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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Hello,

My company currently it's moving to a new phone system and we are stock. our DHCP it's set to IP Scope 192.168.16.xx and I created a second Scope 10.11.0.xx so it can connect via VPN tunnel with the VoIP system of our another office (we are in So. Cal and the other office in Florida) now, To my knowledge I need to create the scopes and the services on DHCP so I can setup the relay to ensure that traffic can go from the 10.11 network using the 192.168 network as gateway and at some point  create a VLAN in my switches to route.

I did all the first part until before the VLAN part, I have some problems.

1-Computers on my Scope 192.168.16.xx are registering on the 10.11.0.xx I need to know how to stop them from doing that, I need to keep them alive but without merging

2-Do I need to create a vlan to route all my VoIP traffic ? we have layer 2 switches and the router it's managed by our ISP or Do I need to setup a a new port in my firewall with that subnet routing all traffic from 10.11 to the public IP

I have a VM running server 2008 R2 as my DHCP I have 2 virtual NICS installed one running on 192.168.16.xx and the other on 10.11.0.xx
I have RRA installed with IGMP installed, and my gut tells me that I did something wrong

I have not done something like this in years so if there is anyone that can give me some guiadence I will really appreciate it.
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Hi

We are currently evaluating option to move our voice to hosted.  We are in the process of two part project for this.  1st is migrating from VPN to MPLS.  The 2nd is to move from PBX/SIP to hosted.

Currently using Shortel and planning on gamma.

Anyone gone this route and suggest any options or caveats?

Thanks
1
Being a network administrator, among other things, I'm often asked by users to open ports in a firewall.
Usually the users don't know much about what they're asking for so they can't answer any questions - just forward what their technical people have provided.

Here is a typical example for a VOIP system:

The full network information for the VoIP system is:
Port Range (Audio): 35000-65000 UDP
Port Range (SIP): 5060 UDP, 5061 TLS
Port Range (Configuration Servers): 1024-65536 TCP source port, TCP Destination ports: 80, 443, 1443, 2443, 6716,
Port Range (Presence Servers): TCP Destination ports: 5222 and 5280.
I guess that's all well and good if you understand the context but that's where I'm not the expert.

I can set up firewall rules but, being conservative, I don't want to open incoming ports just willy-nilly in order to assure that the requestor gets what he/she wants.
If I ask them: "Are these incoming ports or outgoing ports?" they have no idea.
In some cases, I'm sure that some are outgoing.....
What I'm used to, for the most part, is that all outgoing will be allowed and all incoming will be blocked unless initiated by outgoing traffic.
Given this limited view, I would want to set up to allow incoming traffic to certain ports and leave things at that.
But, which ones?

I know this is likely a naive question.
So, in my context of understanding, how would you interpret the specification above?
And, in the details, I've never set …
0
I had this question after viewing "Incorrect Username or Password" on log in.

After setting up a new VoIP phone system from Comcast Business on our network, which required re-configuring our Dell network switch with VLANs for voice and data, we started to see issues with users not able to login to the network even though their credentials are valid. I would like to know if others have a similar experience and if so what is the best solution to avoid this kind of problems. Also, I am still trying to resolve the login problems for the users and the only way I have been able to use thus far is to have the user reboot their PC and then they are able to login again. I had similar problems with my domain admin account randomly on different servers. Why is it that on some servers my login works and others it does not?
0
Hi All, I am attempting to configure an AAPT IP based trunk in 3CX via a dedicated TPG SIP service, and am struggling to get it working.

What I really am after is examples of working AAPT trunk configurations that I can compare my set up to.

If anyone out there could provide some examples of correct trunk configuration, I would be extremely grateful.

Wireshark shows OPTIONS messages successfully hitting the phone system from TPG SIP server, and 200 OKAY messages being sent to TPG SIP server.

EDIT:

 - 3CX on premise
 - Wireshark shows OPTIONS being received and 200 OKAY being sent to/from TPG SIP IP
 - Dual WAN connection with dedicated SIP service on WAN2
 - NAT on WAN2 to local IP of 3CX
 - Static routes are configured to route required SIP traffic in/out via either WAN1 or WAN2 depending on what port is required

Kind regards,

Nick
0
Hello,

I have a problem with an asterisk server:

I have a SIP trunk from Vodafone. When I call from another provider ( lets say Orange ) redirecting from Softphone to another extension works. When I call from the same provider ( Vodafone - my sip trunks use vodafone ) and try to redirect from the softphone to another extension, the call is intrerupting.


This is extension.conf related to extension number used for redirecting: (I masked the real number)

exten => _yyy268,1,Set(CALLFILENAME=${CALLERID(num)}_${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)})
exten => _yyy268,n,MixMonitor(/var/inregistrari/in/${CALLFILENAME}.wav,b)
;exten => _yyy268,n,Goto(ivr-liber,9,1)
exten => _yyy268,n,GotoIfTime(18:00-23:59|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,GotoIfTime(00:00-08:00|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,Dial(SIP/268,20,tk)
exten => _yyy268,n,Dial(SIP/241,60,tk)
exten => _yyy268,n,Congestion()
exten => _yyy268,n,Hangup()

THis is sip.conf related to extension used for redirecting:

[268](sets)
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
notifyringing=yes
call-limit=2
callerid=268
nat=no
mailbox=268@default
secret=sssssssss
canreinvite=no
callgroup=1
pickupgroup=1


[241](sets)
dtmfmode=rfc2833
t38pt_udptl=yes
qualify=yes
notifyringing=yes
call-limit=2
callerid=241
nat=no
mailbox=241@default
secret=rrrrrrrrrrrr
canreinvite=no
callgroup=13
pickupgroup=13



THis is sip.cinf related to the trunk used:

0
Greetings,

We are having an issue where we receive a call that rings all phones in our ring group with a 3CX phone system. When a user answers the call, no one is there. The call disconnects, then the next extension in the ring group will ring. The phones will continue to ring until every extension in the ring group answers the call.

There is no caller id showing on the phones, and no one on the other end of the call. Furthermore, the issue is reoccurring once every week to two weeks. It happens on a random day and time. We are not able to replicate the issue and is affecting clients that do not have a IVR.

We've open a ticket with our SIP provider and 3CX. Our SIP provider says that when this occurs, a SIP cancel is sent to the 3CX PBX but the PBX does not respond and the call gets caught up in the system. 3CX says it's a know issue with the phone system but has been over 2 months with no resolution. Any help would be appreciated.
0
Cisco 8851 VoIP phone.

Trying to setup Personal Directory for my contacts.

On the phone Login it asks for my UserID and PIN.

I have no idea how to determine my NetID/UserID.

Used my phone number.

No luck.


I do not have a pin, but I used the default 343842.
0
hi all,

is this possible.

network 1 - 10.3.3.0
network 2 - 192.168.50.0

they are connected via VPN. all traffic is flowing nicely apart from the phones.

the client has bought a VoIP phone system which needs to be on the same subnet, is it possible to 'trick' the 192 network to have a 1x 10.3.3.0 IP address on its network so that the phones can talk back to the phone system? And then to have the routing on the routers to move the traffic correctly.

Thanks
Gareth
0
Able to connect to a network resource okay when on Wi-Fi, but when connected to Ethernet, File Explorer hangs.

I suspect that the problem is the Ethernet connection speed, for instance, since the computer gets it Ethernet connection via a VOIP phone, which only gives the computer 10 Mbps.
The computer was replaced with a faster model.
Windows Updates were installed.
Reliability history shows the every time the File Explorer stopped responding.
My thinking is problem could be on the server itself, maybe I need to enable SMB on the system.
What else should I be looking at?
0
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Putting together quotes to replace a 12 year old phone system and VOIP is an option, specifically Ring Central. They claim they have no hardware to install and don't even require QoS which I find hard to believe. They use Polycom phones that come with a high rating as well a Cisco option that comes with a high rating.

Question, has anyone actually witnessed a VOIP phone work without any additional hardware and call quality is excellent? My approach is if this is an option great for all parties involved. If it's too good to be true I prefer to put the right equipment in place to avoid any frustration after installation.

If you've used Ring Central even better, if you haven't but have success with another company please share.
0
HI Experts.

I have this policy map on most of the switches at my organization.  
Policy Map AUTOQOSPOLICY

    Class AUTOQOS_VOIP_DATA_CLASS
      set dscp ef
      police 128000 8000 exceed-action policed-dscp-transmit
    Class AUTOQOS_VOIP_SIGNAL_CLASS
      set dscp cs3
      police 32000 8000 exceed-action policed-dscp-transmit
    Class AUTOQOS_DEFAULT_CLASS
      set dscp default
      police 10000000 8000 exceed-action policed-dscp-transmit

We are now replacing the existing phones with a new cloud base phone system and they sent me these requirement for QOS and the vendor gave me this policy to use on the switches

policy-map PM-ASW-IB-User
class CM-ASW-IB-RC-Voice-RTP
set ip dscp ef
police 512000 16000 exceed-action drop
class CM-ASW-IB-RC-Video-RTP
set ip dscp af41
police 768000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-GeneralSIP
set ip dscp af31
police 32000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-Meetings-Control
set ip dscp af31
police 32000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-Other
set ip dscp af21
class CM-ASW-IB-Cust-AF13
set ip dscp af13
class CM-ASW-IB-Cust-AF12
set ip dscp af12
class CM-ASW-IB-Cust-AF11
set ip dscp af11
class class-default
set ip dscp default

Apply on the ports :

interface range Gi1/0/9-20
! no mls qos trust device cisco-phone
! no auto qos voip cisco-phone
! no mls qos trust cos
! mls qos trust dscp
! priority-queue out
! …
0
Hi,
I have a big problem with Cisco voip configuration. I have two CME router which is connected by IPsec over gre tunnel vpn. The flow between router Irak and router CI is correct but we can not make any calls, we heard a busy tone. However the calls between Router CI and Router Irak work well.
I don't know how to fix this issue. I need your help please.

Best Regards,
puttyrouterci.log
puttyirak.log
0
I was trying to bring up a SIP over IPSec peering on a CIsco ISR4451-X. This is to complement and already existing
SIP router (CUBE I guess they call it). Anyhow I had a call not connecting and I ran debug ccsip call. One weird thing
I noticed is that the Source IP Media address 136.10.23.97 address on the external Gig 0/1 interface of the router.
The loopback is 136.10.23.98 and that's what I'm expecting to see for Source IP Address (Media). What determines
what interface/address gets used for Source IP Address (Media)? Thank you.

001812: Dec 17 2018 23:26:18.025 PST: //747/E74BC54D82EC/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 136.10.23.97        
Source IP Port    (Media): 17876
Destn  IP Address (Media): 216.25.35.21
Destn  IP Port    (Media): 55414
Orig Destn IP Address:Port (Media): [ - ]:0
0
I'm wanting to setup QoS for Skype for Business Online. When I do a packet capture I see that the real time
ports that come into play are the UDP 50,000 - 59,999. The article below calls the 50,000 - 59,999 as optional.
Is there any way through group policy to tell skype to use the UDP 3478, 3479, 3480, 3481 only or at least
to prefer it? Marking all TCP/UDP 50,000-59,999 for EF classification seems pretty broad.


https://techcommunity.microsoft.com/t5/Skype-for-Business-Blog/Simplified-port-requirements-for-Skype-for-Business-Online/ba-p/77094
0
Is there software that tests noise on CAT5e and CAT6  cabling? I hired a company to test the cables and they said everything is above 1G and for the most part I believe them as I witnessed them test drops and make repairs on several damaged jacks. My concern is they didn't include a report with their invoice. Extremely disappointed as I mentioned the report was imperative for us to decide if VOIP is an option for my client.

I've insisted they put a report together and I'm aware the Fluke they used provides the one needed. While waiting on the report, and my suspicion is they forgot to turn reporting on, is there any software available that can give an indication of noise or how well VOIP will work on my clients network. Everything tested above 1G but there are still concerns I would like to eliminate. One being noise.

I've tried Solorwinds and wasn't impressed. Any other options available?
0
Please tell me I'm wrong:  When using S4B to call a business that has a telephone auto-attendant, our S4B dialpad works just fine.  However, if I and an employee call a business together in a S4B call, the dialpad buttons do not work.  MS tends to suggest that this is a known bug.  We're about to agree and leave it at that... and leave S4B.

But really?  What an obvious thing to need to do.  We REALLY need to do this to train our employees on calling clients, etc.

MS seems to be moving from S4B to Teams.  Teams seems to be entirely geared toward pre-scheduled meetings where all attendees have agreed to join.  This is not our need AT ALL.  We need on-the-fly ability to add a voice call to an existing voice call AND be able to punch a dialpad for auto-attendants.

Therefore, we're looking for economical (5 users or less) solutions for our VOIP needs.  MS Office 365 E3 (which we'll keep) runs us $20/month, but in addition, for Skype PTSN dialing we also need $12/month/user for Domestic Calling Plan and $8/month/user for Phone System.  Therefore, our phone system needs are $20/month/user.

Can someone recommend some economical VOIP solutions? Thank you, yes we've looked - but that process is EXTREMELY unproductive (i.e. false/misleading claims on websites, feature listings are incomplete, etc., etc.)

Thank you!
0
When our Mitel 5330e is connected directly to the wall jack it works and when the computer is connected to the wall jack data works however when we connect the p.c. to the back on the phone jack it does not work.   On the compute we get a message media disconnect when doing an ipconfig.  We have checked all cables.

Switch config
switchport access vlan 126
switchport mode access
switchport voice vlan 180
priority-queu out
mls qos trust dscp
spannin-tree portfast

Our Firewall is a Fortinet

Phone gets IP address from DCHP on 3300

Computer get IP address from DCHP firewall
0
Hi Experts,

I am able to access the call manager in our organization, I have a phone device and I can see it under Device --> Phone but I want to know how an anolog phone with DID phone number  will connect to call manager using internal extension usually using the last 4 digits as internal ext,

If the product Type Tye says : Analog Phone , does that mean it is a analog phone.
0
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If I have two SIP routes - model 2951 ISRs CUBE - and you want call manager to
failover if one of them can't complete a call - what is required? We currently have
a SIP trunk to one ISR (and the ISR has a TIP trunk to our call center). For redundancy
we want to add a second ISR/SIP Trunk. But the second should only be used in the
event that the SIP peering on the primary goes down. Advice appreciated.
0
Im wanting to host simple PABX for multiple custmers and not sure which vendor to go for
we will host about 40 PABX's each with around 5 phones attached
requirements
support failover we will have instance in 2 separate DataCentres's for redundany/failover
needs to have single SIP trunk to host for all the calling voice channels
need to be low cost around 1$-2$ per extension and scalable as well
meed to be simple to provision and reliable
will be using yealink phones

any recommendation would be great
0
Hi,
I would like to understand this process a bit more and the authentication flow.  Using ClearPass (similar to ISE) as a RADIUS server.

PC authenticates successfully via dot1x (EAP-TLS) when plugged into jack.  However, when plugged in via VoIP, it fails.   Discovered that the pC is not able to auth via MAB because the MAC is not in the MAC Address table.   Once added MAC to MAC Address table, PC successfully authenticates via dot1x and MAB.

What is the relation to VoIP here?  If the PC can auth successfully via dot1x(EAP-TLS) on its own, what triggers the PC to roll over to MAB and fail?
1
One of the Shoretel Server Services showing RED. ShorewareCDRMigration-UPG
Does anyone knows about this service? It seems that server working fine. Thank you!
0
How can I set my Skype default options to be that anyone can present/share files when Skype is opened? My current set up means that I have a Skype meeting and then when the person wants to share I have to change the settings and then re-commence the meeting? It's set so that only people from my organisation can share...thanks
0
I have some new VoIP phones and for some reason they will not configure on my clients network, when i took them home they work perfectly. I tried Wiresharking on a hub to capture the traffic, however i am at a loss as to what it means of what is causing the issue. The DNS is our Win2012R2 server and this then forwards on to the public Google servers.
wireshark-capture.png
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).