similar to: SIP Problem!

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. sip.conf [general] port=5060 bindaddr=192.168.10.195 disallow=all allow=alaw allow=ulaw [101] username=101 type=friend secret=1234 host=192.168.10.195 context=sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] [incoming] exten =>

I am very thankful to you people for helping me as much i imagine but i still need your help, problem is that i am not be able to dial from my analog phone conected to fxs card to my sip phone i change my configs but still no result. sip.conf [general] port=5060 bindaddr=192.168.10.193 allow=all [101] username=101 type=friend secret=12345678 host=dynamic context=from-sip

Hello *'s, i am using Realtime Sip drivers but its not working here is my configs: extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends => mysql,asterisk,sip_friends res_mysql.conf [general] dbhost =

Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder was empty i am compling several times but no luck what's the problem i compiled in the order of zaptel,libpri,asterisk. I send some traces of my asterisk's compilation

hello *'s, Astcc not workin what is correct format for defining 1-database 2-brands 3-trunks 4-routes i define all these things but not workin may be i define in wrong format.I have FXO card installed.can anyone implement it and also my sip phone generates very loud noise wat is that i tried several settings but not hear any voice just noise. sip.conf [general] context=from-sip port=5060

Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

Hello, Im writing my post graduation thesis in SYSTEMS INTEROPERABILITY and im talking about M$ wine and crossover. Can anyone pointout some good literature to begin my work. I want to talk about what needs to be done in M$ programing enviorment so that everything will run smothly in wine. thanks

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

Hi Mahdi, Did you get a chance to verify this fix again? If this fix works for you, is it OK if we move this bug to CLOSED state and revert the rebalance-cli warning patch? -Krutika On Mon, May 29, 2017 at 6:51 PM, Mahdi Adnan <mahdi.adnan at outlook.com> wrote: > Hello, > > > Yes, i forgot to upgrade the client as well. > > I did the upgrade and created a new volume,

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

Hi all, Is there a command that emulates net view? I want to dump a list of all NetBIOS hosts actively on the LAN. Thanks, -- Christopher Barry Manager of Information Systems InfiniCon Systems http://www.infiniconsys.com office:610.233.ISIS (4747) direct:610.233.4870 cell:267.879.8321

Great news. Is this planned to be published in next release? Il 29 mag 2017 3:27 PM, "Krutika Dhananjay" <kdhananj at redhat.com> ha scritto: > Thanks for that update. Very happy to hear it ran fine without any issues. > :) > > Yeah so you can ignore those 'No such file or directory' errors. They > represent a transient state where DHT in the client process

Could you disable stat-prefetch on the volume and create another vm off that template and see if it works? -Krutika On Fri, Oct 6, 2017 at 8:28 AM, Lindsay Mathieson < lindsay.mathieson at gmail.com> wrote: > Any chance of a backup you could do bit compare with? > > > > Sent from my Windows 10 phone > > > > *From: *Mahdi Adnan <mahdi.adnan at outlook.com>

OK. Is this problem unique to templates for a particular guest OS type? Or is this something you see for all guest OS? Also, can you get the output of `getfattr -d -m . -e hex <path>` for the following two "paths" from all of the bricks: path to the file representing the vm created off this template wrt the brick. It will usually be $BRICKPATH/xxxx....xx/images/$UUID where $UUID

Great, thanks! Il 5 giu 2017 6:49 AM, "Krutika Dhananjay" <kdhananj at redhat.com> ha scritto: > The fixes are already available in 3.10.2, 3.8.12 and 3.11.0 > > -Krutika > > On Sun, Jun 4, 2017 at 5:30 PM, Gandalf Corvotempesta < > gandalf.corvotempesta at gmail.com> wrote: > >> Great news. >> Is this planned to be published in next

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

Hi, Thank you for your reply. Lindsay, Uunfortunately i do not have backup for this template. Krutika, The stat-prefetch is already disabled on the volume. -- Respectfully Mahdi A. Mahdi ________________________________ From: Krutika Dhananjay <kdhananj at redhat.com> Sent: Friday, October 6, 2017 7:39 AM To: Lindsay Mathieson Cc: Mahdi Adnan; gluster-users at gluster.org Subject: Re:

The fixes are already available in 3.10.2, 3.8.12 and 3.11.0 -Krutika On Sun, Jun 4, 2017 at 5:30 PM, Gandalf Corvotempesta < gandalf.corvotempesta at gmail.com> wrote: > Great news. > Is this planned to be published in next release? > > Il 29 mag 2017 3:27 PM, "Krutika Dhananjay" <kdhananj at redhat.com> ha > scritto: > >> Thanks for that update.