similar to: BUSYDETECT_* flags

2016-09-06 17:48 GMT+02:00 Tzafrir Cohen <tzafrir.cohen at xorcom.com>: > On Tue, Sep 06, 2016 at 06:37:52AM -0600, George Joseph wrote: > > On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > > > > Where should core file be created when Asterisk is run as a daemon by > > > asterisk user and group ? > > > Is there a setting I

I think were getting closer: I did: - I edited /etc/default/asterisk to include : AST_USER="root" AST_GROUP="root" # systemctl daemon-reload # systemctl start asterisk # ps aux | grep asterisk root 3602 7.1 2.5 60332 26012 ? Ssl 16:00 0:03 /usr/sbin/asterisk -U root -G root -g # rasterisk # pkill -SEGV asterisk Then console showed: Segmentation error (core

Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected,

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

Hi Alexey, Please post the output of 'lsusb' and 'megatec_usb -a sven_625 -u nut -DDDDD'. On Monday 15 June 2009 14:32:16 you wrote: > Package: nut > Version: 2.4.1-3 > Severity: normal > > I have a SVEN Pro+ 625 ups that uses megatec_usb driver. > In ups.conf I describe it like: > [sven_625] > driver = megatec_usb > port = auto > desc =

Hi guys. I have a XP100P Clone , and the busydetect dont work for me.. PSTN---Asterisk---Sip---Asterisk----PBX Any call from pstn side dont disconnect ... I have no disconnect supervision and busydetect dont work... Please Help me. Zapata.conf [channels] echocancel=yes usecallerid=no hidecallerid=no rxgain=0.0 txgain=0.0 signalling=fxs_ks callprogress=no context=entrada channel=>1

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

Hello all, I am using samba-3.0.10-1.fc2 and I HAD 2 directorys called music and Music. I checked the contents and they were slightly diffrent, so I deleted the older one. Turns out I deleted the wrong one. OK the after beating me about the head with a Floppy repeatedly screaming BACKUP BACKUP BACKUP. Can some one tell me please if there is a way to restore my files? 11 GB of mp3s is a

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.

Hi There, Thanks for all your suggestions. I have now compiled asterisk from cvs running on FD4. I have performed all the suggested configurations: > busydetect=yes ;changed 17.03.04 from no > busycount=7 ; added as above > for me the distro asterisk package didnt hang up properly on busy > > signal. I > needed to download the source and uncomment BUSYDETECT_MARTIN

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

On Wed, 1 Mar 2017 17:48:47 +0100 Tony Peña <emperor.cu at gmail.com> wrote: > server role = dc > server role = active directory domain controller > i'm correct ? Nearly, but you should only have one 'server role' line and the second line is the correct one. > > ---- > > on include shares.conf is all share directorys...i got 47 shares... > so

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at

When I enable callprogress I get this error message... (when I call, it will ring forever but asterisk is acting as if it DID pick up the line ... but it never did... ) I'm beginning to think that asterisk can't hangup the line when in voice mail and this, whatever action I'll take... Actually, callprogress doesn't seem to work at all... (in my case) Mar 24 23:37:28

hi All Any body already setup asteriks call routing to Cisco AS5300 with SIP Server ? i need informations sample config for that, or can show how to route docs . thanks Dirgan --------------------------------- Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm analyzing freepbx extensions. When creating ivr with freepbx, it writes like this: exten => 1111,1,Answer exten => 1111,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID) exten => 1111,n(USERCID),Macro(user-callerid,) exten => 1111,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME}) exten =>

Hi iam using Asteriks 1.2.17 Server Side ( provider Side g729) clients side gsm when iam calling, iam getting lot of errors like below and lot of voice breaks Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec gsm. Use RFC2833 any suggestions ram -------------- next part -------------- An HTML attachment was scrubbed... URL: