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).

Not sure if I have setup QoS for VOIP correctly to interact with RingCentral on my switch.

I know almost nothing about this aspect.  Switch is Aruba 2540.

RingCentral is seeing jitter when they diagnosed it, I want to make sure that the switch is prioritizing the voip traffic at the best priority over data but I have no idea if this is the right way to do it, or if there is a best-practice type of approach?  The phones are on vlan 10, phone plugs into switch, pc plugs into phone, PoE is used.  In this instance I just want to confirm the switch is properly configured to prioritize traffic, if the problem is upstream, it's upstream, I am just concerned about the switch at this point.

I don't know what the dscp mapping table looks like so maybe ultimately I'm missing the priority?

hostname "zzz"
module 1 type jl356a
qos type-of-service diff-services
ip default-gateway 192.168.xx.xx
ip route 192.168.xx.xx
snmp-server community "public" unrestricted
snmp-server contact "zzzzz" location "zzzzzzz"
vlan 1
   name "PCS"
   untagged 1-28
   ip address 192.168.xx.xx
vlan 10
   name "PHONES"
   tagged 1-24
   ip address 172.168.xx.xx
   ip helper-address 192.168.xx.xxx
   qos dscp 101110
no tftp server
no autorun
no dhcp config-file-update
no dhcp image-file-update
no dhcp tr69-acs-url
password manager
password operator
I wonder if you could help us with the phone configuration in Dubai.

We have a 3CX phone system in the Uk
We’ve just rented some space in a service office in Dubai
And have arranged  for Port 5060 (inbound, UDP) for SIP communications and
Port 9000-10999 (inbound, UDP) for RTP (Audio) communications to be opened on their firewall and linked to the IP addresses of the phones in the office.

We are told that the public ip is which can be used to access the services (I don’t under this part)
However the phone are showing a SIP error and not currently working.

We’ve only two full days left in the Dubai office and I am reaching the edge of my understanding.

Would you be able to assist ?

Kind Regards

Get Outlook for iOS
I have been trying to get a Grandstream HT814 to communicate with Sonetel. It works fine when connected directly to the DSL line, but as soon as I put it behind the firewall it stops. It cannot make calls, when receiving a call it will ring, but with no sound.

I have tried with the SIP inspection on and off (in the config below it is disabled)

Cisco Config:

Hardware:   ASA5506, 4096 MB RAM, CPU Atom C2000 series 1250 MHz, 1 CPU (4 cores)
ASA Version 9.8(4)12

no mac-address auto

interface GigabitEthernet1/1
 nameif outside
 security-level 0
 ip address dhcp setroute
 no pim
 no igmp
interface GigabitEthernet1/2
 nameif inside
 security-level 100
 ip address xxxxx
 no pim
 no igmp
dns domain-lookup inside
dns server-group DefaultDNS
 domain-name xxxxxxx
same-security-traffic permit inter-interface
same-security-traffic permit intra-interface

access-list outside_access_in extended permit ip any host xxx
access-list inside_access_in extended permit ip any any

access-list global_mpc extended permit ip host xxx any inactive

pager lines 24
logging enable
logging asdm debugging

mtu outside 1500
mtu inside 1500
mtu Proxy 1500

arp timeout 14400
no arp permit-nonconnected
arp rate-limit 16384

nat (any,outside) source dynamic any interface

timeout xlate 3:00:00
timeout pat-xlate 0:00:30
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 sctp 0:02:00 icmp 0:00:02
timeout …
I need to report the number of call from our cisco Call manager that tells me the number of calls for last year that were incoming between 8am and 5pm and were contact type 1, contact disposition type 2,

I tried doing this from call reporting but it will only report on 7 days at a time?
I'm setting up a 3CX PBX on Amazon Lightsail, and I'm having trouble with setting up conference calling that will allow external participants to dial in (like FreeConferenceCall.com, or other similar services).

I have my inbound and outbound calls working, so I assume my basic setup is okay. I have one number I purchased from Skyetel (the VOIP provider I'm using), and I have another number that's being ported in from Skype (not yet active on Skyetel).

However, I'm not sure how to properly setup conference calling on 3CX. I have a single extension (00), and I have all of my trunks (inbound and outbound) set with the "Trunk" value of *1949 (the last 4 digits of the number I purchased from Skyetel).

In Settings >> General >> Conference, I set my Conference Extension to 00 (the only one I have), and I've set my External Number to the number I purchased from Skyetel (the one ending in 1949).

Are there other settings I need to host conference calls?
Now that Skype no longer allows a customized voicemail greeting, I'm looking for an alternative.

I have a Skype Number that I'll port over to the new system, so it needs to accept incoming calls on that number. Other than that all I really care about is voicemail. Skype cost me about $100 a year for the number + my plan, and I'd sure like to stick around that price range.

Any ideas?
Hi Experts

I am on a personal mission to under a little bit more about VoIP.

So we run a Gamma \ Horizon system at work. All configured and working via Cisco.

Our Wifi is controlled via Unfi USG, Switches etc. I've plugged in (With Consent) one of our spare Voip phones into a spare port on one of the Unifi switches and configure on it own Vlan etc.

The phone can call external numbers no problem, but if I try to ring an internal ext the internal phone rings but once connected I can not hear the person on the other end.

I get that something is blocking it or its not configure I just do know where to start trouble shooting,

Maybe someone could point me in the right direction?

Please let me know, the best USB Headset for Skype, MS Teams and Cisco WebEx calls, under $100.
Also, share any Gadget for USB to 3.5mm converter.
I need to re-IP all the voice VLANs in a company. My first thought was to just renumber the SVIs at each site related to VOIP. But that would then cut me off from being able to reset the phones from the Call Manager because the gateway no longer works. OR would it be the case that losing connectivity to the Call Manager, the phones might just reboot themselves?

If not that - might there be a way to recycle the inline power at the switches to force the phones to reboot? e.g

int range Gi 2/0/1 - 47
    power inline never


    power inline auto

Any other thought on the most efficient means to reboot all the phones on a switch when they can't talk to the Call Manager?
Need Speech to Text service

I have an iPhone 6, a MacBook and a Windows PC. I need to find a service which lets me speak into some device and get the text version emailed to me. I do not have time for a service, which can cost $1/minute and take 12-24 hours. I need it immediately, which means the text creation needs to be near real time.

I will pay for this service, if needed. I think $1/minute is the highest I could pay.

by "some device" I meant:
1) My iPhone 6
2) A microphone hooked into a Mac
2) A microphone hooked into Windows

Need a quick web UI for a great speech recognition API

I sampled an AWS back-end API (using their demo) and found the quality to be excellent. Meanwhile, I have heard Google and Watson also have great API's.

But, my friend, who can no longer type into a keyboard, can not find a way to access any of these great API's.

Can you tell me the names of these services?

Do you know of any consumer focused front ends that would provide access to these awesome API's?

If I decide to throw together a quick front-end, which API is easiest to develop with?

I have a Cisco Voice Gateway 4331 that handles all of our calls in conjunction with Cisco Call Manager.  The voice gateway has a PRI circuit connected to a port and three POTS lines using the remaining three ports.  In this example, I want to have the internal extension 3337 use the specific port of a POTS line on port 0/2/2.  This is for a fax machine (attached to CUCM via an ATA 190) that only sends and I am having trouble with it being reliable over the PRI.  I was hoping to tie it to a POTS line to avoid trouble.

I tried the following and it did not seem to work correctly.  I feel like I am missing an important component:

dial-peer voice 3199 pots
desc ***** Send faxes over pots lines for mailroom *****
preference 1
answer-address 3337
port 0/2/2
forward-digits all

In the above, I am attempting to identify the internal extension of the fax machine (3337) so that it can be directed to use the POTS line on port 0/2/2.  Is there another set of commands that I might be missing?

I have Nec ITX-3370-1W(BK)TEL POE phone and when I attached it to Cisco 3750 POE switch the phone not working while when I attached Nortel 1120E or 1140E they are working.

Any idea to fix the problem?

Good Day,

I recently got a E1 line through digium gateway G100, then bought 3 digium d60 and 1 d80 ip phones, and then a conference phone for my office.

I have been using Wifi for over 1year now with zero issues. Use ubiquiti AC Pro, 2 on the floor. open space of 289sqm.

How can I deploy the IP phones without without having to run LAN please?
I'm dealing with performance issues with a VOIP phone system.
The VOIP service provider provides a dedicated internet connection for VOIP and the "PBX" is externally provided by the VSP.
Since the PBX is external, even internal extension-to-extension VOIP calls cause external traffic.
We provide a dedicated VOIP firewall in the form of an RV320 followed by cascaded SG300 switches - configured with a VOIP VLAN with QoS set up.
We provide a dedicated internet connection and firewall for site data - independent of VOIP traffic.
There are 3 sites, each one with separate internet connections, firewalls, etc.
The largest site has about 20 phones and 25 workstations.
The smallest site has about 6 phones and 10 workstations
The middle site has about 9 phones10 workstations.
Data traffic is modest.

I believe the VOIP system is working overall as intended so the "problems" are a matter of service quality I'd say.
Problems are intermittent and include:
- audio is heard at one end and not the other.
- a very loud "screeching noise" is heard at one end or the other and can be audible at one or both ends.  This is reported to be rather high-pitched and not like loud TV white noise.
- some incoming calls don't arrive on site and go directly to voice mail.
Overall, it's reasonable to say: "while the system seems to "work", service is unacceptable".

Since the 3 sites are each independent of the other re: VOIP, if all sites behave similarly (re: problems) then one might …
For years I have used Plantronics Supra binaural headsets along with the matching Plantronics headset amplifier/interface M10 or MX10. ( I am not at that location now.) I have the requirement that my phone audio be crystal clear at all times. I have never had any problem with clarity until yesterday. I never had considered VOIP because my internet speed was not ideal. Several months ago I got fiber and my up and down speed is 1 GIG with pings at 2ms. With this super speed I thought that VOIP would be an acceptable choice since it would save me more than 50% of my phone bill. Now that it is installed I am told that my transmitted voice is somewhat distorted and there is some sort of slight crackling in the background. I cannot live with this problem. I spoke to level one of tech support last night and he confirmed that I was indeed distorted. I have another line that utilizes the MagicJack. I phoned the tech on that line and the distortion was still present. I also switched from my headset and amp combo to a regular phone on the business phone system (Avaya Partner) and the distortion is still there. Level two is supposed to get back to me today and begin troubleshooting the problem. I was just wondering if any Expert has encountered this difficulty before? With such incredibly high isp speed the is the last thing that I had expected. If an Expert has any ideas please let me know.  

Configuration wise: On the isp's router there are two phone jacks. I go from jack One …
The phone is Polycom VVX 350, provisioned by RingCentral.

Is it possible to somehow program the Polycom phone to produce a distinctive ring if a specific number calls.  A mobile phone can do this.  I wondered if there is any way that I can get this feature? If not Polycom, is there another brand of VOIP desk phone that has this ability?

I am replacing a Cisco 2960 switch with a 9200. After copying & pasting the run config from the 2960 the interfaces are missing the following commands;

srr-queue bandwidth share 1 30 35 5
priority-queue out
mls qos trust device cisco-phone
mls qos trust cos

I also see that in our AUTOQOS policy-map for our Cisco phones that the 'police...….' configuration line detail has not pasted across

I am wondering if these commands are handled differently in the 9200 and have to be reconfigured accordingly.

Expert advice on this would be greatly appreciated.
Good day. I have a Data and Voice VLAN, with the PC's getting DHCP from Microsoft DHCP Server (2012) and the phones from a Linux DHCP. I want to decommission the Linux DHCP and move everything to Windows DHCP. Phones are Polycom Soundpoint IP 331, Server is 2012, Voice VLAN ID:2. Please assist to configure phones to get DHCP from Windows server.
In doing business networks, I've encountered quite a few VOIP installations.  
Many of them are provided by the ISP provider and run on the same office subnet as the computers.  Even these have troubles from time to time but are really quite simple.
Others have the VOIP set up with a separate internet connection solely for VOIP, provided by a separate VOIP provider, and are isolated from the office network with a VLAN.
In the latter case, responsibilities get blurred.
So, I'm wondering if there is a VOIP industry best practice?

Here's an example:  
One desk's phone often has a screeching noise "like grinding metal" in the audio that is audible on both ends.
The phone has been replaced a couple of times to no avail.
Other phone on the same switch end has no trouble.
I've started a Wireshark capture of the traffic.

Another example:
Calls are sometimes such that audio is only working in one direction.  So, only one participant can hear anything.

We know and understand the network.
The VOIP provider understands VOIP and their system.

What is common and best practice for dealing with things like this?
How much support and service should we expect from the VOIP provider?
Who leads in the investigation?

I'm not interested in finger-pointing, just trying to calibrate expectations.
I can well imagine that this is contract-dependent but this must be a common situation with a wealth of experiences.
I am using Freepbx 14 and working fine but I got thousands of attacks and in Intrusion Detection, my public ip  has been blocked sometimes and because of this calls are not working. I am using fortigate firewall and opened the 5060 to 20000 ports for the FreePBX so My question is 1. are ports forward mandatory for inbound route ( if I change the sip registration port from 5060 to other and do same with the trunk provider ) . Please let me know how I can make this FreePBX more secure so call disturbance would not occurred in future.
AWS, Speech Recognition and AWS

I have heard AWS has an excellent Speech Recognition web API, and wonder if anyone knows of a complete speech recognition app which uses this AWS API?

Apparently, the demo emails you the translation and I hear the translation is great.

Anybody got any leans on an app that can handle speech to text?

Good afternoon everyone.   I was wondering if anyone had any experience using the My Fax application.   Currently one of our customer is using the service to send outbound faxes directly from their EMR application.   The majority of the fax numbers associated with the EMR application they send to are coming through without a problem.   There is one fax number (which I believe to be an analog line) when sending , the EMR application shows the fax sends and completes the send however only partial pages are coming through.   This is only happening with 1 fax line out of the rest.   According to the vendor they did state this.

"Good afternoon. Pls note that this change reflects ALL faxes sent. In our experience, this raises a few additional issues:
1) Additional Support Calls in to our team as to why the failed confirmations are delayed (ie could take up to 30 minutes to produce a failed confirmation)
2) If the line we are dialing is of poor quality and cuts off multiple times mid transmission, we will still try multiple times to deliver that fax meaning that they would receive partial faxes…."

Where specifically can I look to find out where the break is happening?   Is there a way to verify if the line quality is poor?  Or if not, and the fax is making it to the destination could the drop happen on the way they are processing the inbound transmission on their end?

Here is some additional information from the vendor..

I have investigated and tested the number …
Hi All,

We have an older Mitel 3300 using predominantly Mitel 5212 phones. We have a dedicated VLAN for voice traffic, which is tagged on our switches alongside whichever VLAN is providing the network for the computers. We are having a strange problem in which when a phone call is made, one side is unable to hear the other (in reverse it is fine). After a few seconds this will sometimes resolve itself, or they will need to hang up and try again. On the retry both sides can usually have full conversation as normal. This happens randomly throughout the day to different number / handsets that are plugged into various switches throughout the organisation. We have had a look at the switches and they don't have any obvious errors / are not reporting they have maxed out on throughput.

We have had our support engineers look at the phone system and they cannot spot anything obvious with regards to the fault.

Any clues as to where to start diagnosing this.

Dear Experts,

We have two Fanvil C600 phones in the office.

One is working well, the other is just showing a blank screen with the led indicator "red".

I have done firmware upgrade (pressing back and home screen button) and reboot the phone but still showing black screen.

I am not sure of wiping out the config as i have not done any backup.

Connected phone to network, the wan ports says "1000 mbps" but I am not able to ping the static ip address

The phone displayed an IP, but I cannot ping it.

Connected to lan port of phone but there is no connection.

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).