similar to: A solution for SIP and NAT

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

https://bugs.freedesktop.org/show_bug.cgi?id=36174 Summary: Xorg crashing in nv44 card in 3D apps Product: Mesa Version: 7.10 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

I have recently begun experimenting with Asterisk, and have been mightily impressed by its capabilities and flexibility. I have run across one problem, however, that challenges my ability to use it as a production system. My Asterisk box has a public Internet IP, and works great with SIP (ATA 186) clients that also have public IP addresses. Unfortunately, most of the locations that I would

----- 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

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

I have been trying to get the Personal Information Manager Sidekick to run on Wine and Crossover. Our company is trying to move over to ubuntu(still fairly new) and after two weeks of searching i have yet to find a contact manager with the functionality and flexibility of sidekick so we are trying to get it working in Ubuntu. The program runs fine but then instantly crashes when i attempt to

All, Please make a note of the new dial-in number for the bi-weekly OpenSFS Benchmarking Work Group meetings. Our new dial in number is: 877-709-0823 And the new participant passcode is: 4840841 Our next meeting will be on September 14th, 2012 at 11:30 AM Eastern. Main OpenSFS BWG goals are: * Research primary I/O workloads in high performance parallel file systems configurations * Provide

All, Please make a note of the new dial-in number for the bi-weekly OpenSFS Benchmarking Work Group meetings. Our new dial in number is: 877-709-0823 And the new participant passcode is: 4840841 Our next meeting will be on September 14th, 2012 at 11:30 AM Eastern. Main OpenSFS BWG goals are: * Research primary I/O workloads in high performance parallel file systems configurations * Provide

http://bugs.freedesktop.org/show_bug.cgi?id=23847 Summary: kernel BUG when using nouveau Product: xorg Version: 7.4 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy: shiningxc at

----- 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:

Hello, I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a

Quick question re. NAT traversal. I understand how sitting behind a NAT could cause problems for a SIP UA. The SIP UA would create SIP mesages using IP addresses from inside the network (i.e. 192.#.#.# or 10.#.#.#) and these IP addresses are of course unnavigable for the recipient. What I don't get is why don't web browsers suffer the same problem? A web brower behind a NAT sends an

> Still trying to get NAT working. Try adding a canreinvite=no. Nabeel

Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer

Hi all. I'm trying to setup Asterisk to act as a purely SIP PBX for Internet based VoIP. I've got it configured with with a couple of users in sip.conf like so: [andrew] type=friend username=andrew secret=<secret> host=dynamic defaultip=192.168.26.21 dtmfmode=inband Calls addressed as 'sip:username@asterisk.server' or 'sip:extension@asterisk.server' work fine

[global] workgroup = IIG netbios name = TUX server string = Samba Server interfaces = 192.168.0.10/24 update encrypted = Yes logon path = \\TUX\profile\%u logon drive = H: domain logons = Yes os level = 65 preferred master = Yes domain master = Yes wins server = 127.0.0.1 wins support = Yes