similar to: Channels in use?

I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to

I have 4 TDM T1's going in to a IVR system. The IVR messages are recorded .wav format - The system appears to crap out at about 40 calls - Would using GSM or some other format help save CPU cycles? Using 1.2, Dual Xeon and 2GB ram TIA -- Barton Fisher Innovative Communications 714-228-5400 Ext 5410 http://www.icpage.com -------------- next part -------------- A non-text attachment was

Is it possible to have a different verbose level full log than to console output? I'd like to keep console verbose at 1, but now full log is at 1 also. Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url :

> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326

I've tried every combination I could find on the net and so far there is no joy The thing is I can do this update from the command line: Maybe some new eyes might find the answer? exten => update,1,MYSQL(Connect connid localhost root password dax) exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\

How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/3b34d38d/attachment.htm

Does anyone have all the Telco intercept prompts (numbers and such) with voice inflections to simulate number referrals and disconnects I could download? TIA, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080522/08b001ce/attachment.htm

I was looking at these SYMBOL NETVISION II NP-3010 VoIP TCP/IP WIRELESS PHONES - I know they have been discontinued. Am I asking for trouble to buy some of these for use on Asterisk? TIA Bart

We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are

I have some old PC's I want to build as a test box - It's up and running OK now. Now I installed a TDM400P and there is nothing I can do to get the card to come up. My guess is the box is not PCI 2.2 compliant or does it need to be to see the card? Thanks, Bart Here's what I know: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips

Does any have or can point me to /ACR/ Anonymous Call Rejection message I can download? The one I found was not not too clear. Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090916/c3b682d5/attachment.vcf

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

I just tried this with CentOS 5.3 as well, and got exactly the same symptoms and dmesg output. (As a point of comparison, Ubunu 8.04 on my work laptop is able to access the drive.) Obviously "not detected" is a misapprehension, though I'm puzzled why "lsusb" doesn't show it. The device is there even though the partition table can't be read. ---------- Forwarded

This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack Bart

I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP. I have one user that is having problems once he connects to asterisk. He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk) which goes to my asterisk IVR. If he presses a dtmf during any message, the press is ignored unless the press was a #, 0 or *. Otherwise, he needs to wait for the

I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Attached is the log. I'm building with: clang version 10.0.0 (https://github.com/llvm/llvm-project/ 3a6da1122b990386edeba0987d0d1fdc9c8dc53d) Target: x86_64-unknown-linux-gnu Thread model: posix On some Ubuntu-like distribution. I also ran with ASAN once and it found a bunch of leaks in bin/tco. Best, -- Mehdi On Tue, Apr 7, 2020 at 4:36 AM Richard Barton <Richard.Barton at

Hi all F18 merging has finished so commit access should be back to normal. Thanks Rich > -----Original Message----- > From: llvm-dev <llvm-dev-bounces at lists.llvm.org> On Behalf Of Richard > Barton via llvm-dev > Sent: 9 April, 2020 16:08 > To: llvm-dev at lists.llvm.org > Subject: [llvm-dev] F18 upstreaming Now! > > Hi all > > We are about to merge F18

I don't remember the details of the particular case where we encountered this, but I think the loop started with the condition check and ended with an unconditional branch back to the beginning. -- Krzysztof Parzyszek  kparzysz at quicinc.com   LLVM compiler development -----Original Message----- From: Philip Reames <listmail at philipreames.com> Sent: Thursday, May 30, 2019 3:00 PM

Hello All, I am looking for an R library/function that allows the specification of a custom link function in a linear mixed model. I've been using glm() in library MASS to fit fixed-effect models with a custom link but my study design demands mixed models. Any suggestions on the best R library/function to achieve this would be greatly appreciated. I have tried, to no avail, to