similar to: 3 goes and your out

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

On Wed, Nov 20, 2019 at 01:22:16PM +0200, Mika Westerberg wrote: > If (((OSYS <= 0x07D9) || ((OSYS == 0x07DF) && (_REV == > 0x05)))) > { The OSYS comes from this (in DSDT): If (_OSI ("Windows 2009")) { OSYS = 0x07D9 } If (_OSI ("Windows

On Wed, Nov 20, 2019 at 11:52:22AM +0100, Rafael J. Wysocki wrote: > On Wed, Nov 20, 2019 at 11:18 AM Mika Westerberg > <mika.westerberg at intel.com> wrote: > > > > Hi Karol, > > > > On Tue, Nov 19, 2019 at 11:26:45PM +0100, Karol Herbst wrote: > > > On Tue, Nov 19, 2019 at 10:50 PM Bjorn Helgaas <helgaas at kernel.org> wrote: > > > >

On Fri, Nov 22, 2019 at 12:30:20PM +0100, Rafael J. Wysocki wrote: > On Fri, Nov 22, 2019 at 11:36 AM Mika Westerberg > <mika.westerberg at intel.com> wrote: > > > > On Thu, Nov 21, 2019 at 11:39:23PM +0100, Rafael J. Wysocki wrote: > > > On Thu, Nov 21, 2019 at 8:49 PM Mika Westerberg > > > <mika.westerberg at intel.com> wrote: > > > >

Hi Karol, On Tue, Nov 19, 2019 at 11:26:45PM +0100, Karol Herbst wrote: > On Tue, Nov 19, 2019 at 10:50 PM Bjorn Helgaas <helgaas at kernel.org> wrote: > > > > [+cc Dave] > > > > On Thu, Oct 17, 2019 at 02:19:01PM +0200, Karol Herbst wrote: > > > Fixes state transitions of Nvidia Pascal GPUs from D3cold into higher device > > > states. > >

Hi, I'm trying to set a rule to dial out through multiple Zap groups so that, say, g0 is the cheaper POTS lines group and must be used first. However, if g0 is busy or disconnected then try dialing out g1. My g0 group is made up of 4 analog lines connected to a 4-FXO card. I disconnected the RJ-11 wires from the FXO card to simulate a line disconnection. So theoretically all calls should

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to

I am running Asterisk and using Answer machine detection with call files on a virtual Vcloud server running Centos 5.3 and LAMP. I am finding that AMD is only detecting HUMAN or MACHINE for about 30% of the calls (I sent over 50,000 outbound calls last week, and 70% said NOTSURE). I have a suspicion that the problem may be due to the timing source on virtual server when its under load delivering

Hi, I can't receive/send calls with Asterisk. Could someone please give me a few pointers on my configuration? Regards, Norman Zhang ; sip.conf [general] disallow=all allow=ulaw port=5060 bindaddr=0.0.0.0 externip=x.x.x.x localnet=192.168.22.0 mask=255.255.255.0 context=inbound-sip maxexpirey=180 defaultexpirey=160 tos=reliability srvlookup=yes register =>

Hello, I come to you beacause i have something that i dont understand : i m using udev on a debian sid with 2.6.15.1 kernel. I have created an deprecated raid at /dev/md0 when i tried doing mkfs.ext3 /dev/md0 i have got : mke2fs 1.39-WIP (29-Mar-2006) Filesystem label= OS type: Linux Block size=4096 (log=2) Fragment size=4096 (log=2) 4643968 inodes, 9277344 blocks 463867 blocks (5.00%)

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports working fine. I am unable to get the other to work. Does anybody have an example configuration to make both work. Both are registering fine but there's just no dialtone on the non working port. TIA

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

On Thu, Nov 21, 2019 at 11:39:23PM +0100, Rafael J. Wysocki wrote: > On Thu, Nov 21, 2019 at 8:49 PM Mika Westerberg > <mika.westerberg at intel.com> wrote: > > > > On Thu, Nov 21, 2019 at 04:43:24PM +0100, Rafael J. Wysocki wrote: > > > On Thu, Nov 21, 2019 at 1:52 PM Mika Westerberg > > > <mika.westerberg at intel.com> wrote: > > > >

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

Trying to find my feet here. If I wanted to connect Asterisk to a PRI and throw in a T1 Adtran channel bank into the mix for fax machines would the following work? Connect PRI line from telco to Port 1 on the Digium Wildcard TE205P. Connect Adtran TA-624-T1 to Post 2 on the Digium Wildcard TE205P also, would I need a crossover to the channelbank or is it a patch lead like the connection to the

On -STABLE as of Mon Jun 9 04:43:55 CEST 2003, "vmstat -s" shows on one of my boxes (uptime: 26 days): -1597015721 total name lookups cache hits (101% pos + 0% neg) system 0% per-directory deletions 0%, falsehits 0%, toolong 0% Weird. I'll have to cvsup again and hope this has been fixed :) -Regard, FH. -- Farid Hajji --

Hello, I would like to use AMD on outgoing calls using analog line. I tested with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other end is analog number behind another cisco/asterisk, also tested calling a mobile number with the same result. What I did: dial is done like exten => s,n,Dial(SIP/<IP gw>/<dialed number>,,M(myMacro)), which tell Asterisk to