similar to: MOH Problems.

Hello. Is it possible to change the defined moh sound file within an extension? I have: exten => 18,1,Answer exten => 18,n,Wait(3) exten => 18,n,SetMusicOnHold(durchwahl) exten => 18,n,Dial(SIP/118,15,m) exten => 18,n,Hangup Now i have the situation someone calls and my phone is ringing while moh (durchwahl) is playing. When i pickup the call and press the hold button during

Hello. Is it possible to "mix" musiconhold music and playback voices? What i want to do is something like this: A person calls a number, gets a playback voice while in background music is playing. The configuration i use at the moment don't do what i want. Someone knows how to do it? Thanks in advance. exten => 18,1,Answer exten => 18,n,Background() exten =>

Hello. Just installed the Beronet BN4S0 card. But i can't connect to my ISDN Line. misdnportinfo gives (what does ":Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib" mean?): best regards and thanks t. asterix asterisk # misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001

Hello. I have problems to integrate an isdn phone. I don't know why but the isdn phone rings only once and than it looses its connection to his base station. I can make a call from the isdn phone to an VoIP Phone inside my network but when i pick up the phone the isdn phone also crashes. misdn.conf: [ntport1] ports=5 context=isdn-telefon msns=* extensions.conf: exten =>

Hello. Someone knows what cause 127 mean. The phone that i'm calling rings once and than the connection interrupts: P[ 5] --> l3id:10040 P[ 5] --> cause:127 P[ 5] --> out_cause:127 P[ 5] --> state:ALERTING P[ 5] --> Channel: mISDN/5-1 hanguped new state:CLEANING P[ 5] $$$ CLEANUP CALLED pid:3 best regards -- Thomas Stein knowledgeTools? ....damit Sie sehen, was Sie

Hi. Asterisk 1.4.12.1. I get a lot of message like this. Someone knows what this message mean? Do i have to worry about it? [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/152 at local-f137,1' not posted [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Agent/152' not posted [Nov 7 15:24:25] NOTICE[31247]: cdr.c:434

Hello. I have a problem recieving fax without a callerid. Somehow the script i'm using fails and i don't know how to fix it. Does anyone have an idea how to solve this? Here an example of a working fax transmission: >fax2mail v2.0 > Triggered on Tuesday, May 29 2007, at 10:38 AM > $1 = CallerID number of fax sender = 02365207150 > $2 = CallerID name of fax sender = >

Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI> moh reload voip*CLI> 1

Hello. Does someone know what "order field" means in followme.conf? The Doku says: number=> <number to call[&2nd #[&3rd #]]> [, <timeout value in seconds> [, <order in follow-me>] ] So an example would be: number=> 123&124&125,10,? It would be nice if someone could enlighten me. cheers t.

Hi All, I am having problems with some SIP peers. I seem to loose registration. If I reload SIP the registration comes back. They usually stay registered for about 2 days before they drop. The problem is not all of them drop usually just the list 2 in the list. The other strange thing is that the 2 the do drop their registration do not occur at the exact same time. It could be many hours

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

hello i was searching for solution to problem (sip->h.323). any one from this list asterisk mailing have any idea how to fix it. i am getting error when i try to call from sip to h.323 user i am successfully registering my asterisk box with gnugk. but when i try to call to h.323 openphone on working on GnuGatekeeper, asterisk is not routing it to GnuGk. i am getting the following error. do

I'm trying to install on a 233Mz PII running RH7.1 kernel 2.4.2 The "astinstall" script pulled files from CVS ok and compiled with no warnings. When I try to run "modprobe wcfxo" and "modprobe wcusb" I get the following identical set of error messages: 15. Creating sample environment...SKIPPED ================= Installation Completed! ====================

I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue with it. We can call 911 which is routed out these new trunks, and it goes to the 911 center, but they are not getting the ANI and hence "no record found". Our LEC is Embarq, and they say they can see the call come in and send: KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST I turned on all the debug

Dear Centos-Virt: Help! Guest=XP Pro, SP2 Host=CentOS5; # uname -r; 2.6.18-8.1.8.el5 VirtualBox-1.4.0_21864_rhel5-1.i586 http://www.virtualbox.org/download/1.5.0/VirtualBox-1.5.0_24069_rhel5-1.i586.rpm.run I have two nic: eth0 is the internal network; eth1 connects to a DSL modem. I also have a fully functioning DHCP server on eth0 (works perfectly with Parallels for Linux). The directions

forgot to mention Gluster version 3.10.6 On 18 October 2017 at 13:26, Alastair Neil <ajneil.tech at gmail.com> wrote: > a short while ago I experimented with tiering on one of my volumes. I > decided it was not working out so I removed the tier. I now have spam in > the glusterd.log evert 7 seconds: > > [2017-10-18 17:17:29.578327] W [socket.c:3207:socket_connect] 0-tierd:

a short while ago I experimented with tiering on one of my volumes. I decided it was not working out so I removed the tier. I now have spam in the glusterd.log evert 7 seconds: [2017-10-18 17:17:29.578327] W [socket.c:3207:socket_connect] 0-tierd: Ignore failed connection attempt on /var/run/gluster/2e3df1c501d0a19e5076304179d1e43e.socket, (No such file or directory) [2017-10-18

https://bugzilla.samba.org/show_bug.cgi?id=10179 Summary: massive test failure on OSX 10.8.5 under Homebrew Product: rsync Version: 3.1.0 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at samba.org ReportedBy: chdiza at gmail.com

https://bugzilla.mindrot.org/show_bug.cgi?id=3207 Bug ID: 3207 Summary: Match blocks ignored in files processed by Include Product: Portable OpenSSH Version: 8.3p1 Hardware: amd64 OS: Linux Status: NEW Severity: normal Priority: P5 Component: sshd Assignee: unassigned-bugs at