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

What is cube ??
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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.
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Hello everyone,
We have a ShoreTel VoIP system at out site and we use the Softphone in our board room to make calls, have small teleconference calls and calls into Spiderphone for larger teleconference meetings.
We ran into an issue during a teleconference meeting using ShoreTel Softphone to dial Spiderphone for a larger group.  When a video file was played on the same computer as the ShoreTel Softphone the participants on the other end of the teleconference could not hear the audio from the video being played.
Is there a way to fix this or do I need to just start using GoToMeeting?
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Hello, I work with asterisk servers and mainly soft SIP clients. Customers would like to have the possibility to hang up an incoming call. That's the first step...no big deal. But what if the call is a "call group" call i.e. a call to multiple devices at the same time and you want the command from the client not only to hang up his device while the others keep ringing, but to hang up all legs of the call - caller and all callees. Is that possible?
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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
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We have a Cisco phone system, we have a new need to dial international numbers and I can't get it to work. It just goes to a busy signal. I looked at the Route patterns etc and cannot figure it out. Is there any simple way of enabling this

Cisco Unified CM Administration


System version: 11.5.1.10000-6
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Does anyone know the simplest process to export Basic Skype contacts to Skype for business contacts?
Skype for business doesn't allow me to import *vcf extensions.
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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.
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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
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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.
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Hi

I've been wrapping my head around this for quite some time - still accomplishing nothing but headache. Here's the setup

Lync 2013 mediation server
Lync PSTN Trunk points to Avaya Aura RED
Avaya Aura RED routes PSTN traffic to Avaya phone central

We have landline numbers in Lync, and cell phone number with a PSTN provider

Lync <-> avaya RED gateway <-> Avaya phone system

Also - we've setup so that calls to cell phone in PSTN, will also simultaneous call Lync number.

so;
1. user call my cell phone (tel: 99 99 99) from his IP-phone/cell phone
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I answer the call with either cell phone or lync - the ringing will stop on the other device

so far so good.

But if this process is repeated, but this time the call is initiated from a Lync client at the internal office

1. user call my cell phone (tel: 99 99 99) from enterprise voice enabled lync client
2. this will call on my cell phone, but will also trigger a call to landline on lync (tel: 22 22 22). Since we've  set this up with PSTN provider
3. this will also make Lync call (at 22 22 22)
4. so the user called cell phone at 99 99 99 and both cell phone will ring, together with Lync client.
5. if I …
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Can anyone help me get a VG224 configured with FreePBX? Here is the ios config I am working with, and i will need info on how to setup FreePBX (or Asterisk) to use the VG224.
!
version 15.1
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname msr-vg01
!
boot-start-marker
boot-end-marker
!
!
enable secret 5 this.is.the.enable.secret
enable password this.is.the.enable.password
!
no aaa new-model
ip source-route
no ip routing
!
no ip cef
!
!
no ipv6 cef
!
!
!
!
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
 fax protocol pass-through g711alaw
 modem passthrough nse codec g711alaw
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
!
voice class codec 1
 codec preference 1 g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
!
voice-card 0
!
!
application
 service dsapp
  param callWaiting TRUE
  param callTransfer TRUE
 !
 global
  service default dsapp
 !
!
archive
 log config
  hidekeys
!
!
!
!
!
interface FastEthernet0/0
 ip address 10.0.0.2 255.255.252.0
 no ip route-cache
 duplex full
 speed auto
!
interface FastEthernet0/1
 no ip address
 no ip route-cache
 shutdown
 duplex auto
 speed auto
!
ip forward-protocol nd
!
no ip http server
!
!
snmp-server community public RO
!
control-plane
!
!
voice-port 2/0
 ring cadence pattern11
 alt-battery-feed feed2
 cptone GB
 timeouts call-disconnect 5
 timing hookflash-in 110 70
 

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I've been asked to setup QoS on HPE 5130 to prioritise VOIP traffic sent on port TCP/UDP 5060.

I can't VLAN the devices as these are normal office PCs running softphones and so need to differentiate by the protocol.

I'm sure I'm missing something obvious but doesn't appear to be achievable from the GUI

Can anyone give me a brief nudge in the right direction regarding CLI commands?
I've examples for other switches but for some reason are struggling with these.

Many thanks,

--Greg
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Hi Guys

we have a voip system and one of the our phone panasonic makes strange connection to domains. I would like to block all connections and only allow to connect to our VOIP provider.

Could you guide me how to do this in Sonicwall ?

Thank you
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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
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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?
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we have a hanful of SG500X-48P switches, but i noticed that the person before me never configured the voice vlan settings (attached screen shots).

switch default vlan 11
voice vlan is 14

end user ports are all general, (vlan11 default, vlan14 tagged) ~ pictured.

we've implemented a new phone system and everything has been working good, but i'm wondering if switching the voice vlan setting in the switch to 14 would benefit anything, mess up anything, or not do anything at all.

thoughts ?
sg500-vlan-members.PNG
sg500-voice-vlan-settings.PNG
sg500-voice-vlan-settings2.PNG
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Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
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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
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Hello,

Does anyone have experience with setting up QoS across a site-to-site VPN tunnel, whereby a portion of the WAN bandwidth is reserved and dedicated to the tunnel itself or certain endpoints and service ports.  I have a remote network with a Sonicwall TZ400 and also am working on setting up a Mikrotik Cloud Core router with a configuration like I have on the sonicwall.  I am looking to do QoS for VOIP traffic.  Our phones are at the remote side and our PBX is on the main side.  I am unclear on whether playing with QoS settings on the remote VPN side has an impact on the WAN traffic shaping because it is a separate interface / network than LAN to WAN traffic.  Ideally I would like to have steps on getting this working on both a sonicwall and a Mikrotik.  I don't want heavy load on my remote side WAN to impact the quality of calls across the VPN for my SIP phones.  Thank you.
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I am using napcodvr in India and installer is saying to recharge everymonth to watch in mobil
0
Hi

We have a 3CX voip server that is hosted in the cloud and we have a T49G phone we need to configure to work with the 3CX server.

Now, 3CX server doesn't officially support T49G phone but they say it should work as a normal SIP phone without any provisioning.

I have tried simple SIP config by putting in the extension number and its password but the SIP registration keeps failing.

The packet capture on the 3CX server is showing 407 proxy authentication error.

Can someone help me configure Yealink T49G phone on a 3CX Voip server?

Happy to provide packet capture or any other logs you may need.
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the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
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Hello all,

we are currently using IVR designer with admin rights. Our corp is removing admin rights on the machines. Is there any way to user Avaya IVR designer without admin rights on machines?

Regards
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).