similar to: FXO- no dial tone- no call progressing

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

V?Mon, Oct 16, 2023 at 12:34:42PM -0700,?Jordan Erickson napsal(a): > On 10/16/23 10:37, Michael C Cambria wrote: > > I'm using icecast via Fedora 37 package and systemd service to start. > > > > I've added multiple <listen-socket> but get: > > > > "EROR connection/connection_setup_sockets Could not create listener > > socket on port

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

On 09/09/2018 07:19 AM, Daniel Walsh wrote: > sesearch -A -s httpd_t -t system_conf_t -p read > > If you feel that these files should not be part of the base_ro_files > then we should open that for discussion. I think the question was how users would know that the policy allowed access, as he was printing rules affecting httpd_t's file read access, and looking for

On Thu, May 4, 2017 at 4:21 AM, Andy Shevchenko <andriy.shevchenko at linux.intel.com> wrote: > acpi_evaluate_dsm() and friends take a pointer to a raw buffer of 16 > bytes. Instead we convert them to use uuid_le type. At the same time we > convert current users. > > acpi_str_to_uuid() becomes useless after the conversion and it's safe to > get rid of it. > > The

On 10/16/23 10:37, Michael C Cambria wrote: > > Hi, > > I'm using icecast via Fedora 37 package and systemd service to start. > > I've added multiple <listen-socket> but get: > > "EROR connection/connection_setup_sockets Could not create listener > socket on port xxx" *snip* That error sounds like it could either be an issue relating to which

I am trying to create a system as follows: incoming call --> ivr --> sent to dummy extension 1000 --> redirects user to ring group 2000 --> (ring group consists of extension 3000 and 4000) --> no answer --> sends to voicemailbox 1000 I want to do this to be able to have one extension number (and mailbox) for a person but allow them to have multiple extensions transparently

On 02/15/2018 04:08 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:03 PM, thelma at sys-concept.com wrote: >> On 02/15/2018 03:44 PM, Joshua Colp wrote: >>> On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >>>> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >>>> >>>> IN audocodes setting I

On 02/15/2018 04:49 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 7:46 PM, thelma at sys-concept.com wrote: > > <snip> > >> >> Thanks again for the hint. >> Here is the output from asterisk. >> >> The call is coming on Audocodes gateway from: pstn-4444 >> >> But asterisk display: >> Found peer 'pstn-9998' for

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3 phone number: pstn-4444 Channel: 4 phone number: pstn-9998 When I am calling " pstn-4444" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing .....

I'm attempting to set up a selenium grid on virtual machines. The default NAT is good enough for almost everything, but I need to be able to send my hub VM requests on port 4444. I've attempted to get this going by adding the following to my iptables firewall: -t nat -A PREROUTING -p tcp --dport 4444 -j DNAT --to-destination 192.168.122.107:4444 -t filter -A FORWARD -m state -d

On 02/12/2017 01:40 PM, Gordon Messmer wrote: > On 02/11/2017 08:56 PM, Robert Moskowitz wrote: >> This seems to be bug 1103439 which was 'fixed' for Centos6. >> >> What should I do about this? Is there a SELinux policy to apply or >> should I the avoid upd-ports option in Bind? > > > It looks like that bug was assigned to the selinux-policy

Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message.

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

Hi list, After reading rsync docu and doing online search, I still failed to find a simple solution for following problem: The goal is to sync two directories using rsync without running the rsync daemon on one of the two hosts. The standard shell sync does not work because of the network topology: Ssh connect to SRC-Host ^ Base host (with ssh keys) v Ssh connect to bridge

Thanks for testing - are you able to see if there's anything in the server logs? I've just committed some extra verbosity in the client's log messages that might clarify where it is exiting (patch attached). -d On Fri, 29 Mar 2019, Adam Eijdenberg wrote: > On Wed, Mar 27, 2019 at 10:04 PM Damien Miller <djm at mindrot.org> wrote: > > > > OpenSSH 8.0p1 is almost

OK, I've traced this a little further on now. I think I have followed the trail via the following items that call or refer to each other in turn: - python samba-tool ntacl sysvolreset - source3/smbd/pysmbd.c: SMB_VFS_FSET_NT_ACL( fsp, security_info_sent, sd); - source3/include/vfs_macros.h:#define SMB_VFS_FSET_NT_ACL smb_vfs_call_fset_nt_acl - source3/smbd/vfs.c:NTSTATUS