similar to: Vitelity offline?

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070324/8bb0be73/attachment.htm

JohnG <mcsjgs@cox.net> wrote: > I run OS X 10.3.7 on a PowerMac MDD G4 on a cable broadband connection. > I have reason to think my system has been tampered with. Security > features in Mac OS X have been left unlocked (Preference Pane - Users) OSX is substantially different from FreeBSD (even without netinfo) despite having some of the same source code. I doubt you'll find

Does anyone know of a way to rate limit ssh connections from an IP address ? We are starting to see more and more brute force attempts to guess simple passwords "/usr/sbin/inetd -wWl -C 10" is nice for slowing down attempts to services launched via inetd. Is there an equiv method for doing this to sshd? Running from inetd has some issues supposedly. ---Mike

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

I am not sure if I had a compromise but I am not sure I wanted some other input. I noticed in this in my daily security run output: pc1 setuid diffs: 19c19 < 365635 -rwsr-xr-x 1 root wheel 204232 Sep 27 21:23:19 2003 /usr/X11R6/bin/xscreensaver --- > 365781 -rwsr-xr-x 1 root wheel 205320 Dec 4 07:55:59 2003 /usr/X11R6/bin/xscreensaver It was the only file listed and I didn't

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ============================================================================= FreeBSD-SA-03:12 Security Advisory FreeBSD, Inc. Topic: OpenSSH buffer management error Category: core, ports Module: openssh, ports_openssh,

Hi all, this is the first time I post to this list. If I am wrong, sorry. I have been trying for three weeks to update openssl-stable, but I cannot beacause of this message: ===> Cleaning for openssl-stable-0.9.7k # # this ports conflicts with your base system # please undefine OPENSSL_OVERWRITE_PORT # and use WITH_OPENSSL_BASE=yes instead. # *** Error code 1 I do not know how to act. Bye

Heya.. Yesterday someone "attacked" by box by connection to several ports.. In other words, a simple portscan.. yet, since my box has "log_in_vain" enabled, so it tries to log everything to /var/log/messages, since the logfile got full and the size went over 100K, it tried to rotate the log to save diskspace. (Apr 16 21:00:00 omikron newsyslog[32137]: logfile turned over due

After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed the problem on the IAX protocol. They told me that as of Asterisk 1.4 the IAX protocol went downhill and many carriers (like VoicePulse) are discontinuing support for IAX. Is this correct? We are all heading for SIP? Thanks, MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a grown list of IPs that I am "deny ip from ###.### to any". Infected machines, hackers, etc.. Is there a way to have this list outside of rc.firewall and just read it in?

I'm dying to use the 1.0 capability to use more than one mysql user pasword database. However, I'm too chicken to use 1.0-test in a production environment. Does anyone know if there is a planned release date for 1.0? Or can anyone say how risky it is to use 1.0-test?

Hi All- ? I have successfully routed calls into our asterisk system from several DID providers in the USA, but for some reason I'm having a problem getting Vitelity to work. ? We are using the IAX protocol, and the symptom is that only about 50% of the calls terminate properly into my asterisk system - the rest get a busy signal.? The ones that do not come in don't show up at all on

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to

I wanted to try AsteriskNOW plus a few others to see which I can wrap my head around the quickest. The issue so far is I can't figure out how to use my Vitelity account with it. I went so far as to put their Asterisk configuration in the sip.conf file but still no joy. Any pointer as to where to search? I found a few threads in the AsteriskNOW forums and one thread from last year on

Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication. [global] type = global debug = yes [transport1] type = transport bind = 0.0.0.0 protocol = udp *local_net=<yourlocalnet I.E. 10.10.10.10/24 <http://10.10.10.10/24>>external_media_address=<your public ip address>external_signaling_address=<your public address>*