similar to: Net2Phone SIP trunk problem

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"

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

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

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

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 all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

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

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

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

What is involved in provisioning Asterisk to use a multiline analog service from our local telco? I will only have one twisted pair entering in on a OpenVox card but am not sure how Asterisk interprets and deals with two incoming calls and/or two outgoing calls? Thanks! jlc

Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel

Dear All, I have an asterisk server version 1.2.12.1 along with trixbox and I am having this nasty problem, I have a TE200P and have an E1 pri attached to it and zttool says it's OK, I have configured the whole 31 channels into one group as follow: Zapata-auto.conf: callerid=asreceived signalling=pri_cpe switchtype=euroisdn context=from-zaptel group=0 channel=>1-15,17-31

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before