similar to: Connecting peer if the peer is already connected

Hi list! I'm working hard to securing my Asterisk... Now I deleted all possibility to access the node as "anonymous" and every call through the proxy will be checked (just known peers are allowed to use it). Furthermore, I restricted the registration of my home phones to the Network I reserved for them and I changed the port on my Firewall, so that I don't use 5060 anymore. Now

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

OK, I have windoze ME installed on my system and have been trying to run IE 6.0 with wine release 20010824. Trying to start iexplore.exe goes along until: Unhandled exception: page fault on read access to 0xdeadbeef in 32-bit code (0xdeadbeef). In 32-bit mode. 0xdeadbeef (_end+0x9df10793): *** Invalid address 0xdeadbeef (_end+0x9df10793) -- no code -- Enter path to file 'lstat64.c':

On Tuesday 09 Jun 2015, Luca Bertoncello wrote: > Now, I tried to register the user of my cellphone using a PC, as my > cellphone was already registered. > And Asterisk accepted this registration... :( Did you actually reboot the server, as opposed to simply reloading your firewall configuration and stopping and restarting asterisk? I've known some moderate to severe weirdnesses

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client <-- OpenVPN --> Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type:

> Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings If you're up to writing a bit of shell script, and are running on Linux, you could automate the conversion process so that it happens as soon as the recording is completed. Look at the

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

> I don't think it's an endpoint issue. I think the SIP packet headers get > over-written by the tunnel (openvpn) protocol. I'd be rather astonished if OpenVPN itself were responsible for this. As far as I know, OpenVPN doesn't do higher-level-protocol rewriting of any sort. It just provides the "bit pipe" through the tunnel. I'd suggest several other

Hi list! So, new day, new problem... I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now. I just tried to call a peer in my network, from a peer not yet registered. And it works... :( The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed

Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -------------- next part -------------- An HTML

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

I have the following situation.... I'm using Action Originate to originate a call for a costumer. Originate goes to a context that call the dial application. Before the application (Dial using the G option) to be invoked i'm setting the variable cellphone like this: [firstcontext] exten => s,1,Set(CDR(cellphone)=${CELLPHONE}) exten => s,n,Dial(IAX2/user:pass at

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > Don't use Port 5061, your SIP-port should be always even like 5060, > 5062, 5064 or 5066. Could you please explain why? I see in /etc/services, that 5060 is the port for SIP and 5061 for SIP-TLS, but I don't find anything for the other ports... Thanks Luca Bertoncello (lucabert

hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP