similar to: Rollover oddity

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

Hello, It's been noticed that mailing list monthly rollovers didn't happen because the machine was down when the cronjob to do the rollover would normally have fired. I will trigger them by hand now... upshot is that I'm not bothering to refile the first five days of May in the correct slot. I sincerely hope that no one cares :-) Monty --- >8 ---- List archives:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go

I am trying to perform a "rollover" when the primary number is busy. This is coming from a POTS line. Apparently I need call waiting on the POTS line as I get immediate busy from the FXS if I don't have it. So my question is this. I have an Aastra 480I CT. The call forward when busy here seems pretty straight forward. Choose the mode as busy enter the extension in the forward number

Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a

Anyone experience ring oddities with extensions.conf rollovers? Let me summarize... One of my extensions.conf file is built to ring during the day, ring/go to voicemail after a certain time: [main-aa] exten => s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1) exten => s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) exten => s,3,Dial(SIP/201,25,tr) exten =>

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why

https://bugzilla.samba.org/show_bug.cgi?id=1529 Summary: 32bit rollover problem rsyncing files greater than 4GB in size Product: rsync Version: 2.6.2 Platform: x86 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org

I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence and virtually no ability to use DTMF in IVRs. I really wish the TDM22B worked, because I much

I was hoping that rolling over to the next trunk would be simple, but it doesn't appear to be so, especially for a newbie. So I'm looking for a simple way where if I get a busy on the first outgoing trunk, I can do something to get connected to the next one. Perhaps something like the big boys do and dial 9 first? I'm guessing a custom dial plan might do that but I haven't

I'm having problems with a TDM22B. The FXO modules work fine. Both FXS modules fail to initialized. The error messages I see when the module loads: Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. I already RMA'ed a TDM22B because of this problem. Now that the replacement shows a similar issue, I want to see if anyone else can think of something to

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

It appears that part of the problem could be that when logger rolls over the mongrel log file. There is a mutex synchronization issue where the various mongrel instances all try to write to the new log file and block each other. This keeps them in a continual Application 500 Error state. Has anyone run into this problem? With log rotation turned off the problem does not appear. - Jared Brown

I just got my TDM22B, but the package did not include anything else, like datasheet. 1. Where can I download it? 2. Which outlets are FXS/FXO ??? (depending on the green /red card positions) 3. Does it need the extra power?? bye Ronald