similar to: Zap channel occasionally misses dialing thefirst digit

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

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

-----Original Message----- From: Peter Svensson [mailto:psvasterisk@psv.nu] Sent: 25 November 2004 10:54 >> They work just fine if your pstn provider is at all serious. If not, >> switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during power outage I was simply speculating on the

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

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><DIV>I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing.&nbsp; The process works to the point of calling the desired number and plays the message.&nbsp; The problem is that the message starts playing almost immediately, so if 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, 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

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

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 **

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

I have a problem that is probably so "doh" I will be embarrassed. However, I have spent all evening on this with no success: I have the following setup (asterisk cvshead as of today) 10 Channel EuroISDN<=>Asterisk<=>Meridian What I can do: Call from outside into the asterisk, dial an extension, and pass through to the meridian. WooHoo. What I can't do: Call from

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

Hi, I have a polycom reboot script which sends a NOTIFY with check-sync. It worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone has the same problem? Thanks, Richard

I've run into a problem with multimarkdown, and would like advice: From Fletcher's doc: > > Unlike PHP Markdown Extra, all definitions are wrapped in <p> tags. First, > I was unable to get Markdown not to create paragraphs. Second, I didn?t see where it mattered - the only difference seems to be aesthetic, and I > actually prefer the <p> tags in place. Let me

Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt

Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Regards, Stuart -- No virus

Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

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