similar to: migrated to new ver on voip connection vs1 server voicemail problems

I can not figure out what the heck is going on. I went back to my old version and I still get errors when the voicemail system tries to load any of the greetings, unavail messages, etc. the normal voicemail prompts work, but any user recording don't work. Leaving a new message appears to work, but the system wont replay them, it throws errors. Here is an example of the errors: Oct 11

Whenever we call IBM, the call counter on the phone never starts and in the CLI the zap channel never gets the answered signal from the PRI. See below. -- Executing Dial("SIP/5933-645d", "Zap/g1/18004267378") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/18004267378 At this point, I am in IBM's menu system. However the call never

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

Thanks for the reply. Forgive me for being na?ve, however have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk

I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is possible to customize the email message sent to the pager email address. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216 agoss@ntad.com

If anyone could tell me what this error is all about, I would be very grateful. Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/INBOX': Operation not permitted Oct 12 01:42:53 WARNING[2724]: app.c:1109 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/5933/Old': Operation not

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug

Hi everyone, After reading the external file into R it will become a R dataset. Now my querry is i want to dispaly only character variables from that dataset. please help me in this aspect. if possible send me the code. it will be very helpful for us. Thank you in Advance. -- View this message in context:

Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss > Sent: Monday, October 10, 2005 5:58 AM > To: Asterisk Users

Hi all! I've got a machine setup to be an "RSYNC Server", i.e. running rsync in daemon mode waiting for connections from various other machines on my network. This machine is running Debian Sarge and rsync 2.6.3. For the past several days, I've been getting notices like this in my backup logs: ============================================================================ =====

Question for all: Right now i have a Table in a mySQL DB that has a row called Ingredients. When the data is entered into the DB its enter like so from a text area: 1 1/2 lbs. beef top sirloin, thinly sliced 1/3 cup white sugar 1/3 cup rice wine vinegar 2 tablespoons frozen OJ concentrate 1 teaspoon salt 1 tablespoon soy sauce 1 cup long grain rice 2 cups water 1/4 cup cornstarch 2 teaspoons

Hi all, I am new to R. I downloaded and installed R 2.15.1 I tried typing R.exe --help at the DOS Command line C:\", but I keep receiving: [quote] R.exe is not recognized as an internal or external command [/quote] I tried many variations of R.exe --help, but roughly the same response Any ideas? thx w -- View this message in context:

Hi, Please consider the example below. How can I manage to overlay the points the way I want in the second case? Thanks, Joh library(ggplot2) # Modify data to match "real" case myDiamonds <- diamonds myDiamonds[["clarity"]] <- as.character(myDiamonds[["clarity"]]) myDiamonds[myDiamonds[["clarity"]]=="I1","clarity"] <- 1

Hi, I am using levelplot, and would like remove from each panel (condition) its unused x levels. e.g. Remove from panel vs=1 the cyl level=8. data(mtcars) levelplot(mpg~factor(cyl)*factor(gear)|factor(vs)) Thanks for your help, Ronny -- View this message in context: http://r.789695.n4.nabble.com/Lattice-levelplot-remove-unused-levels-per-panel-tp4656087.html Sent from the R help mailing

Following is a log from an attempt to record and playback a file in wav49 format from an AGI script. COMMAND: stream file aa/after_the_tone "" 0 RESULT_LINE: 200 result=0 endpos=41920 RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')} COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 "#" 20000 0

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

This is really doing my head in! For some reason, my asterisk box can't playback beep.wav. I have this extension defined in my internal context: '10001' => 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Record(/tmp/asterisk/10001:gsm) [pbx_config]