similar to: HELP ME!!!! Asterisk don't do calls

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

I've upgraded Asterisk from CVS, spandsp and app_txfax and app_rxfax but i'm still unable to send/receive faxes :-(. I'm using amp_fax to send and this is what i get from logs: Sep 6 11:02:52 VERBOSE[10750]: -- Attempting call on Zap/g1/666 for application txfax(/var/tmp/ast_fax-1125997371.10240.1804289383.0|caller) (Retry 1) Sep 6 11:02:52 DEBUG[10750]: Dialing

Hi everyone, not exactly security, but might give the wrong impression about the state of your security... Greetings, Jan-Philip Velders <jpv@jvelders.tn.tudelft.nl> -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- | Nederlandse Linux GebruikersGroep : http://www.nllgg.nl | -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- _---------- Forwarded message ----------

i have searched through the list archives and google, and can't find anything similar. client is running OSX 10.2.6 server is openbsd 3.2 and/or linux 2.4.x rsync on both sides is version 2.5.5. when i try to rsync a directory, all files get overwritten on the server side with 0 file length. here is an example (server is .5): [beldridg@rover]$ rsync -a -vvv -e ssh tool-etc/

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

hi, i'm testing asterisk i would like to log some agents in a queue and see how works the system. i've added to queue.conf Code: [700] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext= queue-thereare= queue-thankyou=queue-thankyou queue-callswaiting= music=default monitor-join=yes monitor-format= maxlen=0 leavewhenempty=no joinempty=yes

Thank you all very much: it solved my problem. exten => 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1) ... [novm] exten => _X.,1,Macro(exten-vm,novm,${EXTEN}) ... The remaining dialplan is not "my" specific, is the standard dial plan provided by FreePbx, which I integrate in ...._custom.conf file thanks, Andrea Chi ricevesse questa mail per errore e' gentilmente

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Alex, ping? There was a thread about moving Risc-V out of experimental but I think it didn't go anywhere? Separately, do the listed patches sound okay for merging? Thanks, Hans On Fri, Jan 25, 2019 at 4:40 PM Bruce Hoult <brucehoult at sifive.com> wrote: > > In https://llvm.org/svn/llvm-project/llvm/branches/release_80 I find > that RISCV is still in

Is there a way to run a macro in a GotoIfTime statement ?? from the wiki documentation it seems not, but...... I would like to do something like this: ......... 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567)) it does not work, as expected from documentation any workaround to call an extension WITHOUT vm (also if vm for that extension is present...) as a consequence of a Time

Hi all, I have a problem with logrotate and I don't know why. In /var/log/messages appears these errors: Apr 8 15:01:01 plzfnsm02 logrotate: ALERT exited abnormally with [1] Apr 8 16:01:01 plzfnsm02 logrotate: ALERT exited abnormally with [1] Apr 8 17:01:01 plzfnsm02 logrotate: ALERT exited abnormally with [1] Apr 8 18:01:01 plzfnsm02 logrotate: ALERT exited abnormally with [1] Apr 8

See this thread: https://www.redhat.com/archives/libguestfs/2020-June/thread.html#00012 This commit also adds a regression test of vddk password=- and password=-FD. --- tests/Makefile.am | 4 ++ plugins/vddk/vddk.h | 1 + plugins/vddk/reexec.c | 43 ++++++++++++- plugins/vddk/vddk.c | 2 +-

On 21 March 2018 at 20:07, Alex Bradbury <asb at lowrisc.org> wrote: > On 23 November 2017 at 09:38, Alex Bradbury <asb at lowrisc.org> wrote: >> On 14 November 2017 at 16:03, Alex Bradbury <asb at lowrisc.org> wrote: >>> Dear list, >>> >>> At the RISC-V BoF at the LLVM Dev Meeting and the longer working >>> session the day after,

I agree with David's sentiment. The second method appears to be easier to follow. IMHO, this would be easier for external users that desire to modify the backend for their own custom extensions/instructions. On Fri, Aug 18, 2017 at 5:05 AM, David Chisnall <David.Chisnall at cl.cam.ac.uk > wrote: > On 18 Aug 2017, at 10:55, Alex Bradbury <asb at asbradbury.org> wrote: >

I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? ---------------Logfile-------------------------------------------- Nov 14

On 23 November 2017 at 09:38, Alex Bradbury <asb at lowrisc.org> wrote: > On 14 November 2017 at 16:03, Alex Bradbury <asb at lowrisc.org> wrote: >> Dear list, >> >> At the RISC-V BoF at the LLVM Dev Meeting and the longer working >> session the day after, those of us working on RISC-V with LLVM decided >> it would be worthwhile to schedule regular