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Le 01/05/2015 00:05, Andrew Martin a ?crit : > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> >> To: asterisk-users at lists.digium.com >> Sent: Thursday, April 30, 2015 4:43:33 PM >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In >> >>> I am running Asterisk 11.12.0 on CentOS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: > > > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> To: >> asterisk-users at lists.digium.com Sent: Friday, May 1, 2015 6:42:38 >> AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently >> Cannot Call In

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > > internal phones are located on

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- > >> From:

Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear.

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line. Don't

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. Not sure

I'm looking for a device that will announce sounds from other devices to Asterisk to be heard on my Grandstream sip phones throughout the house. The Grandstream phones are already set up for ammouncements by pressing a dial code. I need a device that will detect and play sounds automatically from other devices such as a talking clock, driveway sensor or other home automation devices like

Hello, I'm trying to do something really easy : transfer a PSTN call to a H323 client. This works great. Now I'm trying to use the SetMusicOnHold function. I din't find any doc about it, I've just seen some mails in the list archive, but it still doesn't work. That's my extension.conf : [incoming] exten => s,1,SetMusicOnHold,default exten =>

Hello, I've seen various suggestions thrown around for hardware when people ask, but can we all agree on some basic hardware recommendations for a few basic setups(and post them on a website) to make it easier for new people to avoid some of the hardware/software pitfalls when they are setting up their first systems. Something like this: (THIS IS JUST A PROPOSED LAYOUT SO PLEASE BE GENTLE)