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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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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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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!
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Is there a way to use google voice with a VoIP phone that has only Cat 5/6 port ?
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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
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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.
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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
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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
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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?
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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
I use Audacity for Mac,

Thanks
0
I have an old CallManager (4.3). it works great and no one wants to upgrade it. I have several small offices and individuals working from home offices and in order to have working phones in their locations I have to do site-site VPN's to each location.
Is there way to create some port forwarding and avoid VPN? Which ports? Any downsides?
The firewall is Cisco ASA5510 and they have Cisco 7941 and 7970 phones if that matters.
Thanks!
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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
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?
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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?
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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?
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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
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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
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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
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

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).