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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" (Can forward the full * sip log and ata186 log if it would help) Many thanks Mark -------------------------- INVITE sip:...

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: <http://lists.samba.org...

...255.255.255.255 5060 Unmonitored 100/100 192.168.5.144 D 255.255.255.255 5060 Unmonitored tmp*CLI> sip no debug SIP Debugging Disabled when I call in, no packets even hit my server. Thanks, r kurt x wrote: >Russell, > >Do you have a SIP trace of the N2P call. I would like to see if N2P >is getting the Invite and what it is replying with. > >Kurt > >

Thank you for the comments on my n2p sip issues, I'll post again when (if) I get it going.

...INVITE but I'll dig further on that. Has anyone connected Asterisk to a different SIP proxy and used SIP to communicate between the two? Can anyone further explain why our Asterisk is not replying to Net2Phone's INVITE? Here is the entry from our sip.conf file: [net2phone3] context = n2p-in host=<Net2Phone's IP Address> disallow=g723.1 allow=g729 type=friend dtmfmode=rfc2833 Thanks in advance! chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040910/68e40f2a/attachment.htm

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

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, Which fax-modem would you pick if you had to test fax capabilities ? For instance, before releasing a new PBX system offering fax connectivity, you would like to make sure you "comply" with most fax machines and protocols. As you can't afford you buy and maintain tens of such fax machines nor can't afford to test by hand each protocol, it's tempting to buy an

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