similar to: Can get IAX connection but no SIP connection?

On Thu, Sep 05, 2019 at 03:48:56PM -0400, Vivek Goyal wrote: > This object is used both by fuse_connection as well virt device. So make > this object reference counted and that makes it easy to define life cycle > of the object. > > Now deivce can be removed while filesystem is still mounted. This will > cleanup all the virtqueues but virtio_fs object will still be around and

On Thu, Sep 05, 2019 at 03:48:56PM -0400, Vivek Goyal wrote: > This object is used both by fuse_connection as well virt device. So make > this object reference counted and that makes it easy to define life cycle > of the object. > > Now deivce can be removed while filesystem is still mounted. This will > cleanup all the virtqueues but virtio_fs object will still be around and

Greetings, My computer has these OS: 64-bit blfs linux R2.15.1 If one has downloaded a tarball for example MASS.tar.ga and unzip it Are there R commands to generate the following?:- a) the INDEX file b) 'R' files such as MASS.rdb and MASS.rdx advice would be appreciated sincerely luxInteg

Hi, I have tested ControlPlayback and grabbed files via an apache server with no issue. I want to be able to grab files via aws S3 which would require me to add some headers to authenticate. Is there any way to have Asterisk add headers or would I need a http proxy in the middle? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until

On Fri, Sep 06, 2019 at 01:03:09PM +0100, Stefan Hajnoczi wrote: > On Thu, Sep 05, 2019 at 03:48:56PM -0400, Vivek Goyal wrote: > > This object is used both by fuse_connection as well virt device. So make > > this object reference counted and that makes it easy to define life cycle > > of the object. > > > > Now deivce can be removed while filesystem is still

I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vvvvvgc), it fails to run. I get the following errors, 1st error: [app_prepaid.so] => (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 16 14:27:27 ERROR[-1085267840]: app_prepaid.c:127

Hi All, I would appreciate a lot if you could help me. I have installed Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also installed 1 FXO port card: Digium TDM400P. After loading zaptel driver I could see my digium card's led glow green. Tested with zttool that its in OK state. I have configured fxsks=4 in zaptel.conf(channel 4 cause FXO module is on port 4). I

Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack -- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack > data = hfcpci/b17 > capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00,

Hi, my zaphfc is flooding my syslog with two messages (even without asterisk running). Is this "normal"?: -------------------------------------------------- zaphfc: bchan rx fifo not enough bytes to receive! (z1=1360, z2=1353, wanted 8 got 7), probably a buffer overrun. zaphfc: dropped audio (z1=2712, z2=2695, wanted 8 got 17, dropped 9). With Asterisk running (asterisk -vvvvvgc)

Hi, I have spent a fair amount of time trying to solve this problem and tried things like disabling checksums etc. The problem is like this: 1. I boot my dom0 without xend running and I can resolve and ping google. 2. I start xend (without and domU domains up) and after a while I can no longer resolve domain names, but I can ping the public IP addresses. 3. I start up a domU (xm create -c xyz)

Hi, Apologies in advance for the lengthy email. I'm new to asterisk and have trouble with isdn4linux. The setup is very basic like this: winxp ------- asterisk -------- winxp x-lite | x-lite | pstn The hardware involved is: Compaq EVO with RH9/kernel 2.4.20-30.9. Fritz!Card PCI v2 Asterisk CVS-04/17/04-21:36:18 Basically

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

A plea to all! Has anyone had any success with two or more avm fritz pci cards with either misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x? I have managed to get misdn to load under 2.6.13 and detect two cards using misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but the second card/controller doesn't answer or dial calls. But if I try misdn

On Sun, Sep 08, 2019 at 07:10:03PM +0800, piaojun wrote: > > > On 2019/9/6 3:48, Vivek Goyal wrote: > > This object is used both by fuse_connection as well virt device. So make > > this object reference counted and that makes it easy to define life cycle > > of the object. > > > > Now deivce can be removed while filesystem is still mounted. This will >

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

This object is used both by fuse_connection as well virt device. So make this object reference counted and that makes it easy to define life cycle of the object. Now deivce can be removed while filesystem is still mounted. This will cleanup all the virtqueues but virtio_fs object will still be around and will be cleaned when filesystem is unmounted and sb/fc drops its reference. Removing a

Hello I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora 5 for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card, so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone phone fails registering with Asterisk :-/ Following the "Asterisk - The Future of Telephony.pdf", here's what I did: 1. Installed Fedora 5,

On Wed, Jun 6, 2018 at 6:18 AM, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Wednesday 06 June 2018 at 12:02:38, Dovid Bender wrote: > > > Hi, > > > > I have tested ControlPlayback and grabbed files via an apache server with > > no issue. > > ControlPlayback is an Asterisk dialplan function. > > How have you integrated this

Hi, I'm a newbie to the list, but have been screwing around with Asterisk for the last 6 months or so (on a purely experimental basis so far). I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm unsure where the line is drawn in terms of Linux issues or Asterisk issues. At present, I have to manually start Asterisk from the command line, but I'd like to