similar to: - RE: Missing digits on TDM400P incomplete dial string - DTMF problem? - Email found in subject

Displaying 20 results from an estimated 5000 matches similar to: "- RE: Missing digits on TDM400P incomplete dial string - DTMF problem? - Email found in subject"

2004 May 07
1
Missing digits on TDM400P incomplete dial string - Email found in subject
Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com
2004 Apr 22
0
[SPAM] - Re: Adtran TA750 Noise - Email found in subject
I believe it is not fiber, but I am not sure. I am going to take one of them home tonight and hook it to my POTS line there, which for sure is not fiber. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Welter Sent:
2004 Apr 22
2
Adtran TA750 Noise
All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750
2004 Apr 24
1
snom reporting busy when it shouldn't - Email found in subject
Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original
2004 Jun 18
2
FXO Issues
All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank
2004 Apr 24
1
\ Adtran Channel Bank? - Email found in subject
Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April
2004 Apr 23
0
Adtran TA750 Noise - Email found in subject
Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks
2004 May 07
0
- Re: Routing by called interface - Email found in subject
That does work, I use that same approach to get analog extensions in a norstar system to dial a specific sip phone in *. Works really well. We then also tie the calleridname to which channel they dial out from as well. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com
2004 Apr 23
0
PSTN Call drops randomly - Email found in subject
Set busydetect=no in your zapata.conf file. That should stop the random hang-ups. If you really need busy detection, try setting busycount=8 or even 10. If you still get random hang-ups, turn off busy detection and turn on call progress. May help the situation. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From:
2004 Nov 24
3
Haven't got a clue ...
On how to even search for this "feature" as I have no idea on what it can be. I've got a meridian linked to * (by EuroISDN) which is linked to a ISDN30. I can make calls from the meridian, and receive calls into the meridian. Great stuff. However, if someone dials an invalid number, then instead of hearing a "three tone", the line just drops and goes dead. The console
2005 Jun 20
3
QuadBRI: How to set the outgoing callerid (KPN - NL)
Hello all, Recently I purchased an QuadBRI card from junghanns.net after some playing around, reconfiguring dialplans etc with the exception of 1 thing everything seems to work: I seem to be unable to set the outbound callerid. The dutch telecom operator (KPN) provided me with 4 MSN's on 1 BRI interface. In the past years I'm more then used to setting the MSN without the leading 0, this
2008 Oct 20
0
Problem in extensions.conf Configuration ${CALLINGPRES}
Dear Everybody, I have to store variable from ${CALLINGPRES} and get birth date of our client and get back to him his birth prediction as numerology (numerology digit value is between 1-9). I have also mentioned below example here suppose client's birth date is 27-01-2000 then 2+7+0+1+2+0+0+0 = 12 and then 1+2 = 3. 3 is result so client can get his prediction as this 3 digit. Please
2005 Mar 24
1
Missing CallingPres Application
I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed
2009 Jul 09
1
Connecting two Asterisk together via SIP + DISA
Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |
2010 Jun 08
0
Flac -ts differs from flac -t
Hello, I'm runing Debian Stable linux with flac version 1.2.1. Weekly, I run a cron job to test all my flac files for problems using the -ts options. I have several computers storing the same information over various raid arrays, and occasionally I do find the odd file that has had problems and can then change hard drives and re-synchronize. Anyways, I have recently encountered a couple of
2007 Apr 23
0
R: extract from a data frame
Oats[Oats$Variety %in% c("Victory", "Golden Rain"),] or subset(Oats, Variety %in% c("Victory", "Golden Rain")) Stefano -----Messaggio originale----- Da: r-help-bounces at stat.math.ethz.ch [mailto:r-help-bounces at stat.math.ethz.ch]Per conto di elyakhlifi mustapha Inviato: luned? 23 aprile 2007 9.56 A: R-help at stat.math.ethz.ch Oggetto: [R] extract from
2007 Apr 23
1
extract from a data frame
hello, I'd like know how to do to extract data from a frame for example how can I do to extract only the data where variety=victory or variety=golden rain thanks. > Oats Block Variety nitro yield 1 I Victory 0.0 111 2 I Victory 0.2 130 3 I Victory 0.4 157 4 I Victory 0.6 174 5 I Golden Rain 0.0 117 6 I Golden Rain
2007 Jan 11
4
Parked calls and the # key
I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.
2005 Apr 21
2
do not understand what to do to correct this error
---------------------------------------------------------------------------- ---- FULL RSYNC LOG ---------------------------------------------------------------------------- ---- /usr/bin/rsync -av --force --delete-excluded --exclude-from=/usr/local/etc/snapback/snapback.exclude -e ssh --delete peru.cbm.mercyships.org:/ /home/backup/peru/hourly.0/ <bunch of lines deleted> wrote 873039
2007 Dec 10
0
diferents events between ast1.2 & ast1.4 ??
Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I