similar to: Problem with * not detecting hangup on FXO and VM going into an infinite loop

I think the problem it's in your dialplan, extensions.conf: ; voicemail management [voicemail] include => misc exten => 6245,1,VoiceMailMain2() exten => 6245,2,Hangup() Check the last line, I have the same problem and was because I wrote wrong the Hangup instruction... Regards! Date: Thu, 10 Jun 2004 09:54:32 -0600 From: Chris Hirsch <chris@base2technology.com> To:

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Hey all...for whatever reason my caller id doesn't appear to be working. My setup is simple (Wildcard FXO and thats it) and I'm just expecting the Caller ID to show up on the console. I'm seeing this: *CLI> -- Starting simple switch on 'Zap/1-1' NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID failed checksum NOTICE[262161]: File chan_zap.c, Line

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Hey all...I'm brand new to * and I want to convert my home into a pbx type setup. I've figured out that I want a Wildcard X100P to bring my single POTS CO into my Linux box. My problem is that I'm sure sure what I need to do to get my analog phones connected up into a structured phone system. It *looks* like I can go the route of the Cisco Analog -> VOIP for about $100 on ebay.

Hey all...I'm just getting my * setup and right now all I have is an FXO but no FXS. I wan't to get rid of telemarketers by having * pick up the phone if there is no CID present, give the caller the Zapateller tones and then ask the user to input their phone number via Privacy Manager (yes I realize that this won't get us any where given that I can't re-ring the phones without FXS

I'm working on migrating my home POTS phone system into an Asterisk PBX. Currently I have my FXO and FXS setup and working great. Zapateller is nice! I'm working on my voicemail now and currently have my 3 mailbox answering machine plugged into my FXS along with all my other analog phones. I'd like to slowly migrate away from the FXS answering machine in favor of Asterisk Voice Mail.

To the list ... I got the IAX2 stuff simplified & working (for now). See my earlier posting to the list. Now, here's a question for you all. I found a posting by J Todd where he gives BW utilization for various IAX2 codecs with trunking on. Now, the number of calls I can sustain over an IAX channel, obviously is going to be determined by the capacity and state of the physical pipe.

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all...I can't seem to figure out *exactly* what needs to be done when I can't flash over to an incoming callwaiting call on FXO from an FXS card. Right now if I get a callwaiting call from the FXO and hit flash nothing happens. I've been over the archives and google and it appears that this is possible and I guess it can be done with *0 but I haven't found the config on

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

Hi, all. I know this is ancient history, but I have a couple of DOS machines that host older device programming hardware. I've been able to access an older version of Samba for years without incident. Last weekend I upgraded my server to Ubuntu 18.04, which provides Samba 4.7.6. Unfortunately, after hours of frustration I find I'm unable to connect from any of the older machines.

*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards

On Mon, 24 Dec 2018, Luke Barone via samba wrote: Thanks, Luke. Unfortunately that makes no difference. Still getting: Error 5: Access has been denied at the MS_DOS console. > Try adding "server max protocol = nt1" in the [global] section? > > On Mon, Dec 24, 2018 at 1:59 PM Steven Hirsch via samba < > samba at lists.samba.org> wrote: > >> Hi, all.

Now that Solaris 10 11/06 is available, I wanted to post the complete list of ZFS features and bug fixes that were included in that release. I''m also including the necessary patches for anyone wanting to get all the ZFS features and fixes via patches (NOTE: later patch revision may already be available): Solaris 10 Update 3 (11/06) Patches sparc Patches * 118833-36 SunOS 5.10:

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

I would add the below: -Recommend using CentOS 4.0 -Use squid rpm, no tar (this is for new users I'm guessing). -Recommend using etherape and iptraf (available as rpms) for a graphical overview of traffic. http://etherape.sourceforge.net/ -Recommend the use of chkrootkit, and TCP Wrappers (at the least put ALL: ALL EXCEPT PARANOID in /etc/hosts.allow) to protect servers. -Provide some

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So there are no command line options like in oggenc to do artist and misc tags? Sure if you've got a solution that I don't have to do from scratch then post away! :-) Thanks! Chris benny k. wrote: >hi chris, > >On Thu, Mar 14, 2002 at 09:22:06AM -0700, Chris Hirsch wrote: > >>Hey all..I'm brand new to FLAC and learning to use it. Can somebody tell >>me how