similar to: Net2Phone equipment and different VOIP providers

I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized"

Hi! Net2Phone is getting a common SIP status code, "404 Not Found," when trying to place a call to our Asterisk server. We're hoping someone on the list can shed some light on why this is happening. We can process a call from Asterisk to Net2Phone without any problems. Net2Phone sends the INVITE but immediately gets the "404 Not Found." The "To:" field

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

When I try call to net2pohe sip service in my debug I look next: ---------------------------------------------------- We're at 192.0.0.0 port 27916 Answering with preferred capability 1 Answering with preferred capability 2 Answering with preferred capability 256 Answering with capability 4 Answering with capability 8 Answering with capability 16 Answering with capability 32 Answering with

Hi - The answer to this question may help to resolve larger questions about intrinsics and vectorization that were discussed at the dev mtg last week, but let's start with the basics: Which, if any, of these is the canonical IR? ; ret = x < y ? 0 : x-y define i32 @max1(i32 %x, i32 %y) { %sub = sub nsw i32 %x, %y %cmp = icmp slt i32 %x, %y ; cmp is independent of sub %sel = select

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

Hello, My server is Mandrake 10.1 eth0 is WAN with static IP connected to 512k DSL eth1 is LAN. I am using squid proxy for internet with NSCA auth. I am able to send and recieve mails. One of the client system wants to be able to make net2phone calls. As of now he is not able to. Howto allow net2phone calls ? Thanks Varun

Hi All, As Net2Phone don't permit more than one session per account, I configured about 10 sip trunks and configure multiple trunk routing but once the first trunk is used I cannot make additional calls, I also cofigure my dial plan in other way using the chanisavail command but still not working. The chanisavail command configuration is correct as I can make calls using other trunk than

Has anyone had any success running net2phone on wine. I tried it but I recieved an error message: ]$ wine net2phone.exe Invoking /opt/wine/bin/wine.bin net2phone.exe ... /usr/bin/wine: line 380: 6945 Terminated tail -f $log_name Wine failed with return code 2 Even if I can't make it work I'm really interested to understand what's happening. (though net2phone is basically the

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

As a newbie, can anyone advise me if Asterisk can route international calls to a US based service such as Net2Phone so we can take advantage of the internet and save on calls? That would be my main reason for an Asterisk based PBX. Chris Mason masonc@masonc.com Box 340, The Valley, Anguilla, British West Indies Tel: 264 497 5670 Fax: 264 497 8463 Cell: 264 235 5670

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

Thanks, Hal and Matt for the feedback. As usual, my instincts about canonicalization were probably wrong. :) I thought that @max1 vs. @max3 would be viewed as an unknowable trade-off between reducing the dependency chain and the pseudo-canonical min/max form, so we'd add intrinsics, and defer that decision to the backend. I'll wait to see if there are any other arguments presented.

I know SAM2 (or rather, SAM3 now) has this functionality built-in: http://spacialaudio.com/ It is disabled by default and difficult to step though. Not to mention the cost of the entire application. What I found worked the best for me is to have my website output to a file whatever song somebody clicked on. A script would check the file every few seconds, and then execute my media player with

hi all since three days we've got our personal webserver. now i've installed samba 2.0.5a on it. while we were perparing the machine we set the net-settings like this: ip 192.168.1.23 network 192.168.1.0 broadcast 192.168.1.255 netmask 255.255.255.0 now we moved the machine to the outher side of the firewall with the following settings ip

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

I would like to push a file, /home/cfg/temp, from client to a server with rsync in daemon mode to /n2p/cfg directory. How should I set the filter/include/exclude parameters on the server side rsyncd.conf file so server will disallow receiving any files from the client other than /home/cfg/temp? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a Linux 2.6.16 machine with two gigabit ethernet adapters that I would like to use for testing Ethernet equipment---right now just an Ethernet cable, but later switches, bridges, etc. I tried assigning the two adapters IPs of 192.168.50.1 and 192.168.51.1 and then setup the routing table to direct 192.168.51.0/24 traffic to the first adapter, and 192.168.50.0/24 traffic to the second,

On Sun, Jan 18, 2015 at 1:07 PM, Charles Lepple <clepple at gmail.com> wrote: > On Fri, Jan 2, 2015 at 7:31 PM, hyouko at gmail.com <hyouko at gmail.com> wrote: >> Before I merge it (well.. provided that it works..), I'd like to know >> a couple of things more: >> 1. does the load.off command ('S00R0000') work? > > load.off works. > >> 2.