similar to: Net2Phone, Asterisk, and "404 Not Found"

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"

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, 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 we are a small call center with only 8 lines we use max4 and the 2-2 port gateways from net2phone . There equipment is good but we are getting hit by lower cost competition. We need to be able to compete. We have a couple of providers who are 50% less in some cases even more. So it makes sense that we would like to be able to compete . Since we have spent quite a bit of money on existing

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

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

I was wondering if anyone has encountered and overcome this situation: We've got a PRI to our Asterisk system and notice that if a call comes in from a phone on our network, both caller name and caller number are delivered in the D Channel setup message. If a call comes to our switch from off-network, i.e. the LEC, long distance, or a cellular provider, only the caller number is sent in

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

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

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a "9" and make a local call through the Mediatrix. Thanks! chris --------------

Let's say you were wanted to terminate calls onto your Asterisk system but your only available codec was G.729 and you had no control over the remote SIP proxy sending you the traffic. What would you do? Does anyone have an update on Asterisk supporting T.38 with SIP? Thanks! chris -------------- next part -------------- An HTML attachment was scrubbed... URL:

If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

Stealth Communications Announces Registry to Avoid Access Fees Posted on: 04/23/2004 Stealth Communications Inc. today announced the official launch of a registry that allows service providers routing calls over the Internet to avoid paying local phone companies access charges. The VPF ENUM Registry allows carriers to map telephone numbers to IP addresses for such things as SIP phones and

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