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Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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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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NEW Veeam Backup for Microsoft Office 365 1.5
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NEW Veeam Backup for Microsoft Office 365 1.5

With Office 365, it’s your data and your responsibility to protect it. NEW Veeam Backup for Microsoft Office 365 eliminates the risk of losing access to your Office 365 data.

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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I am the Office 365 administrator.

What steps need to be followed to record all Office 365 Skype for Business instant message chat conversations?
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Hello - im looking for hold music that is free to use. Anyone have a link to download one?
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Greetings,

I have a site where I have several Yealink T26P phones connected to multiple Aruba (HP) 2530 POE switches.  There is a stack of 4 switches, with the primary switch connected to a SonicWALL TZ300 router.  All the phones on the primary switch work normally.  The phones do not work on the secondary switches.  I have two VLANS (data and voice).  Basically, I have every port tagged with the voice VLAN and untagged with the data VLAN.  The only difference is that I have two ports on the primary switch that are untagged for the voice VLAN only (voice uplink from SonicWALL and phone controller).  In testing with other phone types, I've been able to plug into the POE port on any switch and have it work.  This is my first interaction with the Yealink phones, and it is remote.  As the only difference between the switches is the uplink port config, I'm guessing that I need to have a port untagged for the voice VLAN only and a different port untagged for the data VLAN only?  This would seem like a very poor design, as it would effectively require the use of up to 4 ports per switch.  So, what am I missing?  My SonicWALL is serving as the DHCP server and seems to be working fine and it actually shows the non-working phones actually get an IP.  They just don't connect.  Move them to the first switch and they immediately connect.

Thank you for any assistance that you can provide.

Jeremy
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I have multiple Cisco 504g phones, that don't seem to take the attendant console changes. I have set up the first 3 extensions as a direct speed dial to other extensions on the network (the extension is set as disabled), and all 3 speed dials with flash orange, even though they are subscribed and working. I know to add c=g in the serv subscribing section of attendant console, but these phones seem to not take any changes I make. Hopefully somebody has an answer for this, as these are for a high end hotel, and I cannot have these lights constantly flashing.
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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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Hi Everyone, I have a question regarding call forwarding from a Cisco 6000 UC to a typical analogue fax line on a MFP. I understand that typical call forwarding can be done from a callmanager to any external telephone number. Is the same possible for fax lines as well?  I have fax line numbers that now ring at a Cisco VOIP phone which I need forwarded to a MFP, so people can receive faxes on it, this would need to work both ways (incoming and outgoing)

Thank you for your help in advance.

Mayson
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I have one user that is not able to search the GAL for contacts.

He is searching within the “My Contacts” tab but his searches show no results.

He is using SFB on Windows 10. The rest of our users have no issues doing the same.

What am I missing?
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When I connect an Avaya 9600 phone to a H3C 802.1x port, the phone just works, it doesn't need or attempt any sort of authentication. I do not have to have a pc connected in line with the phone.  I like this behavior, but I don't understand it.  I have the voice vlan oui's set statically, so it's operating in secure mode.  Could someone explain why 802.1x on the port bypasses for the phone?
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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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the numbers that need transferring are:

44113246xxxx
44121236xxxx

And they both need transferring to:

0113 337 xxxx

How would I port these #'s in cube router  

example:

voice translation-rule 1
  rule 1 /4175209020/ /1000/

!        
!              
voice translation-profile INCOMING
  translate called 1
!        
!

dial-peer voice 200 voip
  description *** Incoming Dial-Peer ***
  translation-profile incoming INCOMING
  session protocol sipv2
  session target sip-server
  incoming called-number 4175209020
  dtmf-relay rtp-nte
  codec g711ulaw
  no vad





Thank you,
Riz
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hi

i have microsot access 2016 and skype 7.40, and i use this code to dial with skype:
Dim skpSkype As SKYPE4COMLib.Skype
Dim calCall As SKYPE4COMLib.Call

Set skpSkype = New SKYPE4COMLib.Skype
'Set calCall = skpSkype.PlaceCall("+17181234567")
Set calCall = skpSkype.PlaceCall(Me.Phone_number)

Open in new window


is there some way that i can with code program that after a few seconds it's will dial an extention number for example 1#.

thanks
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I have a client who recently purchased VoIP service from ShoreTel and also installed Jabra 9450 wireless headsets.  They use ShoreTel Flex software on their computers instead of hardware handsets.

The issue is that they are unable to answer incoming calls with the button on the headset;  they have to click on a button in the software, which is not very convenient.

ShoreTel hasn't been much help on this as they say they don't support this feature.

Considering how basic and useful this feature is, I'd be very surprised if there's no way around it.  I'm hoping that someone on EE is familiar with this and can suggest a workaround.  Using different software (such as ShoreTel Communicator) is certainly possible as well as a different brand or model of headset.

Any useful input would be greatly appreciated!
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We have call manager 10.5. Caller ID is set to show individual numbers were we dont put anything in the number mask, where nothing is set in the Route patterns for this office. The Caller ID DN on the Gatway, E1 port is set to 225660XXX, which means that everybody is sending out his individual number. How do i block Caller id on an Individual basis?
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In MS Skype document "Media Quality and Network Connectivity Performance in Skype for Business Online" they say to that in order to estimate the RTT to Skype network edge you should ping Anycast VIP 13.107.8.2. When I do that it's very low at 30ms or so. But when I make actual calls in Skype for Business Online and sniff the wire - I see that real time communications happens with a distant server in the range of 52.112.0.0/24. This is close to 100ms away. What I'm trying to find out is exactly what is the relationship between this Anycast VIP and the actual server you end up peering with for your Skype calls. Any experts on here familiar enough with networking and Skype able to address this? Thank you.

Media Quality and Network Connectivity Performance in Skype for Business Online
https://support.office.com/en-us/article/Media-Quality-and-Network-Connectivity-Performance-in-Skype-for-Business-Online-5fe3e01b-34cf-44e0-b897-b0b2a83f0917
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I've been struggling with VOIP call quality. One thing I notice is where calls to PSTN or Conference are made from work on the west coast or from home on the west coast - the traffic takes a 100ms trip to the east coast to get to the Skype voice control and RTP gateways. Is there any way this can be altered so you can use gateways on the west coast to reduce hops and delay to the Skype voice gateways?
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What does it mean to be "Always On"?
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What does it mean to be "Always On"?

Is your cloud always on? With an Always On cloud you won't have to worry about downtime for maintenance or software application code updates, ensuring that your bottom line isn't affected.

Hello

We are using AVAYA IP Phones.

When I open AVAYA IP Office Manager, I get the following msg:

"The security certificate will be expired in xxx days."

I have found the following solution on internet:

Security settings > system > identity certificate > delete the certificate and save security configuration IPO will generate a new self signed certificate.

Actually, I am not an expert in managing IP phones and have limited knowledge regarding this, so I am a bit confused and afraid of applying the above mentioned solution, because after applying if any other problem appears then it might be a problem for me to solve, then I have to again go through internet finding the solution of that newly created problem.

Any advice and assistance shall highly be appreciated.

Thanks
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Have been on a Cisco/Linksys E3000 router for a few years.  My Mitel 5360 IP phone has worked flawlessly connecting to my office the entire time.

Replaced my router with a Netgear Nighthawk X4 R7500V2.  The IP phone gets an IP address (DHCP) but hangs at "Contacting Server."  Cannot connect to my office.

Reconnected the old router and the IP phone works fine so it's not the phone.  

Suspect I have to open up a port on the new router but have no idea where to start.

Any suggestions appreciated.
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Normally I've seen MOS scores on a 1 to 5 point scale. But in Skype for Business Call Analytics they're expressed a decimal. e.g.

Average network degradation 0.009717 MOS

What are they saying? Is that a 4.93 MOS score?
ScrnGrab2671-171018-17.21.jpg
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I am curious if there may be a better phone system backup strategy other than what I am currently doing. My current backup involves  rerouting the main number to a series of cell phones that are configured in a chain (no answer/busy forwarding enabled on each)

I have noticed there are some online phone system backup solutions. In the event that our onsite phone system goes offline for whatever reason, I need a solution that is cloud based, and does not require any local hardware other than a computer with internet access. It would need to be able to accept at a minimum two telephone numbers that would be forwarded. Does anyone know of any service that could handle this?

Thanks in advance!
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Hi ALL,

We're trying to plug in a phone into a switch that is primarily used for data but was configured for data and phone, however, when the phone is plugged in it does connect to the network. Below is config from the data switch configured for data and phone and a switch configured for only phones ( or at least that what we use it for).  Please let me know. Thank You.

Switch configured for use with phones - both data and phone work

interface FastEthernet2/0/39
 switchport access vlan 10
 switchport voice vlan 20
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 spanning-tree portfast

Switch configured for DATA - Only Data works (what do we have to configure for phones to work?)


interface GigabitEthernet3/0/46
 switchport access vlan 10
 switchport voice vlan 20
 spanning-tree portfast
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For Skype for Business O365 MS offers a Call Quality Dashboard that shows quality trends. e.g. 1000 good calls, 20 unclassified, five poor calls. But I'm not seeing a way to search what were those five poor calls and when did they happen, what was the latency or jitter, etc etc. Am I missing something? How do I drill into call quality with this tool? Or are there other tools that would get this done?
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Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).