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,

I need some help with configuring the B-ACD with cisco cme 8.6 currently I dont have any pstn line all I have is 2801 router with CME 8.6 Installed with some phones up and running.Phones

 

Scenrio :
when ever some one call 600 It should go to IVR I did some basic configuration but it didnt worked

 

application
service aa flash:/bacd-3/app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param number-of-hunt-grps 2
param handoff-string aa
param dial-by-extension-option 3
paramspace english language en
param max-time-vm-retry 2
param aa-pilot 600
paramspace english location flash:/bacd-3/
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param max-time-call-retry 700
param service-name queue
!
service queue flash:/bacd-3/app-b-acd-3.0.0.2.tcl
param queue-len 15
param aa-hunt1 610
param queue-manager-debugs 1
param aa-hunt2 620
param number-of-hunt-grps 2

 

dial-peer voice 100 voip
service aa
destination-pattern 600
session target ipv4:192.168.100.1
incoming called-number 600
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad

 

telephony-service
max-ephones 7
max-dn 20
ip source-address 192.168.100.1 port 2000
system message "Welcome to Skaks Vocie LAB"
cnf-file location flash:
load 7911 SCCP11.8-5-3S.loads
load 7961 SCCP41.8-5-3S.loads
max-conferences 4 gain -6
moh flash:/music-on-hold.au
multicast moh 239.1.1.1 port 16384 route 192.168.100.1 192.168.20.1
web …
0
Skype for business 2019 Yealink T58A | Trusted Certificate
I need assistance with phones we recently purchased, all T58A handsets with Skype for business firmware.
 
I have tried all 3 latest available firmware and stuck with this version as it offered a simpler login screen for users.
 
Scenario:
Phones register correctly for as long as the trusted certificate is not present.
Periodically the handsets will populate with a CA certificate on line 1 even though everything is set to disabled below and then the users are unable to sign into the phones.


phone2.png

What is causing the phones to download the internal root domain CA certificate?
0
is there a standard way to set up a Residential Internet service for VOIP Phones?
the quality of the voice is poor and i was told there was a way to tweek or configure the home router to prioritize voice traffic
0
I have an NEC Univerge 8100 phone system which has just changed over to VOIP from ISDN via a SIP gateway.  The system is operational now however when dialling out, the phone system is not sending the handset caller ID and the receiving handset dispalys "No Caller ID".  I have quizzed my new carrier on this and they have advised that the phone system is only sending the phone number, but should be sending the area code as well.  See my response from the VOIP support.  

Hey Daniel,

Looking at the call logs i can see your phone system is sending only an 8 digit number without the area code.
this will need to be altered to send the full 10 digit number.

this is an example of what we are seeing

From: <sip:93050561@Metrofreight>;tag=fb1ca9b91a


Is there some way I can set the phone system to send the area code as well?  In example above, it should send 0393050561

The VOIP carrier wont take responsibility for the NEC system and unlikely my previous carrier will want to assist.  

I have admin access to the web interface.  

Thanks in advance, Dan.
0
GXW410X PSTN Gateway is not saving configurations after reboot...
0
Looking to see if anyone can assist or has thoughts on Microsoft Teams Voice using AudioCodes M800B to push our numbers coming from our CentryLink PRI to teams. The issue I am having is Degraded Audio on the recipients side, sporadically there will be points in a call where the recipient looses audio for a second or 2. CenturyLink is only a best effort network but our 500mb pipe isn't even hitting about half that and at max we maybe have 10 users on the phone at once. When I look at the calls in teams a large majority looks like they are good but there are about 5% of calls the come back with a 408 and 504 Response code.

I have made sure on my Meraki network I have set up all QOS and I have tried to trouble shoot with AudioCodes, Microsoft, and CenturyLink but so far no luck... my company has invested a large amount of time and money and I have about exhausted all my resources trying to track this down... any thoughts that may help me resolve this would be great;ly appreciated.
0
I have setup Enterprise Voice / Direct Routing for Microsoft's 'Phone System' element of Office 365 and Teams. We're using a Ribbon SBC Edge (1000) border controller which has a very useful wizard and it's own FQDN certificate from godaddy.

I've enabled enterprise voice in Office 365 online powershell and setup routing policies to allow outbound calling to +44 uk numbers.

In Direct Routing, the SBC is showing as active and in use. I can make calls TO MS Teams clients using DDIs on our SBC routing calls successfully to Teams. However, Teams clients cannot dial out to the PSTN.

We've run trace logging on the SBC which shows all the SIP traces - it can see all the 200 OPTIONS coming to and replying from Microsoft, but it never sees microsoft attempt an invite to a call. For some reason Teams is not reaching out to our SBC when we try to make a call.

Does anyone have any advice for troubleshooting in O365/Powershell to determine why the call doesn't reach our SBC?

Thanks
0
We have CUCM and all calls are showing up as the main number. We have configured that mask in the DN. When I do the call analyzer this is what I get,

Calling Party Transformations
External Phone Number Mask = YES
Calling Party Mask =
Prefix =
CallingLineId Presentation = Default
CallingName Presentation = Default
Calling Party Number = 7033

I checked the route pattern configuration and the mask box is checked, The route list detail use external mask is set to default.

 I am missing something but I do not know where. Any help would be appreciated.

Results Summary
Calling Party Information
Calling Party = 7033
Partition = ENT-EXT:Admin-SVC:Admin-xxx-Paging:Admin-PSTN:ENT-SVC:ENT-VM
Device CSS =
Line CSS = Admin-DEVICE
AAR Group Name =
AAR CSS =
Dialed Digits = 81xxx2
Match Result = RouteThisPattern
Matched Pattern Information
Called Party Number = 81xxx2
Time Zone = America/Chicago
End Device = xxx-TF-LD-RL
Call Classification = OffNet
InterDigit Timeout = NO
Device Override = Disabled
Outside Dial Tone = NO
Call Flow
Route Pattern :Pattern= 8.@
Positional Match List = 8:1:xxx:xx:xx2
DialPlan = North American Numbering Plan
Route Filter
Filter Name = LongDistance
Filter Clause = (AREA-CODE EXISTS AND LONG-DISTANCE-DIRECT-DIAL EXISTS AND OFFICE-CODE EXISTS AND SUBSCRIBER EXISTS)
Require Forced Authorization Code = No
Authorization Level = 0
Require Client Matter Code = No
Call Classification = OffNet
PreTransform Calling Party …
0
I am building the system and I am not sure sure in my architecture of the future system: The system enables people call to companies and organizations though web browser using WEB-RTC. Use-cases are following:


VOICE CALLS
FROM browser to HOME PHONE
FROM browser to mobile phone number
FROM browser to browser
FROM browser to existing corporate call-center (IP)
VIDEO CALLS

ONE TO ONE
ONE TO MANY
Question:

Since all calls are recorded, there are different options how calls might be implemented: peer to peer, through media server. Also, there might be more than 1 million concurrent calls. System parts should be free/open source. In my understanding, system design is following(attached). However, i am not sure in this. Please, help to build right architecture.

VOICE CALL
VOICE CALLS
VIDEO CALLS
VIDEO CALLS
0
I'm trying to connect (send command)  to an IP phone (Yealink) from outside the network.
It's already working from the inside with this simple URL to call:
"http://192.168.XX.XXX/servlet?key=number=5145555555"
It's very easy and can make the phone call a client directly from my custom app (Access database).

I see that IP phone company have remote access to my IP phone and would like to know the Technic to do the same.

I have users using the custom app on a remote server, but the IP phone is at another location, and I want to make there phone call by a click of a button the same way I do inside the network.
0
VOIP calls suffering from silent patches.

We had BT Cloud Voice put in around 6 months back and have suffered with issues since it was put in. We put this down to our FTTC lines, however we have now had a Virgin Leased Line installed (100mb/1gb).

We currently have around 90 computers and 90 phones. These are run through Netgear GS752TPv2 switches and to a Draytek 2926 router.

Looking on the router, the syslog is showing quite a lot of errors similar to the one below:

 [VoIP_QoS] VoIP RTP[45348] Rx Loss detected: 16/115 ...

There is also a few errors as below:
 [VoIP_QoS] VoIP RTP[45344] latency detected...220.

I'm assuming the first error is the receive packet loss (of around 15% in the case above) which could be causing the silent patches.

As well as this, I've setup a ping to the router from two different computers that are in different switches. They are both getting timeouts at the same time, one or twice a minute (sometimes there is a longer gap, sometimes there will be 2 within 10 seconds).

Any ideas where to start looking to try and resolve this issue?

Thanks, Shane
0
Hi,

We have some  voip software called unified communications UC. It suddenly stopped working for authentication back the the Service provider (but only doesnt work when on our LAN going via the ASA 9.0(2) . If hot spot by phone or any other method its fine.

Ive checked all ports to be open - in fact tried opening all outbound ports from my computer - same result. - If I do a wireshark ip.dt == to the Service provider SIP endpoint then try log in to the UC software I see totally nil  SIP traffic to that IP in fact nil sip traffic total. when i  run same wireshark when hotspotted to phone on same device I see the registration sip successful. ? all is fine

ive tried turning off and on sip inspection on ASA many times - doesnt fix. strangely however I did ONCE get a successful registration - and then never again!!! same failure to do anything even nothing in wireshark for sip. I cant see any denies on asa monitoring as well. ACL's all in place. Im a bit los from here service provider saying asa fault. please help
0
I have a VoIP phone with an OpenVPN client.   There is an issue when used on OpenVPN, so I want to inspect the data, SIP trace etc to help find the problem (fast then then the vendor)

My problem is the data I want to see is wrapped/encrypted with OpenVPN.   I have the private key and client certificates, therefore I thinking it must be possible to decrypt the capture and find the information I'm looking for?

How can I use Wireshark (or other opensource/free tool) to achieve this?

I have reasonable experience with Wireshark, used not used it to decrypt traffic before.


Many thanks


OpenVPN pcap
0
Looking for ways to limit users to only using Microsoft Teams for IM, voice and video calls.

We already use Slack and think that Slack does what Slack does better than Teams.

Unfortunately due to Skype for Business going end of life, we are being forced into Teams.

Some of the telecoms, video-conferencing and IM stuff is great in Teams but we prefer Slack for collaborative working.

We don't want to confuse users though, so are looking for ways to lock down the Teams specific stuff but still allow people to use it for the stuff it does well.

Has anyone got any advice about how to lock it down?

One other way we thought we could tackle it was to have notifications enabled for anyone that creates so that we can delete them / advise against.  Any thoughts about achieving that?


Jon
0
Experts,

I'm trying to make calls from my Outlook client. I in order to achieve this I needed to signup with a provider that provides SIP Trunking.

I've signed with a company called Dial 9 https://www.dial9.co.uk/

I'm now trying to set up my Outlook using the following link

http://www.voip-connections.com/contacts/how-to-dial-contacts-from-outlook/

Everything, so far, appears to be going in the right direction, however when I attempt to configure the TAPI settings it is greyed out .. see image

modemThis would suggest that I need to access the Phone and Modem setting with Administrative rights, but I'm not given the option to access using admin rights.

Can someone let me know where I'm going wrong?

Thanks
0
Hi,

I have a Samsung Note 9 phone.

Sometimes I would get a call from someone while being in a call already with another person.

The person trying to call me never knows that I was in a call with someone else so don't ring back.

To the person trying to ring me, their phone rings for a bit and then goes to voicemail.

I would like to have the person trying to call me in this situation given a message saying "person in a call" or busy.

Is there a setting for this on Note 9 phone?

Thanks,
Robbie
0
I am looking for a way for users/extensions to have the ability  to login/logout their particular hunt groups (on-demand) in CME 8.6. The main purpose is to have all calls forwarded to an answering service when all users/extensions are logged out the hunt group

Has anyone setup something like this?

Thanks!
0
Hi,

We have a Skype account for our company.

We sign in with our company email address to access our Skype account on our Windows 10 Pro PC.

For people to get in contact with us, they usually request a Skype ID.

Our Skype ID displays as something like the following: live:" "_(numbers)

Other companies have their Skype Name/ ID displayed without the "live:" part like "walmart" for example.

We would like to change our Skype ID/Name to something better as it does not look professional.

We are aware we can change the display name for the account.

How do we change the Skype Name/ID for our Skype account?

Thanks,
Robbie
0
I have Asterisk version 1.8.32.3. Sometimes my recordings have the audio of the agent and the audio of the client desyncronized. I want to know how could this be resolved.

I tried put jitterbuffer in sip.conf, and I changed to res_timing_dahdi.so

Best Regards.
0
Issue: Some SPA502G Cisco phones freeze without any warning,

Some users have found that their phone does not work and must restart it to recover it.
About 20 cases reported in the last two weeks (before this had not happened). We have almost 300 devices spa502g.
The trigger of this issue was not found, so the scenario cannot be reproduced.


SPA502G
Software version 7.5.6a
Hardware version 1.0.4

Platform
Freepbx 2.11.0.43
Asterisk 1.8.7.0

No recent updates have been made.
0
Hi,

Can anyone help me figure out the easiest way to configure an EdgeSwitch 24 250w for VoIP QoS?  There is surprising little clear/concise info out there on how to do so and the support I'm seeing for Ubiquiti products has me wishing I would have gone with a different brand.  /miniRant

I have an IPSec tunnel connecting two buildings, the 'remote' building has QoS configured on the Fortigate router, but the switch is basically in default mode.  I have 7 IP phones on site and we are having intermittent quality issues, so QoS on the switch is step one in my problem solving.  Browsing around the gui it looked like the OUI based method would be something I could fight through, but it's not quite that simple after all.

I'm not sure I understand what the OUI is...I though just the first half of the MAC.  The phones are Avaya model J169, and Google tells me the OUI is 00:04:0D, but according the client info from the switch, all of the phones MAC addy's start with C8:1F:EA, so wouldn't that be the OUI value?

Do I still need to create a VLAN with this method or does the 'auto-voip' setup take care of that for me?

Obviously a little over my head here with new stuff, but still disappointed that there isn't a 'how to' I could find...this has to be a very common request, no?
0
Recently I have migrated the 3cx on-premises to Cloud and all Ext. are connected through SBC. All inbound and outbound calls are working fine except the voicemail....I am not getting any voicemail after migration and in log I got Main line SBC:Unavailable  .......Internal voicemail are working fine.
0
Hi All,

We are looking for an analytics product to report on our usage of our on-premises Mitel VoIP system and Skype for Business Online. Can anyone recommend a product?
Reporting for Skype for Business Online is the priority, but we will likely need both in the future.
We are a medium sized charity and so our budget is limited.

Cheers, Ian
0
I am using 3cx to connect with twilio.

I know that 3cx normally uses Elastic SIP Trunking, but I need to use programmable voice.

I have everything working as I think it should except when making an outbound call.

If I use an app directly connected to twilio then it's fine, but when routing through 3cx via the app or desk phone I get an error.

Error - 32009 The user you tried to dial is not registered with the corresponding SIP Domain

The logs show that 3cx is trying to place the dialed number in sip:+1##########@name.sip.us1.twilio.com

When it should be sip:username@name.sip.us1.twilio.com in order for twilio to be happy.

I've tried changing settings in 3cx to use the AuthID when sending out and I think I even got it working for a moment, but when I tried to repeat the process I couldn't figure it out again.

So does anyone know how to use programmable voice with 3cx?
0
I am trying to setup a new Asterisk v13 machine to accept SIP connections from an oldr set of machines which are still running Asterisk v11.

When one of the older Asterisk 11 servers sends a call to V13 I get the following errors :-
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate

I have attached the relevant parts of the pjsip configuration. As I understand it the type=identity line should match the IP address and point the incoming SIP to the 'gateways' endpoint which is then configured to use the intergateway_auth authentication.
pjsip-config.txt
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).