similar to: Net2Phone Multiple SIP Trunk Not Working

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

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"

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! 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

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 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

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

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

Hello Everybody, I?m trying to make a work with procmail and deliver on Fedora 7. Part of flow are the following: - procmail get the mail from Sendmail - procmail ask to deliver/dovecot if have any retriction (sizer mailbox, etc) - procmail put the mail on inbox user. I had include the following line in the promail config: | "/usr/libexec/dovecot/deliver -m $DEFAULT" and the

hi, Longing for your help. I came into a problem ,Now I want to configure asterisk sip peers from MYSQL database dynamic, flolling the introduction of asterisk realtime,i set the cofiguration of sip users,but I need to cofigure sip peers too. Where I can find some infomation about cofiguring sip peers? What is the difference of configuration sip peers and configuration sip

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

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Thanks, it helped :-) I'm now building the sources and apparently my mingw installation does not support pthread and therefore examples/ParallelJIT.cpp fails: make[2]: Entering directory `/c/llvm1.8/generated-llvm/obj/examples/ParallelJIT' llvm[2]: Compiling ParallelJIT.cpp for Debug build c:/llvm1.8/llvm/examples/ParallelJIT/ParallelJIT.cpp:20:21: pthread.h: No such file or

I've two net2phone MAX IP10 with propietary net2phone firmware loaded and want to use with my Asterisk box.<BR><BR>I know that is possible to change the firmware to enable the SIP features on this phones. Any one knows where to get the firmware?<BR><BR>You can check the phone features at:<BR><A

All: I'm trying to run the cygnus-port on Win2K Server. I have the dll and the ported code (exe). It seems to run without access violation or anything. Since Windoze doesn't have a good RSH, and I'm only going to use this between my two laptops, I decided to setup an rsync server on one of them...no problem. Seems to run, shows "Starting..." in the log...good conf file,

Hello, I've noticed that when I try to reencode VBR mp3s to broadcast @128k , they are not being reencoded at all. They stream as normal VBRs I am unable to reproduce this at lower bitrates (56-112k i've specifically tested and all work fine). It does reencode all non-VBR mp3s appropriately to 128k... Any ideas why VBR reencoding would suddenly stop working at 128k? I've attempted