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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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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The subnet calculator helps you design networks by taking an IP address and network mask and returning information such as network, broadcast address, and host range.

One of a set of tools we're offering as a way of saying thank you for being a part of the community.

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?
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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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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
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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,
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Got a ip pbx and i want to send the voice mail via e-mail in the office we got a Miicrosoft 2011sbs standar with exchange 2010
altho i have create the account voicemail@xxxxx.org and configure the ip pbx the pbx is not able to send the email with the voicemail. i have testet the email created and work.  The pbx is nec sl1100
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I am using this Arris NVG443 router in conjunction with an Obitalk 200 to use internet (DSL)  VOIP to make and receive phone calls. The problem is no matter what service I use ie. Google voice (free), Onesuite (subscription),  ALL calls made and received drop within minutes. Using Onesuite provider  all calls drop at exactly 9 minutes every time. Obitalk support says to configure my router to this:
In order for your OBi to be able to send packets w/o interruption, please configure your router as follows:

Allow Outgoing:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060, 5061, 10000 to 11000, 16600 to 16998, 19305
Allow Incoming on UDP Port: 10000

Problem is: There is nowhere to enter all these ports? Just one range.
 Router has port forwarding and port triggering . In port forwarding there is only a global port range  one port # to another port #, and one base port.
In port triggering there is just a trigger range same as above, one port to one port, and one open port range same as above.

Not sure what to enter into router and can't seem to find any answers ?
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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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Hello all

I have set up a Cisco 2801 router.

I will be doing SIP to POTS hand off. I have everything up and working with the SIP provider. I am able to make calls inbound but not outbound.

Any help would be great.

here is my dial peer config


dial-peer voice 100 pots
description INBOUND TO ANALONG PORT 0
destination-pattern .
port 0/1/0
forward-digits 0
!
dial-peer voice 1 voip
description **CCA*North American*10-Digit Local**
destination-pattern ..........
progress_ind setup enable 3
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
dtmf-relay rtp-nte
clid network-number xxx-xxx-xxxx
no vad
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Free Tool: Port Scanner

Check which ports are open to the outside world. Helps make sure that your firewall rules are working as intended.

One of a set of tools we are providing to everyone as a way of saying thank you for being a part of the community.

We're just gotten signed up for Skype for Business through Office 365.  We have the PBX and the calling plan.  Got phone numbers too.
 Works beautifully using our computers for audio... HOWEVER, we need to be able to use our (new) VOIP phone sets.  We see that perhaps Skype for Business is VERY VERY VERY VERY limited to the phone sets it can work with.  We're not sure if that's just CYA mumbo-jumbo, or if it's really that limited.

Specifically, we have RCA Model IP170s phone sets.  (I think those are re-branded from another manufacturer.)  In setting up these phones, it's asking for SIP info... and that's about the limit of my understanding.

Do you think we can get these phones to work with Skype for Business?  Are we limited to using JUST the "supported" phones?
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I am looking to make a change to the next-hop address on an already active local-policy on a Acme Packet Net-Net SBC. I am able to make the changes and 'done' out of the config. When it presents the config following the 'done' it shows correctly. When I view the running config following the changes it still shows the old next-hop address. How do I apply this change.
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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
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I use Audacity for Mac,

Thanks
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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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Hi

With this config from POE switch s5500 connected to core S5120 i got the phone working but i could connect or ping the S5500 ip address switch ... What's wrong !!

Is it the write config for a voice vlan setup like this (Core switch + poe) phone-pc connection !!

------------------------------------------------------------

Port connected from S5500  to core  S5120 port 10

interface GigabitEthernet1/0/48
 port link-mode bridge
 port link-type hybrid
 port hybrid vlan 210 tagged
 port hybrid vlan 1 untagged

--------------------------------------------------------------------

Phone port on S5500

interface GigabitEthernet1/0/19
 port link-mode bridge
 port link-type hybrid
 undo port hybrid vlan 1
 port hybrid vlan 210 tagged
 poe enable

---------------------------------------------------

Port on core switch S5120 connected to S5500 on  port 48

#                              
interface GigabitEthernet1/0/10
 port link-type hybrid          
 port hybrid vlan 210 tagged    
 port hybrid vlan 1 untagged    
 undo voice vlan mode auto      
 qos priority 6
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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 …
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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
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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
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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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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
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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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What are the advantages / disadvantages the Office 365 VOIP solution has over other VOIP solutions for small business?
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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).