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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Hi,

We have a switch stack of 7 3750 switches. One switch just seemed to stop working, still has power. After restart, using the sh switch command, the switch seems to be stuck at initializing, after restart the of the stack, the switch shows ready. Its a POE switch and plugging a phone directly into the switch, no power. However plugging in a laptop works, data is working just not power no data. I used some basic commands, show config, ver, vlan, int and compared the configs to the other switches and everything looks good.  The switch in question has no error using sh int. Any suggestions greatly appreciated. Below is a output from sh int, for the switch in question,  all ports are shown the same.

FastEthernet5/0/20 is down, line protocol is down (notconnect)
  Hardware is Fast Ethernet, address is 0017.94b5.d016 (bia 0017.94b5.d016)
  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec,
     reliability 255/255, txload 1/255, rxload 1/255
  Encapsulation ARPA, loopback not set
  Keepalive set (10 sec)
  Auto-duplex, Auto-speed, media type is 10/100BaseTX
  input flow-control is off, output flow-control is unsupported
  ARP type: ARPA, ARP Timeout 04:00:00
  Last input never, output never, output hang never
  Last clearing of "show interface" counters never
  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
  Queueing strategy: fifo
  Output queue: 0/40 (size/max)
  5 minute input rate 0 bits/sec, 0 packets/sec
  5 minute output rate 0 …
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If I have a Skype for Business Online account  most folks reach me by a regular 10 digit telephone number.
But should people be able to reach me also at a sip address like sip://joebob@acme.com?
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I'm working at integrating Mitel 6867i phones into our network.  
At first, the provider said that there had to be firewall ports open (and, obviously NOT forwarded to any particular IP).  That is, the ports had to be opened incoming and outgoing.
Well, usually outgoing is open by default.  So that would only be mentioned to CYA.
And, if outgoing is all that's needed then saying incoming is needed seems double CYA.
Right?
My hope is that no firewall ports need be specifically opened at all.  

But, I started on this path by accepting what they specified and decided I would NOT run these things through our main firewall at all.
So, there will be a separate firewall as we have enough public IP addresses to accomodate.
And so, there has been a VLAN set up just to service the VOIP phones and to keep the firewall LANs separated.

I hope this much sounds like a reasonable approach.

If you're familiar with these phones (and I hope you are) then you'll know that the phones will accept a trunk line connection and "pass through" the main LAN / e.g. VLAN1 untagged to the computer on the desk.  (This saves running a new cable to the desk assuming the computer is already cabled).  And, the phone functions are supposed to run off a tagged VLAN / e.g. VLAN100 that's combined on the trunk.

All this seems fine so far.  I'm a little uncomfortable with the phones being so integrally connected inside the network but it seems like accepted practice.

As they say, "the devil is …
0
Skype (Windows 10). The Video Button has disappeared from its position on right side top - alongside the telephone icon. Have tried for answers without success. What does this signify?
Has it anything to do with paying money.  It is not a problem with my computer, as I have opened the account on a second computer with same result.
0
I need a small dual-WAN router to interface with an ISP for VOIP.
It needs to provide VLAN100 tagged on the LAN side (some older routers like RV042 won't do this).
It needs to have some ports opened - not forwarded, opened. (some newer routers won't do this).
I'd rather have a 4-5 port router as ports are unimportant here.
I'd like to add a management port/VLAN and block internet access from that port's connection.

What might you recommend?
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We are having a POE issue on some ports on a Cisco SG500-52MP.  About 5 users phones are not powering up.  We eliminated a cabling issue and a phone issue by plugging in the phones directly to the Cisco switch, however some ports are not powering up the phones.  For instance.

Joe's phone is not working and was plugged into port 25
Mary's phone is working and is plugged into port 41

If we swap phones, Mary's phone does not power up on port 25 and Joe's phone powers up on port 41.  This tells us something is going on with the switch.  We don't want to power cycle the switch just yet as other workstations are working.  The system has been up and running for about a year and no changes were made.

Is there something that we can look for on the switch itself to see why those 5 particular ports are not powering up the phones?
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I always get confused / use firewall / router rules so infrequently that I don't know  what the right way to set these things up.

Can you help?

I have a VOIP service using a Grandstream HT-502 V1.2A gadget.  The call quality isn't always good.  That device is plugged directly into one of the ports on an actiontec router (left over from when we had verizon fios, but now we have cable internet 15down / 5 up speed.

I have lots of other devices plugged into a gig network switch in the basement that might be using the internet at the same time as the calls?  That gig network switch has 1 cable going over to the actiontec also (so there's only 2 cables on the lan side of the router).

To improve call quality, that's a job for QoS, right?

The attached picture is what I did in the actiontec router.  The grandstream has the ip of 192.168.1.52

But then i thought, should this be on the Ethernet/Coax or  Broadband Connection (Ethernet/Coax) sections? Did I at least get outbound rather than inbound correct?

But that just gets the call out of the house with highest priority.  Once it's on the web, it's fighting with all kinds of data / can't prioritize it, right?

Does the VOIP provider have any bearing on the quality of the calls? Iwe are using VOIPO.com).  is there  a way to substantiate / test where the poor qiuality - dropped fractions of a second in the conversation, etc.  NO stuttering, max headroom type things.  Just a m ssing sound here and there.

And most …
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related question/answer
https://www.experts-exchange.com/questions/29095322/2-phones-ring.html?anchor=a42537973¬ificationFollowed=206569166#a42538269

Using verizon wireless for residential (not business)
is there a way to have both phones ring at the same time

*71 is one phone and then the other
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related question:
https://www.experts-exchange.com/questions/29095078/forward-phone-numbers-with-iphone.html#a42537163

Since you have Verizon, you can do this:

Calli *72 plus the forwarding phone number, including the area code (e.g., *72-555-555-5555). You will hear a confirmation message.

This works with all phones with Verizon service, not just iPhones.

Dial *73 to stop forwarding. You will get a tone or message.

Can I have both phones ring

I this answer only gives me second phone ringing.
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Need a voice support that is not TOO loud

I need to give a tour to about 10-20 people in a public area, but my voice does not carry very far. And if the wind is up, it's even harder.

But I do not want to use one of those megaphones. They are loud and use up one hand. I would prefer some sort of clip-on microphone, that uses BlueTooth to connect to portable speakers.

Can I WEAR a small speaker? Is that possible?

Or I guess I could pull a small speaker behind me, but that would also need a battery.
 
Suggestions?
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can I receive (not send) sms to skype united states phone number


not skype to skype message

receive actual text message from a cell phone
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I have an implementation of 3CX with soft phones . Recently i add also telephone (IP/VOIP) except of soft phone. the IP telephones are working properly but now  I need to export the contacts from the soft phone to IP telephone.

I am looking for a step by step instructions how to do it.
0
could I have 2 incoming skype phone numbers

can I receive (not send) sms to skype united states phone number
0
Anyone knows if there is a way to change a hostname in Polycom SoundStation 7000?   So it would broadcast an actual hostname instead of the make/model name.
Easier to manage and troubleshoot.
0
I have a client experiencing voice quality issues and their voice vendor is asking us to perform some Wireshark packet captures.  They have a Netgear GS724TPv2 PoE switch, which supports port mirroring, so according to NEC's instructions I mirrored the ports the PBX and IP phone are connected to and made the port my laptop is connected to the probe port.  I run the captures, making test phone calls from the IP phone, but NEC is saying the captures aren't showing any SIP traffic to/from the PBX and the IP phone.  

I don't run packet captures very often so I'm hoping an expert out there can confirm if the configuration I am running the captures in is correct?  Thanks in advance!
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Does COMCAST have poor quality phone service?

I switched yesterday from Verizon FIOS to COMCAST and had terrible telephone calls today on the land line.

I think we were sold >100 GPS upload speed. So, I doubt the poor quality is due to a slow internet connection. But maybe the old Verizon FIOS phone number was what they called a POTS line, a Plain Old Telephone Service line. Those were the ones where you cold hear pin drop.

Did I move from a POTS Line to an Internet based phone? And is that the cause of the major drop in quality?

My new modem is...

Xfinity
Dual Band Wifi
802.11.ac

The phone cable goes into the back.

Would an Internet phone do better?

Is there anything I can do?

I really despise having poor phone quality and may go back to Verizon to escape a bad connection.

Suggestions?

Thanks
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skype windows10
automatically adjust microphone settings checkbox

does this reduce background noise

does this cause moments of silence

will I do better when speaking to a machine (voice to text)

what else does this checkbox do when checked/unchecked

automatically adjust
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I have a network with Cisco Catalyst 2960X switches. We are rolling out a new phone system in phases. We want to keep the old system in place as we put in the new phones. To this end, we need multiple voice VLAN's. Another question on here pointed to a solution using MAC Authentication Bypass, but it did not give an example configuration. I am not familiar enough with the VoIP side to configure this, can someone please assist? (BTW, the other phone system is Allworx.)

https://www.experts-exchange.com/questions/29020081/Multiple-Voice-Vlans-on-Cisco-switch.html
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How does DHCP work with VoIP phones?  Do they phones need to get restarted every time the DHCP lease expires?  Or it works no different than a regular computer client?
For the first time phones need to be booted to obtain IP addresses.  Also when DHCP or DNS server changes.  What about a regular DHCP refresh at the end of the lease duration?
We use Polycom ip phones over poe.
Please advice.
0
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Avaya IP Office 500 switch/hub going into a patch panel. - All seems OK

Running Avaya  IP Office 8.1 Manager on the PC. Our phone system is no longer greeting callers. It just rings and rings and rings.
I'm able to ping the gateway and static ip. Everything was working up until a few days ago apparently.

I tried rebooting the Avaya and also the IP Office 500 Switch.

Any ideas or direction is greatly appreciated.

TY All...
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I want to make a Call Manager unavailable for CIPC registration. What services would I need
to shut down so that any CIPC that attempts to connect goes to the secondary in its
configured list of CUCM servers?
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I've been tasked with the possibility of upgrading our offices phone system.
Our current phone system runs off CAT3. Newer systems CAT5 or better. Our network is pretty old. I don't have CAT5 everywhere there is a phone and not even everywhere there's a computer. Those users end up on wifi. Anyway, from what I gather we can use a passthrough on the phones to connect the phone then the computer back to the network. We've got about 55 users and about the same amount of phones in our main office. Our plan was to also roll out VPN phones to multiple locations as we have jobsite trailers all over several states.

Is it advisable to use the passthrough on these phones? Or is having a dedicated line going to be significantly better? We also have a few places in the building where I've had to add a few small switched and I'm concerned we'll get some call degradation because of this.

Like I said, our network is a bit older and a lot of it has been band-aid fixes by adding some wifi AP's and daisy-chaining a few small switches in some areas for additional personnel over the years.

Personally I'd love to have our building rewired but I'm not totally sure it's necessary, however I don't want to cause more problems or increase load on a network that's barely adequate as it is.

If you need more info from my I'd happily provide it. I'd like some advice from anyone else who's done similar upgrades or implemented phone systems on current networks.


Thanks
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Hello,

Could someone help resolve issue where our SIP dial peer is ignored/mismatched and explain how to fix it without detriment to the existing call routing???
 
THE SETUP

We have an existing UK office collab setup which consists of:
 
CUCM (9.1) <-> ISR2921 (15.2(4)M4) <-> PSTN (ISDN30)
 
Currently all calls are routed from the CUCM to H323 gateway then out to PSTN.
We are in the process of testing SIP as a possible way forward (to replace PRI connection at some point).
 
I have setup everything in terms of SIP there is to be configured (as far as I can tell).
I have also configured route pattern on Call Manager to test calls to my mobile.
 
Calls come from CUCM to h323 gateway, get matched by multiple dial peers but in the end are send out the old PSTN connection. I am attaching debug dialpeer result for my test call -> test_call_debug.txt for anyone interested enough to take a look.
 
Here is my voip/sip config:

voice service voip
 ip address trusted list
  ipv4 172.0.0.0
  ipv4 46.165.252.40
  ipv4 5.150.254.205
  ipv4 72.251.241.166
  ipv4 83.222.249.39
  ipv4 54.172.60.1
  ipv4 54.172.60.0
  ipv4 54.172.60.2
  ipv4 54.172.60.3
  ipv4 35.156.191.128
  ipv4 35.156.191.129
  ipv4 35.156.191.130
  ipv4 35.156.191.131
 address-hiding
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 h323
 

Open in new window

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Does skype limit callers to 50 calls per day

https://voicent.com/kb/index.php/general/370/skype-limits-outbound-calls-to-50-a-day

This page is old 2009; maybe skype allows more now

I dont know how to test because I call less than 10 per day
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I've got a NETGEAR  GS724T V4 Managed switch with a VLAN set up using the Auto VoIP setting for 6 ports to serve VoIP phones at a client's location. The other ports are set to default. The IP for the switch is 192.168.1.200.  The switch is on a 192.168.1.1 network.


1) Do I need to set up another subnet for this VLAN (192.168.2.1), or will these ports just be isolated from the traffic on the other ports of this switch.
2) If I do need to set another subnet, how do I set either static IPs or DHCP for these 6 ports on the subnet?

Thanks lots for your insight.
0

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).