similar to: incoming calls produce multiple quarter rings and asterisk never answers.

Tip side open on the analog line? Have you taken a butt set or normal phone and attached it directly to the outside line to see if you get dial tone? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Jon Gabrielson > Sent: Friday, January 28, 2005 11:09 PM > To: Asterisk Users

I have an adit 600 with an fxo card connected to a digium T1 card. If I try to make an outgoing call and the T1 cable is disconnected, asterisks returns congested like it should. But, if the adit 600 is connected to the T1 card, the adit 600 immediately "answers" the call even if there are no physical lines attached. I even removed the fxo card and the adit 600 still

I have financial data on a a set of firms, with a quarterly period (fundamental data). The data spans 10 years, and four quarters per year. The present file (.csv) reads the Date columns as "200706" for the second quarter of 2007; "199809" for the third quarter of 1997. Is there a way I can convert it to something like "2007 Q2", "1998 Q3"? I am aware of

I don''t know why the bitmap in toolbar displays only a quarter of picture on Windows XP, but it displays OK on Ubuntu. The attachments are results both on Windows XP and Ubuntu -- flyerhzm@hotmail.com _______________________________________________ wxruby-users mailing list wxruby-users@rubyforge.org http://rubyforge.org/mailman/listinfo/wxruby-users

Hi Everyone, The FreeBSD Status Reports for the First Quarter of 2008 are now available at: http://www.freebsd.org/news/status/report-2008-01-2008-03.html Regards, Brad Davis

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have two date strings, say "1972-06-30" and "2012-01-31", and I'd like to get every quarter period end date between those dates? Does anyone know how to do this? Speed is important... Here is a small sample: Two dates: "2007-01-31" "2012-01-31" And I'd like to get this: [1] "2007-03-31" "2007-06-30"

Hela, When I call my sip phone the phone rings and when i answer it all works great however while the phone rings the caller doesnt hear the 'beep, beep' I've had this before and fixed it too bad i can't remember how I did it , Anyone any suggesstions ? some specs : Freebsd 5.3 Asterisk 1.0.6 Phone : ArtDio IPF-2000 -Armand

I am a little confused about the FXO ports on the Carrier Accesss Bank 1. The wiki, and people on the mailing list have stated that they do not have impedance matching, but Carrier Access states in their documentation that they do have impednace matching. Did Carrier Access make a typo, or does it depend on the version of the FXO card? I current have a CAC AB1, and I am having echo problems

Dear Gavin, This is really great, thank you! I created some long loops to get rid of extra months at the beginning and the end of my data but your code is great for putting it then together quarterly. thanks again, Denise On Mon, 2008-08-18 at 14:31 +0100, Denise Xifara wrote: > Thank you very much Stephen, but how will aggregate deal with months that > fall outside annual quarters? eg,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 FreeBSD Project Quarterly Status Report - 3rd Quarter 2016 As focused as we are on the present and what is happening now, it is sometimes useful to take a fresh look at where we have come from, and where we are going. This quarter, we had our newest doc committer working to trace through the tangled history of many utilities, and we

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 FreeBSD Project Quarterly Status Report - 3rd Quarter 2016 As focused as we are on the present and what is happening now, it is sometimes useful to take a fresh look at where we have come from, and where we are going. This quarter, we had our newest doc committer working to trace through the tangled history of many utilities, and we

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hi List, I have a Polycom 601 that when the user is on the phone they only hear one beep and the CID of the second incoming call is not shown. Is there a way to have the CID show up for the second call ? And a way to configure the phone to beep more often if there is another call coming in. The problem is that if the receptionist is on the phone and looking up something on the PC she some times

I have asterisk connected to an adit 600 with fxo ports. When I place a call with asterisk, asterisk will try to dial out on the first line even if the first line is already being used by someone else. Any ideas on what I'm doing wrong? Thanks, Jon.

# Dear List, # I want to characterize a time series according to its Quarter components. # My data ("a.ts": http://docs.google.com/View?id=dfvvwzr2_478cr9k4cdb)? look like: #???????????????? Qtr1????????? Qtr2????????? Qtr3????????? Qtr4 #?? 1948 -0.0714961837? 0.0101747827? 0.0654816569 -0.0227830729 #?? 1949 -0.1175517556? 0.1151378692? 0.1015777858 -0.1971535900 #?? 1950?

Thanks a lot for all the suggestions! Unfortunately, it still gives problems. Most common error message is "ast_realaudio_callback Failed to write frame" after "paying the beep". Then it says "User disconnected". Also, it doesn't react to any extension entered and doesn't do any forwarding (as it should in "exten =>

Hello - I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me. Here's the relevant info: Ingress SIP trunk: IP: 123.45.45.3456 DID's XXX-XXX-XX00-XX10 sip.conf: [general] useragent=Asterisk port=5060 context=default tos=lowdelay disallow=all allow=ulaw

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minimum usage or something. Any info as far as actual costs and/or voice quality would be appreciated.

I have asterisk set up to ring a local SIP phone (on an ATA186) for incoming calls and to divert to voicemail after 20 seconds. However, when the 20 seconds is up, asterisk answers (I know this because I have another phone on the same line as asterisk) but the SIP phone keeps on ringing for a few cycles. How can this be? Can it be stopped? I'm running last week's cvs. Iain