similar to: How can I know if a user is busy or not connected?

One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I am planning on dabbling with some VOIP providers. I was thinking of Teliax first. My thinking is that the first LD call would go to teliax and the second (etc.) calls would go out to the PSTN. I could then verify bandwidth and quality to decide when to add more trunks and to Internet connections. I have been doing some concept testing with FWD for toll free calls, but I am using 393 as a

Good morning My situation is as folowing I have a numer that connect to my asterisk I configured another phone to transfer to this number So when somebody call me he will be transffered to the number which asterisk connect to i.e my asterisk connected phone is not the originated number My question is: How can I know if the call has been transfered to me and what was the number originated by

Did anybody figure out how to make dial detect a busy on a zaptel channel on a pri interface when using overlap dialing? According to the documentation dial should return to priority n+101, if the called party is found to be busy. I can see a DISCONNECT message with "user busy" coming from the network when I turn on pri debugging, but the dial application does not seem to notice.

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

We all know you can monitor multiple devices in one hint, and it shows busy if any device in the group is busy. This is good for a user with multiple devices, but not useful for teams where any person could take a call, like a customer service group. Does anyone know if it's possible to have a hint with multiple devices which only shows busy if every device is busy? -- Carlos Alvarez

This is almost certainly not an Asterisk-specific posting, but due to my inability to find a VoIP-focused Cisco list, I'll post here in the hopes of finding a more diverse user community. I am using a Cisco 7960 (version 6.0 SIP firmware) with Asterisk, and have been experiencing situations where the phone locks up. "Locks up" means that the bottom part of the screen

Hi people, long time without ask nothing... I need know if exist any way to get the users "windows" home directory instead of unix home directory with some command, wbinfo or smbclient. I know that info is stored in some place. Im making a autosyncHome script for linux machines. My intention is sync windows homes with linux homes for users of ubuntu and debian. Thx on advance Guys.

On 27/08/2019 14:08, Juan Ignacio via samba wrote: > Hi people, long time without ask nothing... > > I need know if exist any way to get the users "windows" home directory > instead of unix home directory with some command, wbinfo or smbclient. > > I know that info is stored in some place. Yes, by default it is in the Windows registry, unless you explicitly set the

Yes i have something like that on the user. But my objective is from the linux client machine get that attribute or info to mount or sync the windows home. I have that line on my script for example: rsync -avzX -e "ssh -o StrictHostKeyChecking=no -i /home/public/id_rsa" --progress -m --chown=$USER:$GROUP root at 10.20.1.5:/domain/samba/jfk/homes/staff/adscripcion/$USER/

On 27/08/2019 14:54, Juan Ignacio via samba wrote: > I set the homeDirectory directly in the AD object. > For it i like know if i can access to that info throw some tool. > Have you set it something like: \\fileserver\users\username If you have, then you only need a root preexex script and these lines in smb.conf: ??? winbind use default domain = yes ??? template homedir =

I resend this message, as it was not posted on the list first time I send it.... Dan ----- Original Message ----- From: "Dan" <dtoma@fx.ro> To: <asterisk-users@lists.digium.com> Sent: Friday, June 27, 2003 10:13 AM Subject: Re: [Asterisk-Users] Basic Asterisk questions - personal coments > > Why is it that most users who don't understand threaded email is on

I feel this is in error some place. If I call a sip device that is not registered or not connected at the time. Asterisk will send that call to voicemail to busy not unavailable. Is there a way to correct this? Ariel Batista Kasi International - Computer Networking Ph: 305-574-6721 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, vmkit requires to know the size of a jitted method before emitting the method. This allows to allocate the correct size for the method. The attached patch creates this functionality when the flag SizedMemoryCode is on. In order to implement this functionality, i had to virtualize some MachineCodeEmitter functions. Is it OK to commit the patch? Thanks, Nicolas --------------

John Todd wrote.... > Cisco has an 802.11 phone called the 7920, which is apparently > shipping now. It is very expensive (>$550 USD) and only runs SCCP at > the moment, which is Cisco's proprietary VoIP protocol. However, if > it falls in line with some of Cisco's other high-end VoIP equipment, > that means it should have a trailing-edge SIP image running by

Hi, Two questions. 1) How are you computing size of the method being jitted? 2) Why not simply add the functionality of allocating emission buffer of specific size to MachineCodeEmitter instead? Thanks, Evan On Mar 30, 2008, at 12:05 PM, Nicolas Geoffray wrote: > Hi everyone, > > vmkit requires to know the size of a jitted method before emitting > the method. This allows to

Trying to update actionmailer gem, and it''s complaining about the version of actionpack not being correct. C:\eclipse\workspace>gem update actionmailer Updating installed gems... Attempting remote update of actionmailer Install required dependency actionpack? [Yn] Y ERROR: While executing gem ... (Gem::GemNotFoundException) Could not find actionpack (= 1.12.4) in the repository

> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <dave@toad.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > All, > > I know you can use, say, a 2620 w/2 port FXO card as > a SIP gateway. Clearly you can use the 5300, 5800, and > MGX8850 too. > >