similar to: Asterisk -> static nat -> laptop w/siproxd -> cisco 7960

http://www.pcphoneline.com/skype "The VPT1000 is NOT a simple last generation USB phone audio device but is rather a next generation integrated gateway and USB phoneset with simultaneous dual mode Skype and SIP calling support. Skype is not forecast to have "SkypeIn" available until June 2005 but you can have the capability now via its built in SIP capabilities." Is this a

In that example you could make outgoing calls only correct? (since incoming likely needs port forwards) I guess the questions becomes "how often are you going to do that to justify the extra $100 or so you going to pay for a wifi sip phone?" Paul Fielding (paul.fielding@shaw.ca) wrote: > > I think some people are missing the point. You can't throw your cordless >

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this

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

----- 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

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

I'm interested to hear how folks are handling NAT SIP routing issues "in the wild" for commercial use. Are you using a commerical SIP-aware NAT router solution? If so, what? Are you using a software SIP-proxy like SER or siproxd? If so, which? Do you set everything to "canreinvite=no" in sip.conf? Any comments about real-world implementations would be welcome. Thanks

Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: -- Remote UNIX connection -- Executing Dial("SIP/2001-ffd4", "SIP/2002") in new stack -- Called 2002 -- Got SIP response 486 "Busy here" back

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Might be a stupid question, but is there a default gateway set on the 7960? -----Original Message----- From: Paul Mahler [mailto:pmahler@signate.com] Sent: Thursday, December 18, 2003 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT? I have a 7960 running behind a firewall running NAT. From a telnet session to the 7960, I can't ping

----- 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:

Well I think this system is back on 3.5. How do I tell? Have not used it in a while... I need a NAT for some quick testing and this box was available. Only a 6gb drive, so I can't install Astaro (which I have licenses for). So is there a simple way to turn on NATing? Should I upgrade to 4.2? This box is behind a firewall, so security risks are not the issue. This time.

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

I'm using a Grandstream IP phone to call someone through our asterisk pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses 2x ZAP-HFC cards. When I call someone, if the call isn't answered and then I hang up, I get "487" coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten =>

I'm having an interesting problem with a Cisco 7960 phone, and two Asterisk servers. I'm not sure if this problem is specific to the 7960, or even to Asterisk for that matter. Here's the scenario. I have an * server at one location with a public IP address (i.e. not behing NAT). I have a second * server and 7960 phone at another location. This one is on a private LAN, and uses NAT to

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Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the