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Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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Hi

I am tasked with setting up a lone staff alert button on a 5320 IP phone that will ring or alert all other phones (12) on the system, is this possible on the mitel 3300 system?
Each phone would hopefully have a key to press that will alert all other phones that the user needs help,

Thank you in advance of any help.
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Meet the Family that is Made for Collaboration
LVL 2
Meet the Family that is Made for Collaboration

The TeamConnect Family product group as part of the Sennheiser for Business Portfolio comprising high-quality, technically well-conceived meeting solutions for business communication – designed for any meeting room and any meeting situation.

I'm trying to add a single queue 'csqFabFursEscal' in the working UCCX script (FabFurs_Working.pdf).  I've made changes in an attempt to add the queue in the broken script (FabFurs_broke.pdf) and it doesn't work.  Everytime I load the broken script into my test application I get a system message saying their are problems and to call back.  

This isn't a complex script and I know this is an easy solution for someone better at scripting.  I would really appreciate a UCCX scripting guru take a look.
FabFurs_broke.pdf
FabFurs_working.pdf
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Hi,

Has anyone managed to get Cisco 7942G to work with RingCentral.  If so could you share the XML file, ours is just stuck on registering.

Thanks
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On cisco IP phones (model number 7911) it stores some useful information about placed/received calls in the directories application- is this data stored locally on some storage within the phone, or would this be stored in a central database in a managed voip environment, if so being a cisco device can you elaborate where that information may be stored.
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We have 3 Comcast business lines  and we have 5 phones working with the 3 lines perfect and I purchased 5 new phones but when I swap them the new phone are not working is not give me dial tone

any idea why?


this a regular lines not IP
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Hi all, I am in the process of diverting an incoming call to a particular number on Sonus gateway from a number (Inbound let's say 5000 number) to an external PSTN number 9999 and I would like to do this on the Gateway's end not on the Lync client.

I was able to do this on the client end by simply entering the number in the call forward on client side but if I do this through the gateway with Normalization rule (using Transformation table) with the appropriate Signaling group and Call routing table I get no error in the logs if I choose one ITSP or I get Proxy Authentication Required if I choose another ITSP as signaling group destination.

How do I solve this authentication issue when forwarding calls from a number to another? I have been reading articles but mostly it says it has to do with sip manipulation but I don't want to manipulate the sip, I just want to forward call from a number to another.

I would appreciate any suggestion.
Thanks
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Hello all, I am in the process of rolling out a VOIP solution for an office of 15 to 20 lines.  The customer is debating between hosted solution like Zultys Hosted with a 3 year contract for $305 recurring fee and zultys 36G phones for $194 per phone, or go with an appliances  and manage it myself a switchvox E510 for $695. and phones are D60 for $139 each and software registration code for $1000, and extension licence and subscription fee 0f $80 a line per year. I am really leaning towards the appliance but just not sure of the switchvox.  I really need something inexpensive but dependable any suggestions??
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Hello We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I am trying to set up my dialplan entry so we can call out from Canada to Irland and Trinidad, our current Dialplan allow us to call only witin north america but not outside,
Here I have the country and area codes that need to be called out from Canada
Irland 353-287-xxx-xxxx
Trinidad 868-788-xxxx

Here I have my Dailplan that i have made up to call out but no luck can someone please guide me correct dialplan please

exten => _9353NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91868NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9353NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _91868NXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9353NXXXXXXXXX,3,Hangup
exten => _91868NXXXXXX,3,Hangup


Here is the call_log but no luck (last 4 digits of phone number being modified to xxxx)

[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:1] AGI("SIP/298-00017de5", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:48:07] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:48:07] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101134807_298_1868788xxxx)
[Nov 1 13:48:07] -- <SIP/298-00017de5>AGI Script agi-NVA_recording.agi completed, returning 0
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:2]
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When I have someone on hold, and I'm engaging in a consultative transfer (speaking with the person I am about to transfer them to), the headset switches to speakerphone for 1 second every few minutes. It results in me not being able to hear them mid-speech and vice versa; its a real pain. It happens to all 7 of our phones, throughout the building. Have tried speaking with Mitel, system help companies, looking through the set-up manual, looking through the troubleshooting section, googling the exact issue.... etc etc. Nothing! Please can anyone help? My boss has given me the task of "sorting it out" and I just can't seem to :( Thanks, Sophie.
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I have a pair of C-level users who both experienced a problem at one of my sites. I'm trying to determine if it's a GoToMeeting problem (not my problem) or a phone problem (definitely my problem). You guys will give me a quick answer, I just know it. :-)

When the users join the audio portion of a GoToMeeting event from their Cisco desk phones (on my CUCM-powered phone system), right after the point where they enter their PIN number for the meeting, they are supposed to press the pound key "#" to join the audio part of the meeting. Every time each of them presses the "#" key on their desk phones, either the meeting or the phone hangs up the call. When they join using their cell phones it works fine.

This just started happening and no changes have recently been made to the phones or the CUCM settings. I'm told that it happened once before but was not reported, and that other occasion occurred several months ago but then the conditions returned to normal operation and they didn't see this again until yesterday.

Can I get some opinions on which party is responsible?

Thanks experts!
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Concerto's Cloud Advisory Services
LVL 5
Concerto's Cloud Advisory Services

Want to avoid the missteps to gaining all the benefits of the cloud? Learn more about the different assessment options from our Cloud Advisory team.

Hi,

We have VOIP service through intermedia, they do not offer call recording. Is there some third party service i can use to record calls?
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My Company just bought the Polycom 7000 and want it hooked up to the Samsung 7400. Documentation for just about any of the Samsung 7400 systems is rare and even harder to find how to set anything up. If anyone has any experience setting up Non-Samsung Phones to Samsung phone systems I would be grateful.
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Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
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There is a fellow it guy that keeps forwarding his line to my extension when he is "busy"coding or something else, he always forgets to disable it and I've had enough of it so I want to disable his ability to forward calls, or disable my number from accepting forwards from him only if possible or altogether from anyone, I don't want do disable the function globally just in his ext number forwarding or mine receiving from him/any. ( I have console admin access)
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I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
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For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.
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The network provider in my area recently shut down their 2G service and now my GSM modem doesn't work. Is there another kind of modem that would be compatible with a 3G network? I've looked a bunch of other places and it doesn't really seem to be terribly clear about it.
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Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
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Is there anyone out there IT Wise who has gone thru an NBN migration for multiple clients?

Our area is in the midst of deployment, and a number of my clients are reporting getting calls that sound suspect.  
  • One client, a law firm, was called by someone saying they were their PABX provider, and that the firm needs to replace their PABX because it isn't NBN compatible.  The trouble is its only a few months old and I would have thought any PABX sold would be NBN ready
  • Another was called by someone from their telco saying the same "you need a new unit" however the customer luckily has notes from the negotiation of contracts showing the PABX has NBN components already.
  • Another called by their telco, and has 6 month old system, was told by the telco they have no records of how the gear was configured or what's fitted in the PABX box, so the customer has to foot the bill for a new PABX

Questions
  • How are you helping clients assess what gear they already possess is NBN capable
  • Are there any resources you can recommend that can be used.
  • What has been your experience with the NBN rollout in your area
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Eye-catchers on the conference table
LVL 2
Eye-catchers on the conference table

Challenge: The i-unit group was not satisfied with the audio quality during remote meetings. They were looking for a portable solution with excellent audio quality for use in their conference room but also at their client’s offices.

the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
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I want to know about Best call center campaigns providing company of America with lots of free campaigns for off shore call centers.
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Hello all,

we are currently using IVR designer with admin rights. Our corp is removing admin rights on the machines. Is there any way to user Avaya IVR designer without admin rights on machines?

Regards
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This has just recently started happening.
On making or receiving a call my phone is switched to loudspeaker. The loudspeaker icon Is greyed out. Clicking the icon turns it green and the phone is still on loudspeaker. A second click greys the icon again and the phone reverts to normal operation. It's becoming annoying having to switch it every time.
I have moved a whole load of apps to the sd card, but I haven't loaded anything new recently.
Can anyone suggest how I can stop it being on loudspeaker for every call please?
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How do you get a Cisco 3CX SPA514G phone out of Do Not Disturb mode?
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Any idea how to fix this error?
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Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.