similar to: How to make/recieve call using asterisk whenthereis a power failure?

I have tried sjphone - worked well, although I think my 3 year old IPAQ had a bit of a hard time keeping up with the pace as there was quite a delay in the speech. Probably says more about my ancient IPAQ than SJPhone. Sam Lex Lethol <lethol@gmail.com> wrote on 23/09/2004 15:31:39: > I tried the xten one and didn;t like at all.. > > Havent tried to SJPhone, but my guess is

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

Now that's quite interesting - yes, this is a two way span. I thought that the D-channel negotiation that happens before the B-channel is set up would have implicitly avoid the glare condition through signalling the intent-to-use to the resource to the remote end (a-la E&M wink start). So what seems to be happening in my case is my CPE is set for a 'Descending' b channel sequence

Dear R-list, I try to transform a mathematica script to R. #######relevant part of the Mathematica script (* p_sv *) dd = NN (DsD - DD^2); lownum = NN (L-DD)^2; upnum = NN (H-DD)^2; low = lownum/(2s^2); up = upnum/(2s^2); psv = NIntegrate[1/(s^NN) Exp[-dd/(2s^2)] (Gamma[1/2,0,up] + Gamma[1/2,0,low]),{s,sL,sH}, MinRecursion->3]; PSV = psv/Sqrt[2NN]; Print["------------- Results

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. On Thu, 29 Jul 2004, Walter Klomp wrote: > However, if I dial-in from the SIP phone to my

I'm wondering if anyone has run across that and MUCH more importantly, if the data can be recovered somehow. I'll put as much details as I can at the bottom but here's the gist of the problem: I added my wives computer (which contains 8 years worth of pictures) to the domain. When I logged into the new domain account it was empty and my wives domain users had no access so I did the

Dear R-list members, I have a problem with translating a mathematica script into R. The whole script is at the end of the email (with initial values for easy reproduction) and can be pasted directly into R. The problematic part (which is included below of course) is <--- Original Mathematica ---> (* p_svbar *) UiA = Ni (Dsi - 2Di A + A^2)/2; UiiA = Nii (Dsii - 2Dii A + A^2)/2; psvbar =

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Hi guys, How's things going ? Got a bit of a weird one here that I've been unable to solve. I have a Panasonic PBX linked to a Sirrix Quad BRI card that is running in TE (ptp) mode in a Asterisk box - this then links through Internet to another Asterisk box via IAX2. When a user on the Panasonic PBX system dials the extension of my Sirrix Asterisk box, Asterisk answers and says

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

This is yet another problem with the Nabble interface to the list. On Wed, Feb 29, 2012 at 6:21 PM, Richard M. Heiberger <rmh at temple.edu> wrote: > This line > > ?TT <- *Temp*+273.15 > makes it unexecutable. ?that is not the error you mentioned. On nabble, that variable is in bold. When it's reformatted for the plain-text email list, the formatting is converted to **

I have read most of Eicons information on Q.SIG, and I am able to load the Q.SIG protocol (instead of ETSI for example). No strange logging in divacrtl mlog. But how do I tell Asterisk to understand Q.SIG? My PBX is configured for QSIG, but I cannot see anything on my trace when trying to make a call via the S0(Q.SIG) Janne > -----Ursprungligt meddelande----- > Fr?n:

Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange

Hi, I am supportive of the asterisk, but I have some concern, though the concern also applies to traditional pbx as well. Hope someone can shine some light into it. Thanks. During a power failure situation, analog pstn lines that connect directly to the analog phones will most likely still be able to make and receive calls. However, for the Asterisk implementation, unless you have a

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

How do I get a list of the possible ciphers that are installed on the system for use in ssl_cipher_list? -- They all have husbands and wives and children and houses and dogs, and you know, they've all made themselves a part of something and they can talk about what they do. What am I gonna say? "I killed the president of Paraguay with a fork. How've you been?"

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hello All, I saw some coverage of this in the list archive but no one seems to have posted a resolution. I am using Asterisk@Home 0.06 and when I get a call from LiveVoip over IAX I dump it into my IVR. >From there the call is routed to groups based upon input. However, there is no ringback indicated to the IAX caller. Does anyone know how to resolve this problem? Thanks, Wiley

What i need is to recieve a call in a console! I mean i can call from CLI...but can i recieve calls too? If this is possible how is the console identificated and where! Actually i need to call from one Asterisc server console to another(i know what is asterisc server for, but this is a specific task)! Thanks! --------------------------------- Don't pick lemons. See all the new 2007 cars