Voice Over IPSponsored by Jamf Now

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).

Share tech news, updates, or what's on your mind.

Sign up to Post

What is the best practice of backing up Cisco Call Manager 11.5 ? (VM)
0
Digging into Call Manager 11.5

What is cube ??
0
We just installed ShoreTel phone system connected to a PRI with 20 DID numbers .
We also ported our main number (say 453 270 9999) from another provider to the same PRI line.
how do I route my older number which was ported to an hunt group or a single extension on a PRI?
when I dial  453 270 9999 from outside I am getting a prompt from Shoretel to dial an extension number.
I have to conigure it to ring to a hunt group or a single extension.
0
I installed WhatsApp and need to help someone connect to me. What do I provide?
0
Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?
0
Hi,

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
0
Hi all

I monitor my customers internet connections with PRTG and I am seeing spikes in outbound data.  It seems to be affecting about half of the connections I am monitoring.  It has been noticed because it saturates the upstream on some ADSL circuits providing VOIP, dropping calls.  

I have a customer with two networks, 1 voip and 1 data.  strangely, the bandwidth peaks affect both at the same time.  Then I noticed that it affects other customers at the same time!  I considered whether it is my monitoring platform, but some customers don't have them.    Attached are some graphs, with pink being the outbound data usage.  

The graphs show outages as a series of red dots, but I don't know if the line is down or whether it is simply that the upstream data is saturated so the usage data gets lost or the monitoring software times out before the SNMP response gets there.  

So for the purposes of this, please view the high upstream data and the outages as the same thing.  You will see that the times correlate across all the graphs, approx 9:05, 9~:30-9:45

Any suggestions as to what it is or even better - any ideas to prevent it would be greatly appreciated.  

The routers are all Cisco 867VAE, 887VA or 2821 and I use SNMP to monitor them.
Experts-Exchange-Graphs.pdf
0
I do not have an account yet. Where do I sign up?

I also have an iPhone 7. Once I register, do I download the app and just log in?

I need to share my desktop for a presentation, so hopefully, WhatsApp supports screen sharing, like Skype. I have a Mac and want to present a Powerpoint Presentation running in Windows 7 on Parallels.

For this to work, can I install WhatsApp on my Mac? Then, load Parallels, Windows and Powerpoint?

Or, do I need to install WhatsApp on Windows?

Thsnks
0
So we have a client who would like to have different ringtones for internal and external calls, because when a call comes in, all phones ring, and they generally ignore it unless they recognise the number and then someone at reception picks it up and transfers the call to the right person, and they just think its another external call and ignore it, they cant tell the call has been specifically transfered to them.

The system is Elastix 2.3 and the phones are Grandstream GXP2130.

I found in the GXP2130 settings I can set a different ringtone for a certain number, so I can type "XXX" as the matching rule and all 3 digit numbers (All extensions are 3 digits) will then ring with a different ringtone. That is working fine.

But the problem is when reception picks up an external call and then transfers it, the caller ID is forwarded when the call is transfered and the phone then see's the full 10 digit number as the ID and rings with the normal tone.

How can I stop caller ID from being passed on when a call is transfered?
0
We are moving to a cloud hosted VOIP system thru our phone carrier.

Each desk has a single network drop with the PC connected to the phone. The phones are configured for DHCP. They are currently getting IP's and gateway from my Server 2008 R2 domain controller.

Our network has one TP-Link SG-3216 L2 managed switch as the network backbone with a single IP space.

The phone provider has installed an Adtran with a 3mb link for the phones. I have a Cox 50mb connection for internet going thru a Firebox firewall. The Firebox has unused interfaces.

What's the best scenario for connecting the Adtran to my network? The provider recommends a VLAN with the Adtran connected directly to the managed switch. I'm not really sure how to configure the switch to accomplish this.
0
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.
0
Hi guys,

i am new to networking, we have a requirement where we need to design a network for a small office,
there are 15 users with ip(cisco)  telephony  headsets.

Please suggest me which Cisco router(small bussiness) with part number is suitable for the requirement.
0
Hello,
Need a question answered being asked of me and our Cisco CUCM 11.1 phone system. I need to know if Cisco CUCM 11.1 will integrate with NetSuite CRM via TAPI and/or CTI. If so, any third party apps needed or can it be done within CUCM?
Thanks in advance!
Steve
0
Hi All

I'm in the process of planning out implementing site codes dial codes within my company's CUCM environment. We currently use 4 digit dialing across all sites, but this is no longer scalable with the amount of expansion we're experiencing.

I've set this up in a lab environment, and it works as expected. However, I'm stumbling with the modifying the Calling Party ID when dialing between sites. So when someone in Site A dials an extension in Site B, the Site A caller ID is prepended to include the Site A dial code.

I'm think I have to create multiple translation patterns, intersite partitions, and CSS's to do this; but this also seems messy, so I'm wondering if there's a better way to accomplish this?
0
I'm running CUCM 9 and Unity connection 9.  All screen's on my Cisco IP phones go dim (black) at 5pm.  I know this has to be a global setting in CUCM, as all phones do this, but I can't figure out where to go to change this.  We have recently extended the hours the office is open, so I need to change this.  Does anyone know where this setting is?  Thanks!
0
Is there a way to use google voice with a VoIP phone that has only Cat 5/6 port ?
0
I am brand new to VLAN's so please excuse my ignorance. We have an AllWorx VoIP phone system that has been on our regular network for a few years. Our employees plug their phones into the network port at their desk then plug their laptops into their phone's (yes we have WiFi but wired is much more reliable). So each phone gets an IP from the DHCP Server then acts as a switch and each laptop gets an IP from the same DHCP server.  We are growing and starting to run out of DHCP addresses on our LAN. If I set up a VLAN for the phone system would employees still be able to plug into their phones and get an IP Address on our network?
0
what has happened is a little weird.
we configured these two cisco switches and they have been working fine with the phones all this time
then mid last week we found that several of the  phones stopped working!
I have checked the configuration and cannot find the problem and was hoping having more eyes look into it will help
I have attached both configurations to this ticket
all help is GREATLY appreciated

FYI due to restrictions in types of files we can upload, I renamed the files with a .txt extension; please rename back .cfg and this will enable you to see the complete configuration
propmatt-1.txt
propmatt-2.txt
0
Got a small business needing a new switch, dusty warehouse environment, 16-24 port 10/100/1000, with POE for VOIP phones, doesn't need routing so it can be unmanaged layer 2 ,  Last but not least small business budget.  I would like some opinions (including why) on fairly durable switches that won't break the bank.
0
When you use Microsoft PSTN calling - can you use your old Cisco 7945 phones provided you load them with the SIP firmware instead of the SCCP?
0
Hello Friends.

for a VoIP project I have to install and deploy Skype for business for 50 users. they will going to use most of the skype for business features like:
-IM
-voice call
-file sharing

but video calls and conferences are NOT important and required very often.

my question is what hardware configuration should I use for my server (CPU.RAM.HDD.NIC). as they are on a low budget they ask me to run it on a PC.

tnx in advance for your opinions
0
Recently there was a change on how we dial and they implementing having to dial the are code 360 before the number. Many of our employees are used to dial the number only for local calls. We have a Panasonic TDA50, is there a way we can program the area code 360 and employees can continue to dial like before?

Thanks,
0
Hey all, i have an issue that external calls has noise, delayed audio from the external side. Internal calls are fine.
We recently changed from the Telstra supplied modem to a Draytek Modem. All ports have been opened up the same as they were on the Telstra one, all lines and SIP registered straight away, however i have not been able to resolve the noise.
As Draytek have alot more advanced firewall settings, QOS, I'm not sure what feature / settings i need to change to test.
We are running freepbx 2.11.

Thanks
Matt
0
I have multiple cisco switches, from 2960 to 3750, and we are using voip phones that use the same ports as the computers.
So I'm thinking to to leave the computers on the default vlan, which is vlan 1, and have the voip phones on vlan 200 or some other vlan.  As far as I know, to have each port in two separate vlans, I would have to make all ports trunk ports, is there a better or another way than doing that?
0
For some project, we are exploring Twilio product "Programmable voice" . The pricing for per minute calling on normal voip is 1.5 cents whereas for SIP it is 0.4 Cents.

Thats makes me wonder how SIP differs from normal voip? Can I practically implement SIP in a small office of 4/5 people ? Can some expert exaplain me in simple English (Without using technical terms).

Twilio pricing I referred is here https://www.twilio.com/voice/pricing
0

Voice Over IPSponsored by Jamf Now

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).