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  • Email sent via Google via relayhost being marked as spam

    - by Mark H
    Company email hosted by Google Apps. Company PBX in-house is Elastix. All voicemails received on the extensions of Elastix are supposed to be emailed by the CentOS server (Postfix) to the email address of the employee. Using relayhost on postfix, I am sending those emails through Google Apps (smtp.gmail.com), but some of these voicemail emails end up in the spam. Sending it through Google, and sending it to an email hosted by Google - yet there's spam. Email sent from the Google Apps interface - no complaints of it going to spam - just from the Elastix server. I've just asked our DNS domain guys to add spf records, but is that all that's needed? Some help please!

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  • ASP.NET 3.5 GridView - row editing - dynamic binding to a DropDownList

    - by marc_s
    This is driving me crazy :-) I'm trying to get a ASP.NET 3.5 GridView to show a selected value as string when being displayed, and to show a DropDownList to allow me to pick a value from a given list of options when being edited. Seems simple enough? My gridview looks like this (simplified): <asp:GridView ID="grvSecondaryLocations" runat="server" DataKeyNames="ID" OnInit="grvSecondaryLocations_Init" OnRowCommand="grvSecondaryLocations_RowCommand" OnRowCancelingEdit="grvSecondaryLocations_RowCancelingEdit" OnRowDeleting="grvSecondaryLocations_RowDeleting" OnRowEditing="grvSecondaryLocations_RowEditing" OnRowUpdating="grvSecondaryLocations_RowUpdating" > <Columns> <asp:TemplateField> <ItemTemplate> <asp:Label ID="lblPbxTypeCaption" runat="server" Text='<%# Eval("PBXTypeCaptionValue") %>' /> </ItemTemplate> <EditItemTemplate> <asp:DropDownList ID="ddlPBXTypeNS" runat="server" Width="200px" DataTextField="CaptionValue" DataValueField="OID" /> </EditItemTemplate> </asp:TemplateField> </asp:GridView> The grid gets displayed OK when not in editing mode - the selected PBX type shows its value in the asp:Label control. No surprise there. I load the list of values for the DropDownList into a local member called _pbxTypes in the OnLoad event of the form. I verified this - it works, the values are there. Now my challenge is: when the grid goes into editing mode for a particular row, I need to bind the list of PBX's stored in _pbxTypes. Simple enough, I thought - just grab the drop down list object in the RowEditing event and attach the list: protected void grvSecondaryLocations_RowEditing(object sender, GridViewEditEventArgs e) { grvSecondaryLocations.EditIndex = e.NewEditIndex; GridViewRow editingRow = grvSecondaryLocations.Rows[e.NewEditIndex]; DropDownList ddlPbx = (editingRow.FindControl("ddlPBXTypeNS") as DropDownList); if (ddlPbx != null) { ddlPbx.DataSource = _pbxTypes; ddlPbx.DataBind(); } .... (more stuff) } Trouble is - I never get anything back from the FindControl call - seems like the ddlPBXTypeNS doesn't exist (or can't be found). What am I missing?? Must be something really stupid.... but so far, all my Googling, reading up on GridView controls, and asking buddies hasn't helped. Who can spot the missing link? ;-) Marc

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  • Show dialog on outbound calls

    - by jgauffin
    I want to be able to show a dialog on outbound calls. The dialog is used to ask the user if he wants 1. dial the phone number directly 2. Dial through the PBX. If option two is chosen, i want to dial a specific number and send the dialed number as DTMF. How do I catch and stop outgoing calls? How do I get the dialed number?

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  • Using UCMA to connect to 3CX?

    - by Rodney Burton
    Has anyone used Microsoft's UCMA 2.0 SDK to connect to 3CX's free IP PBX to add voice capabilities to their application? If so, does it work? What I am trying to accomplish is having a windows form app running on 2 or more computers, and each person can connect to another person and carry on a voice conversation using their headset connected to their computer. App is in C# w/ .NET3.5 SP1.

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  • How to enable CDR on AsteriskNow 1.5

    - by Michal Niklas
    I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at least 3 pbx2*CLI> cdr show status pbx2*CLI> Call Detail Record (CDR) settings ---------------------------------- Logging: Disabled Mode: Simple Asterisk shows that CDR modules are loaded: pbx2*CLI> module show like cd Module Description Use Count cdr_manager.so Asterisk Manager Interface CDR Backend 0 cdr_csv.so Comma Separated Values CDR Backend 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 cdr_custom.so Customizable Comma Separated Values CDR 0 6 modules loaded How to enable creating CDR csv files?

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  • Does anyone provide a Skype connection service?

    - by Runc
    Is there a way of offering Skype access to incoming calls while keeping all telephony traffic over our chosen business telephony provision? Is there a 3rd party who can route incoming Skype calls to our telephone system? The business has had requests from contacts wanting to call us via Skype, but we want to keep all telephony via our PBX and phone lines as our geographic location limits our available internet bandwidth. We also prevent installation of non-standard applications on desktops and do not want to add Skype to our build. I was wondering if there were any 3rd parties that provide a connection service that would allow our contacts to call via Skype and us receive the calls via our phone system.

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  • asterisk send mute command to jukebox on incoming call

    - by Jona
    Hi, We're trialling a Asterisk Now server to take over from our ageing PBX system. One of the "nice to have" features would be the ability to pause or lower the volume on the office jukebox if an incoming call is detected. We currently run a linux jukebox which plays music out of the speakers using mpd and can be controlled by the mpc client. We can manually issue the following command to achieve this: mpc volume 20 Does anyone know how to get asterisk to execute this command or some action that we could hook into when a phone call is incoming to specific extensions?

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  • Does anyone provide a Skype connection service?

    - by Runc
    Is there a way of offering Skype access to incoming calls while keeping all telephony traffic over our chosen business telephony provision? Is there a 3rd party who can route incoming Skype calls to our telephone system? The business has had requests from contacts wanting to call us via Skype, but we want to keep all telephony via our PBX and phone lines as our geographic location limits our available internet bandwidth. We also prevent installation of non-standard applications on desktops and do not want to add Skype to our build. I was wondering if there were any 3rd parties that provide a connection service that would allow our contacts to call via Skype and us receive the calls via our phone system.

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  • Suggestions for a SOHO/Home Asterisk system

    - by James
    I would like to buy a small embedded system that runs Linux and Asterisk. I would like two FXS ports ( to plug analogue phones into ) and one FXO port ( to plug in to my real line to allow access to the POTS ). I would like it to have a USB port to hold storage for voicemail. I really want it for home use so I would like it to be under £150 ( say $250 ), given that you can buy ADSL routers for around this much can any of them be made to run Linux and Asterisk? I don't want a PC as the power usage would be too high. I am looking for something like this ASDL router but open and able to run Asterisk or another open PBX. At worst I would like a box which had one FXO and two FXS and just made them completely available over IP to a full Asterisk system on a low power Atom system.

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  • How to enable CDR on AsteriskNow 1.5

    - by Michal Niklas
    I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at least 3 pbx2*CLI> cdr show status pbx2*CLI> Call Detail Record (CDR) settings ---------------------------------- Logging: Disabled Mode: Simple Asterisk shows that CDR modules are loaded: pbx2*CLI> module show like cd Module Description Use Count cdr_manager.so Asterisk Manager Interface CDR Backend 0 cdr_csv.so Comma Separated Values CDR Backend 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 cdr_custom.so Customizable Comma Separated Values CDR 0 6 modules loaded How to enable creating CDR csv files?

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  • cant make outbound calls - asterisk

    - by deanvz
    I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: http://www.atcom.cn/cn/download/pbx/ip01/ATCOM%20IP01-User%20Manual-V1.0-EN.pdf Whenever I try and make a call, it seems that the outbound call rule that i defined does not get regarded as the default rule even though the dial plan lists this as the only outbound call rule. When dialling I see in the log file the following [Jan 1 09:10:07] NOTICE[176]: chan_sip.c:14377 handle_request_invite: Call from '6001' to extension '00765243679' rejected because extension not found. The 00765243679 is a cellular number. Am I missing a configuration in order to make outbound calls? Land line, other Voip numbers and cellular calls have been tried

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  • Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to

    - by Corey
    Does anyone know of a inexpensive NAT router that has the ability to limit access to the Internet to a specific MAC address? I know the Linksys routers have a MAC filtering feature, but it is the opposite of what I need. It allows you to block access to a specific MAC address. I need something that will block all, but allow an exception. I'm dealing with some VOIP issues in my company's network, and I think the answer is to have a separate router on the network for my PBX to use. I want to make sure that other nodes are not allowed to access the Internet via this second router.

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  • How to Switch Chrome’s Default Search to International Google

    - by Erez Zukerman
    Google Chrome’s default search engine is Google. This makes perfect sense; the only problem is that it uses localized Google – for example, Google France or Google Israel. This impacts the interface language, and sometimes even the text orientation. Here’s how you can fix this and get “international” Google results with an English interface. First, we need to figure out what search query we’re going to use. Go to Google.com and execute a simple query for a single word – say “cats”. If you get real-time results, hit Enter so that the address bar updates with the query URL. It should look something like this: http://www.google.com/#sclient=psy&hl=en&site=&source=hp&q=cats&aq=f&aqi=g1g-s1g3&aql=&oq=&pbx=1&bav=on.2,or.&fp=369c8973645261b8 If you wish to customize your search further, click Advanced Search. For example, I would like Google to annotate results with the reading level they require, so I can see what’s going to be difficult to read: Latest Features How-To Geek ETC Macs Don’t Make You Creative! So Why Do Artists Really Love Apple? MacX DVD Ripper Pro is Free for How-To Geek Readers (Time Limited!) HTG Explains: What’s a Solid State Drive and What Do I Need to Know? How to Get Amazing Color from Photos in Photoshop, GIMP, and Paint.NET Learn To Adjust Contrast Like a Pro in Photoshop, GIMP, and Paint.NET Have You Ever Wondered How Your Operating System Got Its Name? Lakeside Sunset in the Mountains [Wallpaper] Taskbar Meters Turn Your Taskbar into a System Resource Monitor Create Shortcuts for Your Favorite or Most Used Folders in Ubuntu Create Custom Sized Thumbnail Images with Simple Image Resizer [Cross-Platform] Etch a Circuit Board using a Simple Homemade Mixture Sync Blocker Stops iTunes from Automatically Syncing

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  • Google suddenly only indexes https and not http

    - by spender
    So all of a sudden, searches for our site "radiotuna" give out the result as an HTTPS link. https://www.google.com/?q=radiotuna#hl=en&safe=off&output=search&sclient=psy-ab&q=radiotuna&oq=radiotuna&gs_l=hp.12...0.0.0.3499.0.0.0.0.0.0.0.0..0.0.les%3B..0.0...1c.LnOvBvgDOBk&pbx=1&bav=on.2,or.r_gc.r_pw.r_qf.&fp=177c7ff705652ec3&biw=1366&bih=602 We only use https for the download of two specific files (these urls are resources used for autoupdate functionality of an app we distribute). All other parts of the site should be served over http. We wouldn't like to see any other traffic over https, nor any of our site links to appear in search engines as https. I'd like to address this issue. It seems that the following solutions are available: hand out an https specific robots.txt as such: User-agent: * Disallow: / and/or at app-level, 301 permanent redirect all requests (except the two above) to HTTP if they come in as HTTPS. My concern with the robots method is that, say (for some reason) google decided not to index http pages, disallowing https pages might mean that google has nothing left to index with disastrous consequences for our ranking. This means I'm inclined to go with a 301 redirect. Any thoughts?

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  • EmblaCom Oy Maximizes Database Availability and Reduces Costs with MySQL Cluster

    - by Bertrand Matthelié
    Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0cm 5.4pt 0cm 5.4pt; mso-para-margin-top:0cm; mso-para-margin-right:0cm; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0cm; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin; mso-bidi-font-family:"Times New Roman"; mso-bidi-theme-font:minor-bidi;} Headquartered in Finland, EmblaCom Oy provides turnkey and cloud-hosted voice solutions to mobile operators around the globe. Since launching the original mobile private branch exchange (PBX) in 1998, the company has focused on helping its partners provide efficient voice communications to their key business customers. The company’s voice solutions are used by millions of subscribers, worldwide. EmblaCom Oy needed to replace several database engines with a standardized, scalable, development-friendly database solution to maximize availability and cut costs. The company chose MySQL Cluster Carrier Grade Edition, which has maximized accessibility to EmblaCom’s services for its clients and their hundreds of thousands of subscribers. The initiative has also reduced, by half, the cost of the database solution installation for customers, as well as lowered maintenance and customer service costs. Read the entire case study here.

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  • Looking for some advice on the next steps to take [closed]

    - by mopsyd
    I am looking for some advice on the next step to take in development of my programming skills. I was directed here when asking this question on Stack Overflow. What I know already Have a solid grasp of xhtml, xml, php, javascript, MySQL, actionscript. Have a working knowledge of vb, and have a slight grasp of java from tinkering with a minecraft server. Some brief exposure to the Unreal Engine in college. Some skills with sql server, ms sql, office integration, etc. Also some knowledge of Asterix and PBX/VOIP. Been coding off and on since the age of 8 but I have no computer science education aside from what I have taught myself or learned from work/freelance. I work in OSX mostly, but can use/troubleshoot windows and ubuntu fluently also. Decent with both UNIX and DOS CLI. What I'm considering I'm looking to learn a scripting language to build web apps, help streamline my home server that I am building and run shell scripts. Being able to help code games later is a big plus. My Question Between java, ruby, perl, and python, which would be the best investment of my time considering what I already know and what direction I would like to take my skillset? What are good resources for your suggested direction? Thanks in advance.

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  • Change Call Screen

    - by vyana
    I need to change or customize the call screen when initiating a call on Android. After searching on google I do not find any way to do it. There is no way to send DTMF tones during a call, the idea is to send a specific number to the call screen. So when a call is made is possible to see the number to dial during a call to the PBX. I tried to putting the number in the "status bar", but the notification hide after seconds and it is not practical. I appreciate any other suggestion. Thanks

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  • Skype a-like PlugIn

    - by balexandre
    This is not really a programming question but please bear with me as I just need some push to the right road. What is and what should I search for to create a Skype A-Like plugin? I want to be able to capture the Phone Number (Web Browser, Word and Outlook for starts) and change it to present a link that the user can click to search that Phone Number in the CRM Application, given him the ability to open that client profile or make a call (we have a PBX Dial software). But I have no idea what this technology is called in order to start looking for some ideas on how to accomplish this. Thank you

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  • SDP media field format

    - by TacB0sS
    Hey, I would like to create a SDP media field with its attributes, and there are a few things I don't understand. I've skimmed and read the relevant RFC and I understand most of what each field means, but what I don't understand is how do I derive from the Audio/Video Format of the JMF, which parameters of the format compose the rtpmap registry entries I need to use. I see many times the fields m=audio 12548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv these are received from the pbx server I'm connecting to, what do they mean in the terms of the JMF audio format properties. (I do understand these are standard audio format commonly used in telecommunication) UPDATE: I was more wondering about the format parameter '0 8 101' at the end of m=audio 12548 RTP/AVP 0 8 101 Thanks in advance, Adam Zehavi.

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  • remote telnet and email

    - by Mustafa Ismail Mustafa
    This issue has been occupying my work for the last few days and I will be understating when I say its driven me up the blasted walls. Essentially, I can ping and tracert the domain jnrcs.org and the subdomains mail.jnrcs.org and mail.jordanredcrescent.org. All three mentioned point to ip address 212.38.147.97. About 4 days ago, when we registered the domain "jnrcs.org" suddenly all external connection to the mail server from outside was lost. Not just mail, but other http based port-forwarded or natted services (such as camera surveillance and pbx services). I tried good old telnet (I'm a linux user) and I get the following output: telnet> o mail.jnrcs.org 25 Trying 212.38.147.97... telnet: Unable to connect to remote host: No route to host telnet> Tracert gives me: traceroute to mail.jnrcs.org (212.38.147.97), 30 hops max, 60 byte packets 1 192.168.1.2 (192.168.1.2) 0.869 ms 0.944 ms * 2 * * * 3 * * * 4 * * * 5 * * * 6 * 212.38.128.118 (212.38.128.118) 33.875 ms 39.187 ms 7 * * * 8 * * * 9 * * * 10 * * * 11 * * 212.38.147.97 (212.38.147.97) 67.621 ms I am stumped. Other friends from all around the world can telnet no problem. What could have possibly happened to make telnet/smtp/pop/imap/http access stop? Please bear in mind I'm primarily a developer but I [am under the delusion] that I can carry my weight in IT administration :) TIA

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  • Transfering call asterisk to different context

    - by Necronet
    I have a Small and basic PBX, and with two contexts wich basicly are sales and supervisor both have different roles and privileges. I notice that it is possible to transfer call from the same context but it have been imposible to transfer anything to another context. Any insight, i am kinda a rookie on asterisk but currently there is no one else in charge... Thanks Edit This is the extension.conf [supervisor] include => from-internal exten => _40XX,1,Answer exten => _40XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _40XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _40XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _40XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _40XX,n,Hangup [sales] include => out-trunksip exten => _41XX,1,Answer exten => _41XX,n,Set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d.%H.%M.%S)}) exten => _41XX,n,Set(CALLEDNUMBER=${EXTEN}) exten => _41XX,n,MixMonitor(/tmp/Para_${CALLEDNUMBER}-${calltime}-De_${CALLERID(num)}.wav) exten => _41XX,n,Dial(SIP/${EXTEN},40,TtRr) exten => _41XX,n,Hangup and the sip.conf looks like this: [supervisor] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=supervisor mailbox=supervisor [sales] username=sales secret=ASUPERSECRETPASSWORD type=peer ..... context=sales mailbox=sales What do you suggest in order to get the supervisor with the same privileges that he already has and the sales been able to transfer calls to him

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  • Linux DHCPD Mac-Address based Groups

    - by GruffTech
    Our Current DHCPD.conf looks like the following. subnet 10.0.32.0 netmask 255.255.255.0 { range 10.0.32.100 10.0.32.254; option subnet-mask 255.255.255.0; option broadcast-address 10.0.32.255; option domain-name-servers 208.67.222.222,208.67.220.220; option routers 10.0.32.5; host Dev-ABaird-W { hardware ethernet 00:1D:09:3E:49:13; fixed-address 10.0.32.94; } ... more static hosts .... } About as basic as it gets. The old router is 10.0.32.1, our company wanted to implement a squid proxy to better monitor web traffic while at work, and if necessary block large time-wasters, IE Facebook.com. However, we've quickly realized that this change has played a mean prank on our Polycom SIP Phones. Occasionally our phones will not ring, the end recipient hears ringing (this is artificially created by our PBX) however the handset never rings. The ONLY thing that has changed in our network is the option routers line. So, Since all Polycom MAC addresses begin with 00:04:F2 would it be possible in DHCP to say any 00:04:F2:::* MAC addresses get option routers 10.0.32.1, and anything else must talk with our Gateway?

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  • dns in a small network with router and AD domain

    - by Felix
    I have a small office network with router (running OpenWRT), Windows Domain Controller (used to be 2008R2; I just backed it up and upgraded to 2012), about a dozen AD clients (3 server and windows workstation) and several non-AD clients (network printer, PBX). The problem is that the clients can't access servers by name (only by IP). I tried all kind of permutations. Right now domain controller runs DNS server for all desktops; but unless I put an entry in hosts file - I can only get by IP. I have router as DHCP server (since not all devices are on AD); and except for Domain Controller all IP addresses, including "static", are assigned by the router. Most frustrating, some servers sometimes just work! for example, I can often get to the Linux box by name (it is part of Domain using Beyond Trust Integration Services); but I can never get to SQL Server box. Seems like non-domain devices see more names than domain members... This network should be fairly typical; but I couldn't get any guidance about how to set up DNS/DHCP service to make all nodes happy. The closest is this question, but still it's different! Thanks

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