similar to: DTMF Tone length

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

I came across this hack while looking through chan_zap.c. Is this still applicable for those w/ zplex cb's? Is it a maybe need or an absolutely need? Any background info would be appreciated. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977

This probably has an easy solution, but I found it yet. How can I get out of a remote console after using ssh to get into the box, making changes, reload etc. without stopping *? Thanks in advance. Sincerely, Andy Hester Consero

Hi, Please let me know if you have any ideas - I am taking wild guesses now.... Here is the situation: I put in Asterisk for a local customer. I have Fractional T-1 with 12 Voice & 12 Data. I have a T100P hooked up to a TDM Card (they call it a chanel bank although it only has 2 outputs) in a CAC unit. The unit also has a router card that runs the data side. My extensions are all SIP

Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero

Hi All, I am considering testing out some analog sets with * for a customer and thought I would ask what analog phones are in use? The customer would require the usual business functionality ie hold, conference calling, and preferably a soft key to vm and line apearances(correct terminology?) in order for secretary to see if their boss is on the phone before transferring a call, etc. I would

Is there a way to get my X100P card to dial "slower" on the line? Mine seems to dial the digits too short/fast for the switch to catch all of the digits and roughly 25% of outbound calls fail to complete. I've monitored on the line with a test set and the audio sounds clean, and I hear the X100P send out digits really fast, but sometimes the switch seems to time out and issue

Hi, I have resources available to host a portal specifically for the Asterisk system, to help correlate documentation, FAQ's and How To I am new to Asterisk, and my hardest work is in locating information on using or configuring the software. Would Mark, John or any of you feel this would be of benefit to host? I am offering a PHP-NUKE portal for this, (Unless you know of a better

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

I'm getting down to the last of my * issues ... After calling a bank, or cc processing center; you have to enter your social security number, or the cc number - followed by the # key. The lovely * voice responds "transfering" I'm sorry that was an invalade selection. Sometimes the IVR on the other end still gets the digits and proceeds; other times it breaks the IVR on the

Wondered if anybody might have some ideas about what could be causing this.... I have a T100p hooked up to an Adit 600 with 12 channels of voice off of a T-1 coming in. I have a t100p connected to a zhone z-plex with 24 fxs going to my stations. Some of the station are 2 line phones. These have 2 zap channels that are dialed when the extension is matched. Also, some extension require ringing 2

#this is my stab at - I am sure that I am missing something. If this doesn't make sense then please ask for more details. #This may show my low level of programing knowledge hester. <- c(1,1,1,1,2,2,2,2,3,3,3,3,4,4,4,4) value <- rnorm(16) x <- data.frame(value, hester.) z <- (if(x[,"hester."]==1){ x[,"value"]*6.250} else if(x[,"hester."]==2){

Good day, I am hoping that someone can assist me with a work around. I have an IVR system that I am attempting to connect asterisk to, however the IVR was written Some time ago, and requires tones to be approximately 2 seconds in length. I am using SIP Phones, Is there any way to adjust the Length of the Tones that asterisk sends out the audio path? Are there any phones that would allow

Has anyone noticed a delay in sending DTMF to zaptel devices? I have a T100P connecting to an NACT telecom switch. All calls are sent just fine, but there are 4 seconds of delay between when the channel goes off-hook & the digits are collected by the NACT switch. I also have some cisco 5300 boxes connected, but there is no delay. Any ideas? -g

Hello, I'm using Asterisk (latest CVS head) to perform outbound call as robot/testing tool for an IVR platform, with a Wildcard T100P configure as ISDN Pri. For develop the exten context script I was using a real PSTN ISDN Megalink (DMS100) to reach the platform and my script was able to correctly detected the DTMF tone send back by the platform to synchronize the script. But for load test,

Seems to have struck a small problem.. Using a t100p & Zhone channel bank.... one extension ringing another.....the following will appear WARNING[18448]: File chan_zap.c, Line 2685 (zt_handle_event): Didn't finish Caller-ID spill. Cancelling. if we are using defaultzone=au change it to us and the problem goes away..... any possible solutions ?? Gary .

Jim, One more note about precedence. It prevents a solution like the one you proposed from solving all of the problems you cited. By my reckoning, a "What comes next is for NSE" unary operator needs an extremely low precedence, because it needs to greedily grab "everything" (or a large amount) that comes after it. Normal-style unary operators, on the other hand, explicitly

I am biased against introducing new syntax, but if one is experimenting with it one should make sure the precedence feels right. I think the unary and binary minus-sign operators have different precedences so I see no a priori reason to make the unary and binary %xxx% operators to be the same. Bill Dunlap TIBCO Software wdunlap tibco.com On Thu, Mar 16, 2017 at 3:18 PM, Michael Lawrence

William, Unbeknownst to me when I sent this, Jonathon Carrol started a specific thread about unquoting and a proposal for supporting it at the language level, which I think is a better place to discuss unquoting specifically. That said, the basics as I understand them in the context of non-standard evaluation, unquoting (or perhaps interpolation) is essentially substituting part of an unevaluated

I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes with T100P cards connected via IAX2. Calls between them work fine, but when I press a key at one end, it comes out the other end as a click, with no tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing call, and it sounds fine in that case; it seems to be only IAX2 that has the problem.