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Search found 410 results on 17 pages for 'voip'.

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  • Skype companywide global contacts list

    - by Martin
    We are a medium sized company based across several sites and with a number of home workers. We have more or less settled on Skype as our defacto method of communication. At the moment the only pain is ensuring that everybody has all the other employees added to their contact list. Can be a real pain when a new employee starts and they have to send details to everyone else and vice versa. Is there a solution that allows us to manage a central contacts list that we can push out to new/existing users?

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  • Skype companywide global contacts list

    - by Martin
    We are a medium sized company based across several sites and with a number of home workers. We have more or less settled on Skype as our defacto method of communication. At the moment the only pain is ensuring that everybody has all the other employees added to their contact list. Can be a real pain when a new employee starts and they have to send details to everyone else and vice versa. Is there a solution that allows us to manage a central contacts list that we can push out to new/existing users?

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  • email to sip voicemail

    - by alfredwesterveld
    Hi all, I don't know if this is the correct place to answer this question, but here it goes. I have been googling for a cheap email to sip voicemail service. That is because I have got a Linksys/Cisco SPA-941 phone which has a led which will light up when a new message comes in inbox(somebody calls me). So what I want is the following. I want the e-mail(title only is enough) recorded(By computer voice) and sent to my phone which I can playback when the led lights up. Like I said above I was unsuccessful googling for a service like this and I hope somebody knows if this service exists. Many thanks, Alfred

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  • Asterisk terminating outbound call when picked up, sends 'BYE' message

    - by vo
    I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway the setup does not appear to have issues registering and setting up the call, RTP packets go both ways and you can hear the ringing from the other side. However it appears that when the call is picked up or thereabouts, the incoming RTP packets cease. Upon closer inspection with Wireshark, there are these particular packets that seem to be the cause: trunk->asterisk SIP/SD Status: 200 OK, with session description asterisk->trunk SIP Request: ACK sip:<phone>@trunk:6889 asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 [..about a dozzen RTP packets in/outbound..] trunk->asterisk SIP Status: 200 OK, CSeq: 104 Bye [..outbound RTP continues, phone is silent..] Then the inbound RTP packets cease, however the asterisk logs dont show any activity at this point. The last entry reads 'SIP/ is answered SIP/'. Then when you hangup the extension, you get asterisk->trunk SIP Request: BYE sip:<phone>@trunk:6889 trunk->asterisk SIP Status: 481 Call Leg/Transaction does not exist My trunk peer settings in FreePBX are: username=<user> fromuser=<user> canreinvite=no type=friend secret=<pass> qualify=no [qualify yes produces 401/forbidden messages] nat=yes insecure=very host=<sip trunk gateway> fromdomain=<sip trunk gateway> disallow=all context=from-pstn allow=ulaw dtmfmode=inband Under sip_general_custom.conf i have stunaddr=stun.xten.com externrefresh=120 localnet=192.168.1.1/255.255.255.0 nat=yes Whats causing Asterisk to prematurely end the call and still think the call is in progress? I have no idea where to look next.

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  • TeamSpeak 3 Disconnects

    - by ArchUser
    I've recently had a few random TS3 mass disconnects and I'm am curious to know where I may find any applications that can help me determine the cause of any types of TS3 server disconnections as we plan on having many more users in the future. I run an almost empty VPS (OpenVZ) server with an ArchLinux template on it. I have 1.5/2GB of RAM, 2GHz of CPU and plenty of hard drive space, to run for the most part, just my TS3 and a low traffic apache web server. This is what I am investigating. 2011-02-04 06:07:05.130343|INFO |VirtualServer | 1| client disconnected 'Valamoor'(id:224) reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.131338|INFO |VirtualServer | 1| client disconnected 'Kevrow'(id:19? reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.191849|INFO |VirtualServer | 1| client disconnected 'scuba'(id:200) reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.192633|INFO |VirtualServer | 1| client disconnected '[Ash] Setna'(id:75) reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.193350|INFO |VirtualServer | 1| client disconnected 'Akiris'(id:254) reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.194047|INFO |VirtualServer | 1| client disconnected 'Marcus'(id:25? reason 'reasonmsg=connection lost' 2011-02-04 06:07:05.194726|INFO |VirtualServer | 1| client disconnected 'Guthry'(id:275) reason 'reasonmsg=connection lost' 2011-02-04 07:18:50.327071|INFO |VirtualServer | 1| client disconnected 'Valamoor'(id:224) reason 'reasonmsg=connection lost' 2011-02-04 07:18:51.339018|INFO |VirtualServer | 1| client disconnected 'Marcus'(id:25? reason 'reasonmsg=connection lost' 2011-02-04 07:18:51.339870|INFO |VirtualServer | 1| client disconnected '[Ash] Setna'(id:75) reason 'reasonmsg=connection lost' 2011-02-04 07:18:51.340515|INFO |VirtualServer | 1| client disconnected 'Guthry'(id:275) reason 'reasonmsg=connection lost' 2011-02-05 04:55:20.797353|INFO |VirtualServer | 1| client disconnected 'JohnyRingo'(id:240) reason 'reasonmsg=connection lost' 2011-02-05 04:55:20.798517|INFO |VirtualServer | 1| client disconnected 'Maloo roots'(id:196) reason 'reasonmsg=connection lost' 2011-02-05 04:55:20.799314|INFO |VirtualServer | 1| client disconnected 'Cpt dravyn'(id:234) reason 'reasonmsg=connection lost' 2011-02-05 04:55:20.839254|INFO |VirtualServer | 1| client disconnected 'scuba'(id:200) reason 'reasonmsg=connection lost' etc... I need to determine if it is my hosting provider or my server, and what tools I can use to determine the issues. My VPS host has told me this... "I checked out the node that your VPS runs on and there is no abnormal system load, or I/O wait from the drive. I also checked the bandwidth history from the server and there have been no spikes or outages."

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  • Understanding Asterisk Server features

    - by Arham Ali Qureshi
    I need to ask few question about Asterisk 1) Does ACL mean by Access Control list here ?If yes than how could i use it? >ip show user 6001 * Name : 6001 Secret : <Set> MD5Secret : <Not set> Context : DLPN_Admin Language : AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 2147483647 Callgroup : 1 Pickupgroup : 1 Callerid : "test" <6001> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No 2) What is mean by "Require Call Token" in Asterisk Digium GIU on Create new User Panel 3) Is There any command from where i can get users VOICE MAIL password ? 4) What AMI or CLI command set call recording on or off for user ? and if i want that file to be stored on client computer not on server memory what could i do ?

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  • Open source chat protocol that handles voice & video?

    - by marcusw
    I'm looking for a chat protocol which: Has easy to use clients which will run on both Windows and Linux. Has a server which I can run myself on Linux (preferably easy to set up). Supports duplexed voice and video with minimal hassle (optional). Is open source/free software. Is there a protocol that fulfils these requirements?

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  • 3CX behind UT7.1 using a callcentric.com SIP account

    - by Corey
    Has anyone had any luck with getting 3CX working behind UT7.1 with a SIP account from callcentric.com? I am willing to reset my current UT box back to defaults, and start from there. I have a static public IP assigned to the external interface. My internal addressing is 192.168.76.0 . My 3CX box has 192.168.76.17 . Would anyone be willing to give me a step by step of changes to make in UT / 3CX. I currently have my UT box unplugged, and have replaced it with a Linksys unit. I have port forwarding setup for… TCP/UDP 5060 to 192.168.76.17 UDP 9000-9049 to 192.168.76.17 … and everything works great. I also have additional external IPs available if that helps.

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  • SIP server ("gateway") for joining accounts

    - by Tomas Srna
    Hello, I have a phone supporting 2 accounts, but I need 4 accounts. Is it possible to install some sort of SIP server/gateway/proxy (on a linux server), that would register those 4 accounts and I would be able to connect to it as if it was 1 account? (With dialing rules, etc.) 3 of the accounts have incoming numbers. Thanks. Tomas.

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  • Reliable applicance for routing IT emergency calls (SIP and ISDN)

    - by chiborg
    We have a fairly big IT installation and our IT staff needs to be reachable 24/7. At the moment we have the following setup for "emergency" calls to our IT staff on our main Asterisk box: An incoming emergency number (connected via SIP trunk and a BRI card in case the SIP trunk goes down). When the number is called during the office hours, all the SIP phones of the IT staff are called simultaneously. When the number is called out of office hours interface, a list of mobile phone numbers is called, one after another until someone picks up. The list can be changed by the IT staff via command line script. The setup works well, but the Asterisk is heavily used in a call center, has experienced some outages and misconfigurations, each of them bringing down the IT emergency number. So we'd like to put the IT emergency call functionality on a separate device. This does not need to be a big server, it even does not need to be Asterisk, it only has one purpose and should do it reliably. It should be very low-maintenance. Any suggestions for hard- and software?

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  • Any plugins for Skype that support "Soundboard" usage?

    - by Axxmasterr
    I would like to find a program or plugin for Skype that allows you to pipe sound samples in to the outgoing audio stream when you are on a call. Ideally it would have some sort of soundboard functionality so that I could have a group of audio samples at the touch of a button. I'd also prefer something that supports mp3 but wav support will also do.

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  • How to remove static IP from Mitel 5312 and enable DHCP

    - by jimbo
    I'm not sure this is the right forum for this question -- although I'm confident I'll be told if not! -- but I've read the fine manual (at least, such a manual as I have), I've googled and I cannot get any insight into where to even start solving this problem. I have a bunch of Mitel 5312 handsets, talking to a 3300 ICP controller. Some handsets are at a remote location, get an address from my DHCP server over there, and use the Mitel "Teleworker" extension to connect in over the Internet. The remaining handsets were set up with static IPs by a BT-supplied engineer, on the same subnet as the ICP itself. So far, so good. I have one remaining teleworker licence, and need to move a handset from the home location to the remote. I've managed to boot it and configure teleworker, but I cannot for the life of me see where I tell it to forget its static IP, and make a DHCP request. Any ideas? Should I be looking on the controller, or holding magic combinations of buttons on the handset itself? EDIT: Following some advice from Robert, below, I've broken out a spare device and reassigned the profile for this user's extension to the MAC of the new phone, and a new profile to the old MAC. Unfortunately this still doesn't get me anywhere -- the new handset now asks for the teleworker install password. I suspect I'm going to have to get a Mitel engineer involved here, since I've never been given that password... Unless anyone has any great ideas?

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  • How to establish SIP connection, when SIP-proxy is required?

    - by LA_
    I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. I should use the following data to connect to the server: Auth name – 7499952XXXX User name/User ID/Display Name – nickname Authorization user name - [email protected] Domain - sip.beeline.ru SIP proxy server - msk.sip.beeline.ru I've also found the following string: [email protected]:password:[email protected]@msk.sip.beeline.ru:5060/7499952XXXX I've tested the parameters on my PC thru X-Lite and it works well (so, assume there is no any problem with the router, no need to do anything with router's NAS settings). But since I am quite new to Asterisk, I can not understand where to input all these data. Asterisk GUI doesn't have fields for proxy: Can somebody please help me with step-by-step instruction? Thank you in advance!

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  • software to allow a friend to look on your screen

    - by acidzombie24
    I want a friend of mine to review my code. I have a mic built into my laptop but i can use another software for voice chat. So i would like to have him to either view my screen (taking control might be fine) or show me his screen so he can talk about a specific piece of code. What software is good for this? We both think about security so an MS product (netmeeting? if that still exist on normal msn live installs) or an open source would be preferred. Should be free.

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  • Asterisk Connection not working

    - by Tamas Ionut
    I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something like 192.168.x.y?? ). However, when I navigated to that url from outside of VirtualBox, that url pointed to nothing. Also I am logged in as root@localhost. Should I be logged in as root@server? I have also validated the installation as described here and everything went well. I am a complete beginner at Asterisk.

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  • Soft sound problem on Sony Viao NetBook in GTalk and GMail Chat

    - by mx4399
    I have a new Sony Viao netbook and have a problem with the sound - or lack of loudness thereof. When using GTalk or chat in GMail with earphones all is OK but when using the netbooks' built in speakers the sound is very very faint (also in the GMail Chat test section). I installed the Sony sound drivers and checked all the sound settings, VLC also plays music at an acceptable level but still not as load as my old MacBook did. All settings are set at 100% - but are still too much too soft to use. Anyone got an idea?

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  • How to split registration and media?

    - by Stackfan
    I have a SIP project. Where i will have SIP server running. Server will do following: will only do routing and receive incoming calls But the audio/video will be peer 2 peer Can this be done with Asterisk? Only the media i have to split but the registration will be with Server. Tools: A) server with SIP B) One PC with SIP client C) Anoher PC with SIP client My goal is: B and C gets connected via A and audio/video packets are not via A

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  • How to establish SIP connection, when SIP-proxy is required?

    - by LA_
    I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. I should use the following data to connect to the server: Auth name – 7499952XXXX User name/User ID/Display Name – nickname Authorization user name - [email protected] Domain - sip.beeline.ru SIP proxy server - msk.sip.beeline.ru I've also found the following string: [email protected]:password:[email protected]@msk.sip.beeline.ru:5060/7499952XXXX I've tested the parameters on my PC thru X-Lite and it works well (so, assume there is no any problem with the router, no need to do anything with router's NAS settings). But since I am quite new to Asterisk, I can not understand where to input all these data. Asterisk GUI doesn't have fields for proxy: Can somebody please help me with step-by-step instruction? Thank you in advance!

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  • Shoretel Reporting Question

    - by MJ
    This might be a bit off topic, but I'll ask anyways. With Shoretel reports, the report that generates how long someone is on the phone(actively talking on the phone), if they are listed as "off hook", like right before you dial a number, does that still show up as being on a call?

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  • Asterisk dialplan context and label clarifications

    - by liv2hak
    I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the web.Can someone confirm if the IVR below is correct? Correct in the sense that if the below is used will it work.I know the IVR does not do much yet.But I am just trying to clarify my understanding. [incoming] exten => 123,1,Answer() same => n(menuprompt),Background(main-menu) exten => 1,1,Playback(digits/1) same => n,Goto(incoming,menuprompt,123) exten => 2,1,Playback(digits/2) same => n,Goto(incoming,menuprompt,123) exten => 9,1,Hangup() [main-menu] exten => n(menuprompt),Background(main-menu) exten => 3,1,Playback(digits/3) same => n,Goto(main-menu,menuprompt,n) exten => 4,1,Playback(digits/4) same => n,Goto(main-menu,menuprompt,n) exten => 9,1,Hangup()

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  • How do I configure additional phone lines asterisk/trixbox?

    - by Matt
    I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card and have 4 more analog lines coming into the Trixbox server. It still runs the 4 fine, but what do I need to do to add the additional 4 phone numbers/lines? I want it to act exactly as before, there's nothing special about the new lines. We just need more lines so that when we have 4 out of state customers call, we can have 4 more call and not get the busy signal. Trixbox CE 2.8

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  • How Do i configure sip2sip, sipdroid, pbxes, and google voice on my andriod g1

    - by Bliss
    I've been trying to configure the above for two days i used ipkall at first and it registered with my sipdroid app because i got a green status light and whenever i make a call from sipdroid it will show it in the pbxes call log but the call drops asap..i downloaded sip2sip to get another number ive gotten the new number from sip2sip now i need to configure sip2sip with sipdroid and google voice...can someone please help me my e-mail is [email protected]

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  • telephone sockets connecting them to an ATA

    - by ageis23
    We have many extension sockets in the house therefore be fairly expensive to buy an ata for each phone. Since extension sockets are daisy chained together, could I just plug it into the FXS socket instead of the BT master socket? Or is the FXS port strictly for one phone only?

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  • Mac internal mic not working with Xlite

    - by Pablo Fernandez
    I've used xlite for a while now but all of the sudden the internal mic from my mac stopped to work with it. It does call, and you can see the red bar rising but the other person can't hear a thing. The mic does work when using Skype so I think it's a problem with Xlite Things I've already tried: Reconfigure Xlite Reinstall Xlite Install Xlite Beta (After every step I rebooted the macbook) Obviously none of them worked. Thanks

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