similar to: Unusual DTMF behavior

OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if someone types an asterisk from the other end of a call, I here it forever till the call hangs up. Looks like it starts the vldtmf, but never ends it from the logs. I am using Digium 400P rev I with one fxs and one fxo module. Any way around this one? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How

Hello, I have a server running CentOS release 5.2 (Final) - Elastix 1.3-2 and I have a problem when I try to join to a conference room. When I dial the number I hear the operator voice saying "dial the room number and the # key". I push 4# but the other end doesn?t recognize and I get the operator repeating the message. In the full log I get: [Jan 28 16:54:09] DEBUG[7838]

Eeeeck! Asterisk is inserting random digits in dialed numbers. So far I've seen it insert a 2 after the STD (area) code and insert an extra 6 or 7 in the STD code. It's pretty repeatable although the inserted number changes. My Config is: Asterisk 1.4.5, Zaptel 1.4.3, Digium TE205P (rev 02). There's an ISDN PBX on the second span and a BRI euroisdn on the first. Calls from the

Is there an issue with running it with the latest from the 1.6.2 branch? I did an svn update and make install and now when somebody comes in via Skype, I get an infinite loop of: [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]: channel.c:2701 __ast_read: read() failed: Invalid argument [Apr 27 09:53:29] WARNING[10471]:

Hi. I was using asterisk 1.2 on a box with sip phones attached and a long distance T1 line as the phone provider. We did a successful test of *1 allowing one-touch recording as set in the features.conf. Because of deadlock issues I decided to try 1.4 (latest svn as of yesterday) and the deadlock went away, but when we tried to use the *1 it was sent over the bridged channel rather than being

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp->dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP & RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

http://bugs.freedesktop.org/show_bug.cgi?id=13914 Summary: xvidtune causes nouveau to crash Product: xorg Version: 7.3 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

Hi, i'm running Samba 4.2.1 compiled from source on an Ubuntu 10.04.4 LTS Server. ACL/XATTR is active and working. I tried to activate the vfs_fruite module and added the sample code: vfs objects = catia fruit streams_xattr fruit:resource = file fruit:metadata = netatalk fruit:locking = netatalk fruit:encoding = native to the share. In addition i tried the following share definition: vfs

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

Asterisk People, I'm currently using Asterisk and with a SIP voip provider and I'm having problems where DTMF input in my IVR app is getting corrupted intermittently. For example, if someone enters 1025, it may come though correctly as 1025, or it may come trough as 10025, or 100255. DTMF digits will just double up. This doesn't happen all the time. Asterisk will just pick times

Hi... In my dialplan, I have the following: exten => s,1,Background(${RECORDING}|m) exten => s,n,Voicemail(${DID_NO}) exten => 0,1,Voicemail(${DID_NO}) exten => a,1,VoiceMailMain(${DID_NO}) exten => h,1,Hangup In version 1.2, when I hit "0" during the playback, I will be directed to voicemail. But in verison 1.4, the call hangs up. [Jan 24 16:05:37] DTMF[5754]:

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2

Greetings, Cohen, Cohen, West, and Aiken 2003 (Applied Multiple Regression-Correlation Analysis for the Behavioral Sciences, Third Edition) on page 273 state the covariance matrix of the regression coefficients is provided by standard programs for multiple regression, including SAS, SPSS, and SYSTAT. How can it be calculated with R. Thank you very much. pbm Peter B. Mandeville cel:

So this issue is quite common now in WoW launchers from all operating systems, and it's gone and happened to mine. When I open the shortcut on my desktop, the client opens, and when I click on the 'play' button, it closes and nothing comes up. I've read everywhere that if you start it from the WoW.exe link in World of Warcraft folder, it'll automatically open, but when I click

Hi, I am trying to get faster search results on our webmail client(Roundcube). Besides using Lucene for FTS are there other options? Would having all mails indexed give fast results? Currently the time it takes to search 25,000mails is 4mins. If indexed how much faster are we looking at? Really appreciate if someone could advise about this. Thanks Kevin

I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", "computer-temp.sh,darwin,") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame