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 Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
!
! No configuration change since last restart
!
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
!
hostname CME-EHM
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
card type e1 0 1
!
no aaa new-model
!
subscriber templating
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
!
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
  ////////////////////////****////
!
voice service voip
 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
  call start slow
 sip
  bind …
0
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Hello, I'm curious, hence I ask you; Is there such a "recent college engineer", either software or hardware, or one that studied both, who would meet all of these requirements:
embedded Linux based data telecommunications system/ VoIP/SIP
satellite network application and
embedded software/ Firmware/embedded systems
automation scripting
Python
Java
Linux operating system/Linux Network expertise
Unix shell scripting
WEB GUI test automation
DO-178B experience
Satellite communication experience
Telecommunication experience/PBX switching systems

Any comments?
Do companies expect to find an "expert" engineer with all this knowledge?
thanks
0
In RTMT you can view "Gateway Activity" and choose say MGCP PRI. But the thing that's weird is that it shows PRI channels per CUCM server. Given the description I would think that it would make out how many active calls are happening at the router/gateway not the UCM. Is there a way to get this broken out to you see activity per actual gateway?
0
If you are using MS Lync to make a phone call to the PSTN and there are multiple SIP trunks to the PSTN (actually these are trunks to Cisco CUCM and then on to PSTN from there) - how does Lync decide which SIP Trunk to use?
0
Hi

I got 2 switch, A5120 and S5500 serie switch from h3c, the a5120 is the lan switch and the s5500 is the poe ..
I would like to know the exact config I should use for both switch, I have already do config but not working .

I join the two config switch, the poe (port 48) is connected to port 10 on the a5120 ..

Thanks for the help
s5120.txt
s5500.txt
0
When a user dials in to a conference, they are prompted to enter the conference ID followed by pound.  When they do they system doesn't recognize they are entering the conference ID and the auto attendant states either that it did not get that or that the conference ID is invalid.
This is not with all phones but it is consistent on the ones that it does not work.
This is a three server frontend pool with a ACME Packet SBC and Mitel switches.
It was working perfectly then suddenly this started happening.
What might be causing this?
0
This is a general question to show my ignorant side.  We have construction going on at my business and they need to widen the driveway.  There are wires that need to be re-located for this construction project and the general contractor notes that there is 1 T1 lines and several fiber lines.  The fiber lines, I know, are run by the local cable company to provide us with digital phones and broadband internet.  I am guessing that the T1 is for the PRI for my ShoreTel phone system.  Is it possible to service my phone system some other way via those fiber lines or is the T1 the only way that I will be able to function?
0
Hi,

I'm receiving the attached error and would like to know how do you actually verify connectivity between these two? I mean the servers can ping and communicate on all ports, but is there a way from GUI/CLI to try to reconnect them?

Thanks,
ELM-Server-Error.jpg
0
What are the advantages / disadvantages the Office 365 VOIP solution has over other VOIP solutions for small business?
0
Hi,

Running CUCM version 9.1.1 and I'm seeing a lot of reverse lookups, they are failing because my AD server is not setup to accept those but what I wonder is it normal to see so many? what causes the CUCM to execute these queries? I can see like 2 million request in the last 8 hours. You can see attached a few examples.

Thanks,
CUCM-queries.jpg
0
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We recently moved our CUCM 10.5 publisher to another data center. Call have been mostly good.
But we ran into a period where callers were getting this recording
"Call not allowed due to restrictions on your account". Can the Cisco
Unified Communications Manager 10.5 possibly be responsible for
that recording? Or would that indicate a problem at the provider?
0
Please provide me with a list of services similar to Magic Jack that allow calls to be forwarded to another phone number.
0
Hello Everyone,

My name is George and I was hoping to get some assistance in configuring a Cisco 7914 expansion module.  I have been thrown into the fire with this Cisco phone system.  I have very experience with the CallManager system but I am making some headway.  The biggest problem is that my company is running very outdated versions of the software.  We are running CallManager Administration 4.1 (0.11).  I know, I know...it's no way to run a business but that decision is above my pay grade!  I am told to make it work, as I am sure many of you can understand and appreciate!

So here is what happened.  We had to do a complete network reboot and I was configuring a new phone when they took everything down.  When the network was brought back up, it took a few minutes and all but one phone came back up.  Of course, the one that did not come back up was the receptionist's.  After some troubleshooting I found that somehow, her phone was no longer showing in CallManager.  I went through the process of adding the phone again and defining it and configured.  Her phone works fine now.  The problem is with the 7914 expansion module.

The module is showing all red lights on the buttons and the display is blank.  I went to Add/Update speed dials, entered all the name and extension information and clicked on Update and Close.  I thought this was all I needed to do but I was wrong.  In the Configure Speed Dial Settings for SEPXXXXXXXXX window, it says 'Speed Dial Settings not …
0
Hi,

Ive been given the following to configure on the above network switch for our SIP/VOIP BT Cloud Voice service, can anyone assist on how I can get this done? I have no CISCO OS knowledge but have gained access into the switches gui so just need to implement the rules below.

The BT Cloud Voice Handsets are already live and on the network and they are have PC's also plugged into the pass through ports in most cases.

qos tcp-port 5060 dscp 011000
qos udp-port 5060 dscp 011000
qos type-of-service diff-services
qos dscp-map 101110 priority 5
qos dscp-map 100010 priority 4

qos device-priority 62.239.32.224/28 priority 5            
qos device-priority 62.239.32.240/28 priority 5              
qos device-priority 147.152.35.104/29 priority 5          
qos device-priority 147.152.35.96/29 priority 5                        
qos device-priority 62.7.201.128/27 priority 5
qos device-priority 62.7.201.160/27 priority 5
qos device-priority 213.120.60.128/25 priority 5

Thanks
SycamoreIT
0
I own an Avaya ACS509 R7 BUSINESS PHONE SYSTEM NAMED PARTNER. One of the extensions,the main programming extension number 10 is not functioning 100%.There is an illuminated button that can be designated as DND (Do Not Disturb). The button does make a click in the receiver but does not turn on the DND feature nor does the led come on. I have tried several identical phones on that  line and none of them work. All of them work different extensions. So, I must assume it is the main controller board. I am an Electronic Technician and probably can repair it IF I could locate a complete schematic diagram along with pcb board layout pictorials and part numbers/values. I do NOT need the programming manual or user's manual. ONLY a technical repair manual. They must exist somewhere since there are several locations that will repair this motherboard. I would really appreciate it if any Expert could direct me to a pdf or similar for this unit. It is also referred to as an ACS-R7 unit. Anyone assisting me will be the recipient of 735 virtual Kudos!

NOTE: This is NOT a voip system. Just a regular 20 extension phone system.
0
I'm trying out a few different UK network sim cards.  For EE under the Enable 4G setting it offers, OFF, Voice & Data and Data only.

What has voice got to do with 4G because the other network sims dont mention Voice and data only data.
0
hello expert i have problem with DT700 IP phone please give me a way about how to solve double assignment issue at ip phone
thanks for All
0
VoIP ISP
Why do some people recommend buying business VoIP from an ISP? What are the benefits to my company? What are the costs?
0
We have a UC520 ver 12.4

Currently I have a ephone hunt group that goes through a list of numbers to dial.
its tied to a external phone number door buzzer .
I would like to change that to a parallel so all extension listed ring but the parallel option is only available on voice hunt group not ephone hunt.

 ephone-hunt 1 sequential
 pilot ******** secondary 110
 list 103, 100, 125, 116, 109, 101, 111, 114, 122, 102, 127, 128, 107, 106, 105, 119, 108
 final 103
 preference 1 secondary 7
 timeout 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20

here is an example of a internal hunt group (not tied to external number just extension 200)

voice hunt-group 1 parallel
 list 100,101,102,103,104,105,106,107,108,109,111,112,113,114,115,116,117,118,119,120,121,122,124,125,127,128
 timeout 6
 pilot 200

Basically I need to combined these 2 types of hunt groups into one that works for me.
Door buzzer dials phone number ******* which dials all listed extensions in parallel, and whoever answers first can buzz the person in.
0
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I need a Google phone number that will ring on my cell number, for my business cards.

That gives me some flexibility if that number gets spammed, I guess.

Do I need Google Voice?

Could you provide me a link to get that number reserved? And was is Google Voice?

Thanks
0
I know that I can make free phone calls from Amazon Alexa Echo to another Elexa Echo, but what I want to research fromn my customers is the ability to uise a voice command to Alexa Echo to call a cell number or call a contact.

I want to research how this could be done.  

I'm thinking integration things like a combination of:
  • VPN's to some service,
  • integration with a Personal Contacts app such as Gmail,
  • Use of IOT
  • ITTT

Anything thoughts and directions to get me started would be very helpful.

Thank you,
Robbie.
0
Hi,

I would like to know how I can send messages using an Amazon Echo to a mobile phone as well as being able to call any phone.

In relation to this:

- The phone being called can be landline or mobile
- It would be a call from an Echo to phone and not Echo to Echo
- The Echo should be able to receive message and calls as well as send
- Would it be free to send messages and make calls in this way
- Also, would anything else be needed to make this work

Thanks,
Robbie

My ref: 1029743
0
I am new to Cisco 7940 phones. Have configured 5 phones which are working well in my office. But two more phones I am trying to configure. When I plug in the phone it shows error "Protocol Application Invalid". I did a factory reset. Still no result. Can anyone give me a solution, please?
0
The topic is setting up remote admin GUI access for FreePBX on a virtual server. There is not an issue to resolve at the moment. The remote admin access will need to be set up in a couple of days and want to learn the correct config before attempting trial and error.
We have been using a different PBX app in the past, which was not compatible with a virtual machine environment. For that app, we had access to the admin interface via SSH port 443, and am not certain the same will apply for FreePBX
Thanks in advance ...
0
How can you monitor DSP usage on a Cisco ISR used for MTP, transcoding, conferencing? Is there an OID which would let  us graph this?

mtp01-sc#sho sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

50630987   251788306    mtp   sendrecv g711u   24918 24576 172.28.72.145                                                                
50630987   251788304    mtp   sendrecv g711u   31384 18902 172.28.16.33                                                                  
50631252   50636167     mtp   sendrecv g711u   19210 17128 172.17.254.1                                                                  
50631252   50636166     mtp   sendrecv g711u   18332 50004 172.16.24.142    
Total number of active session(s) 11, and connection(s) 22
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).