IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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For years I have used Plantronics Supra binaural headsets along with the matching Plantronics headset amplifier/interface M10 or MX10. ( I am not at that location now.) I have the requirement that my phone audio be crystal clear at all times. I have never had any problem with clarity until yesterday. I never had considered VOIP because my internet speed was not ideal. Several months ago I got fiber and my up and down speed is 1 GIG with pings at 2ms. With this super speed I thought that VOIP would be an acceptable choice since it would save me more than 50% of my phone bill. Now that it is installed I am told that my transmitted voice is somewhat distorted and there is some sort of slight crackling in the background. I cannot live with this problem. I spoke to level one of tech support last night and he confirmed that I was indeed distorted. I have another line that utilizes the MagicJack. I phoned the tech on that line and the distortion was still present. I also switched from my headset and amp combo to a regular phone on the business phone system (Avaya Partner) and the distortion is still there. Level two is supposed to get back to me today and begin troubleshooting the problem. I was just wondering if any Expert has encountered this difficulty before? With such incredibly high isp speed the is the last thing that I had expected. If an Expert has any ideas please let me know.  

Configuration wise: On the isp's router there are two phone jacks. I go from jack One …
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I am using Freepbx 14 and working fine but I got thousands of attacks and in Intrusion Detection, my public ip  has been blocked sometimes and because of this calls are not working. I am using fortigate firewall and opened the 5060 to 20000 ports for the FreePBX so My question is 1. are ports forward mandatory for inbound route ( if I change the sip registration port from 5060 to other and do same with the trunk provider ) . Please let me know how I can make this FreePBX more secure so call disturbance would not occurred in future.
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The user has moved into a new apartment.  They brought their Panasonic wireless home phone system, with a base station and two satellite phones with them.  

They would like to just give everybody they know only their cell phone number.  They need an adapter that would transfer all the cell phone calls to their Panasonic system.  This way they have the convenience of the Panasonic stations, and don't have to carry around a cell phone all day.

What is a good unit that will do this?

Thanks
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The user lives in an apartment complex that provides phone service as part of the rent. They were able to connect their Panasonic wireless home phone with 3 stations to this system.

However, even though the wireless phone receives calls, the caller ID isn't working.  Most inbound calls show the ID as DID/DOD.  Outbound calls show the town or just the phone number.  I called my iPhone, and my iPhone recognized the number as being on my contact list, and thus showed the name.

In addition, the answering machine built into the main station usually doesn't pick up.  Management said that an answering machine should work.  The Panasonic HAS an answering machine.

Is there anything we can do to get the caller ID and answering machine portion of the Panasonic phone working?

Thanks.
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I am using PFsense2.4.4 with 3CX 16 and Everything (inbound and outbound calls) are  working fine but I am not able to register the phones over the VPN ( other end firewall is fortigate) I have done everything as https://www.3cx.com/docs/fortigate-firewall-configuration/  . The interesting part is I am able to work with softphone but not with IP phones( tested with yealink,polycom).
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We have a yealink IP SIP phone that needs to connect from the outside.  We have set up the phone and tested it internally and it's good so we moved it externally.  

In the phone, we edited the account and put in the public IP of our router (a Sonicwall NSA4600) and waited for the phone to register.  It fails to register.

I've checked the ports and I DO have the right ports open on the Sonicwall but I've also read a lot of posts about people having trouble with SW and SIP phones.  So, after reading, I have made the changes suggested on the posts I've read and still no joy.  

I'm using https://www.yougetsignal.com/ to test 5060 and it reports that it's closed.

Does anyone have any insight into how to make the SW work well with the SIP phone?

PS:  An alternate port scanning tool tells me the port is filtered.  So, I'm looking up how to turn filtering off in the sonicwall for this service.

Thanks
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Hello,

I am a beginner with 3CX, and unfortunately I'm having issues configuring my new Grandstream GXW4104 gateway. I can get outbound calls to work, but for the life of me I can't figure out how to get the inbound calls to work properly. I've spent 2 days trying different things, tried several configuration guides, read the forums, been all over the internet, but nothing seems to work. I did have 1 successful inbound call routed to an extension (immediately after completing a re-configuration), but was unable to make a 2nd inbound call - it just rings out.

Strangely, after a standard configuration (as per the 3CX guide), with no inbound rules, inbound calls will consistently go to the operator, which is by default sent to voicemail, as I have no phone set up for that extension. If I then create an inbound rule or change any settings, no inbound calls come through. Changing the settings back to how they were also results in no inbound calls! I can only get it to work again by doing the gateway configuration from scratch!

Apart from the situation above, looking at the 3CX activity logs shows no activity when an inbound call is placed (log set to Verbose). Within the GXW4104 web interface, it shows that the line has a call coming through, yet 3CX reports nothing. I have set up the syslog server in the GXW4104, and from my untrained eye, when a call comes in, it is spitting out heaps of data, yet 3CX activity shows nothing. The gateway status always has a green light …
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I'd like to know how to be able to subnet a PepLink Balance one router for less than 50 users. More concretely I'd like to have the IP phones on a  separate subnet. How can I go about that?
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Please see the attached Fixed Line Operator (Telkom) Voice Mail answering Service. There is a distinctive tone indicated on the telephone handset when picking it up to indicate a message is waiting. Is there a device available that can indicate that a message is waiting and automatically retrieve the message and play it for the subscriber?
 This would save the trouble of each time dialling the prescribed phone number to retrieve and messages. It would be useful if the Message Waiting Tone could be detected and displayed fro the user. It would be wonderful if the device could play back the messages through a loudspeaker. I would imagine this would be a programmable dialler of sorts. The device should be senior Citizen friendly.
189_CallAnswer.pdf
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Hi Experts,

I am having an issue with my freePBX and the registration with my Net2Phone provider.
I have a SIP trunk setup for our phone lines

Apparently the connection keeps dropping because the registration refresh is not matching the provider's one. My provider ask me to change the refresh from 45 to 60 but whatever settings I put in the Registration Settings does not change the setting.
Can someone help me fixing the registration so are calls never drops?
Apparently I need to change the refresh from 45sec to 60 and the expiration to be 120sec.

Thanks


This is what shows under the Chan_Sip Registry:
                  
Host                                    dnsmgr Username       Refresh State                Reg.Time                
siptrunk.net2phone.com:5060             Y      8764774091          45 Registered           Fri, 14 Jun 2019 10:06:52
1 SIP registrations.


and the attached file shows the registration settings



Registration setting
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Hi We used to use term Call manager. Now we always say CUCM, whose long term is Cisco Unified Communication Manager.  My question is why we say Unified?
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What is the best Opensource chat or team servers that you can install on your premises ? Something like Matrix (Synapse) , RocketChat or Zulip?

Something that supports Chat and PBX integration.
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I have a Cisco 2900 Series Router which has a EHWIC 3 card installed for my phone system.  I would like to replace this with a Layer 3 Switch or a firewall that can sit on the edge for "firewall" stuff (Intrusion protection, RTDPI, GeoIP, Anti-malware, etc.) which will allow me to connect my Cisco 7942 phones and allow remote connection to our network.  I will then connect to our switch stack on the inside.  Any Suggestions?  If you have other network suggestion on how to do this that would be great too.
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Hi, we have just had Avaya IP Office and Voicemail Pro installed on a Hyper-V Virtual Server (A Dell Poweredge R7040 with 32GB RAM Intel Xeon Silver 4109T 2Ghz (2 Processors). The Avaya virtual server has been running fine since Christmas but over Easter it crashed. The telecoms guys that support it say that for 1-100 users it should be at least a 4 Ghz Processor (we have 75 users) and 2-3 cores.

Now, I can shut it down tonight and give it an extra processor, but would we really need to upgrade the processors to faster ones? It's a fairly new server and I thought that the Xeon Silver 4109T although low Ghz are pretty good? Thanks for the help.
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This is (hopefully) a dirt-simple question for the Avaya IP Office gurus, but over 30 minutes of Google searching has yielded no results for me.

I have an existing Voicemail Pro module ("VM:Support") that works perfectly, and I know how to redirect external incoming calls to the module, but I need to create a single extension (x5555) that points directly to the module.  How is this done on an IP Office 500 / Voicemail Pro system?
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Does anyone know if there is a good soft phone or virtual phones for users to use in place of handsets?

Just wondering what people are using instead of handsets as alternatives.
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Hi there,

I'm looking for a solution to make a 3 way voice call between 3 sites and hang 5 lines off each of those sites.

Two of the sites have 1 PSTN line only and one site has two PSTN lines, and each location would like 5 POTS/physical handsets to be able to use at any given time (not cell phones).

Site A - 1 PSTN
Site B - 1 PSTN
Site C - 2 PSTN

Each location also has IP connectivity between the 3 sites but it is setup like a WAN so is a closed network and there is no internet breakout. Therefore no hosted solutions can be used.

So far this says simple PaBX either traditional or IP based.

Each site can call out which is fine but there maybe times that each of the sites need to be on the same call together and have more than one handset from each site join a call.

The bit that I'm not clear on is if each of the sites want to have a 3 way call between them and have each of the 5 local lines connect into the same call. Would the easiest way to do this be simply have a PaBX system on each of the 3 sites and have two off the sites (A and B) with 1 PaBx dial in to the site with the 2 PSTN sites (Site C). Then have the PaBX on site C merge the calls?

This seems a little clunky, is there a better way to do this? Is there an VOIP/SIP solution that would work better?

Trying to keep this uncomplicated and keep margin for error to a minimum so simple PaBX solution was my initial thought. If this is the best solution, does any one have any …
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What is the command to trace a sip call inbound or outbound on a Cisco 4431 running CUBE if I want to check that the carrier is sending me numbers correctly?
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VoIP - My customer reports that their phone lose connection to the hosted service every day.  They have to reset multiple times per day.

I will be onsite sometime tomorrow (April 11) - hoping to be able to access some expert assistance.  Meanwhile, if somebody could point me to a link that I'm sure exists, to help me in troubleshooting VoIP.  I'm good at networking, but have minimal experience in troubleshooting VoIP.  Thanks in advance.
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I'm trying to add a VM to a user in Exchange and I keep on getting this error.  We use 3CX as our VoIP system, in a VM, and Exchange is where the VM is stored.

Starting yesterday, when I try to add a VM to a user, I keep on getting this error. NO changes to my 3cx server or Exchange, so this is odd.

I checked my event logs app/system, and didn't find anything that would point to an issue.  
Also did a wireshark capture for 10 seconds on my exchange during the time I received the error, and still, I didn't see any issues in the wireshark that could be a problem.  
I looked on my 3cx system, in the logs, and nothing shows up there as an issue.

I don't think this would have anything to do with it, but my manager disabled TLS 1.0 on our exchange server to make it more secure, so I doubt that would have anything to do with it?

Not sure what else to do, any thoughts?
exchange VM
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Does anyone know how to unlock a Shortel desktop phones (model 210 for example) to work with Ring Central?
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Dear experts,

I have Cisco voice gateway routers and I need to know if there are any fax lines active in this organization. Is there a way to figure it out and what is the process?
unfortunately, the company does not have enough info but they have many MFP printers with fax lines and are working.

Thank you
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I am looking for a way to check the trunk and the DID charged  for the 3cx which is currently setup and working properly.
How can check these information?
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Hey all, one of our users (at least we think it is only one) is having issues with caller ID, note that we use skype for business/O365 online.


For example: When he calls me I see his proper skype/office # on  the caller ID because he is calling me from his skype for business account....but if I miss the call, I do get an email notification....but in the details it shows as if the call came from his mobile #.


Any idea where the issue could be?


I did notice something in the desktop client settings...under Options - Phones, I do see his office and mobile #s listed, but am unable to change the mobile# because the "Mobile Phone" button is greyed out (see attached image)

I found this article that said its an admin setting, but I can't find it anywhere...again, not sure if this is the cause but figured I would mention it.

https://support.office.com/en-us/article/change-my-phone-number-for-skype-for-business-20e03cc1-c023-4e5d-bafd-064ddb59ed5e?ui=en-US&rs=en-US&ad=US


Thoughts??
skb.PNG
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Hello Experts,

I at a customer site and they a VALCOM V-2006A Amplifier , I configured the SIP paging adapter and connected it to the Valcom V-2006 Amplifier, the SIP adapter has paging extension. Now the question is how the speaker should connect to this amplifier? What do I do to make this work?

I don't know much about the VALCOM V-2006A amplifier and I need this to work.

Thank you,
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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