similar to: Problem with parking

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

Thank you for the reply. On 8 Dec 2010 at 13:38, Danny (Danny Nicholas <danny at debsinc.com>) commented about RE: [asterisk-users] Configuring Softphone: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary Kuznitz > Sent: Wednesday, December 08, 2010 1:27 PM > To: Asterisk

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

Hi, I have the following dialplan. Problem is, if the user authenticates, * starts counting as billable seconds even if i hangup the phone before the called party answers..And also as disposition.. it accepts all calls authenticated as 'answered' If i commentout the authentication line everything works as it should be. How should i use authentication that, it should accept it as aswered by

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

We are currently working to trunk from a Nortel 81C to an Asterisk Server 1.4 running on Red Hat Linux. We have two PRI trunks which work with the exception of the clock slips, which is causing the Nortel to reset the PRIs once a hour. Thanks for any suggestions. 81C MSDL Asterisk Digium TE110P REQ prt TYPE adan dch 10

Hi I'm running CentOS 6.2 with samba-3.5.10-114 , and LikewiseOpen 6.1 . How do I fix these errors ? Mar 16 20:25:43 nzhmlfpr05 winbindd[2556]: [2012/03/16 20:25:43.639871, 0] winbindd/idmap_tdb.c:287(idmap_tdb_open_db) Mar 16 20:25:43 nzhmlfpr05 winbindd[2556]: Upgrade of IDMAP_VERSION from -1 to 2 is not possible with incomplete configuration Mar 16 20:25:43 nzhmlfpr05 winbindd[2556]:

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

> Dear list, > > I have some data on a geneaology, here is a subset: > warmerge[1:11,c(1,6,25)] > Warrior SibID birth.year > 1100 3793 2013 1926 > 4 2013 2024 1934 > 1094 3769 2024 1918 > 632 2747 2037 1928 > 176 2083 2039 1944 > 187 2085 2039 1949 > 192 2086 2039 NA > 495

Hi all, I''ve been having trouble for a long time with backgroundrb processes that suddenly vanish without a trace. What happens is that at some point I discover that all the backgroundrb processes are suddenly gone. Nothing special is seen in any of the log files. This has happened intermittently for a long time, and I was hoping that upgrading to 1.0.4 would somehow help me out, but

https://bugzilla.mindrot.org/show_bug.cgi?id=2552 Bug ID: 2552 Summary: ssh -X and "ForwardX11Trusted no" break most applications, distros turn on "ForwardX11Trusted yes" Product: Portable OpenSSH Version: 7.2p1 Hardware: All OS: All Status: NEW Severity:

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay

On 28.05.2014 15:27, Laine Stump wrote: > On 05/27/2014 09:07 AM, Jianwei Hu wrote: >> Hi All, >> >> I have one netcf question, please help me to resolve it, thanks. >> >> I can set a IFF_RUNNING flag to a bridge device which are no interface device attached. What status of a flag on a bridge device in current kernel?(w/o interface), is this a new change in kernel

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")