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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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I will have a chance of interview for the subject job position.

Can you share with me what I should look up and prepare before the interview?
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Just want to get folks opinions.  Anyone using them?  Any feedback?
Trying to implement sparkboards in every new office and eliminate things like conference phones, polycoms, and all that legacy stuff.

Thanks.
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Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?
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Hi,

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
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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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SIPp is a free SIP traffic generating tool for Linux.
http://sipp.sourceforge.net/

SIPp user manual says you can install SIPp under CYGWIN on windows. However I am not experienced  with compiling applications to run under Linux and need help getting SIPp up and running under CYGWIN on a windows10 machine.

I have successfully installed CGYWIN and included the following packages (all successfully)
gcc-core
gcc-g++
gcc
libncurses
make

After the CGYWIN install, I put C:/cygwin64/bin in the win10 systems’ environment variable PATH – so far all ok and CGYWIN seems to be working fine.

In addition, the SIPp install instructions state:

Warning
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.


QUESTION 1 -  Do you know what this is???    IPv6 extension for CYGWIN http://win6.jp/Cygwin/ 
is it a CYGWIN package, and entire install version??
What/how do I need to do to check/install?

QUESTION 2 – Nothing happens when I try to run “autoreconf -ivf” ...but this might have to do with Question 1 not being addressed yet.

 /cygdrive/c/Backup/tools/SIPp/3.3
$ autoreconf -ivf
-bash: autoreconf: command not found


+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
FROM DOC


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…
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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?
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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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Hello Experts,

I am in need of options to connect an analog phone to an old, Nortel PBX that is in another building with no physical, wired connection between the two buildings.  There IS network connectivity between the two buildings, and I've been told by our phone support \ vendor that there are devices available that would connect to the analog port on the PBX on one side, and to the phone itself on the other side, and then both devices would connect to the LAN to provide connectivity between the PBX and the phone.  Our vendor, however, has very little info on these.  I am writing to see if anyone here has any familiarity with this concept, and if so, can provide any recommendations or guidance on makes \ models they have used successfully in the past?

Thanks in advance for any help that can be offered,

Russ
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We have ISDN PRI line 30 channel terminated into Digium TE133 card. Sometimes when we make outbound call on any mobile number we got error "All circuits are busy now". Once we redial the same number the call connects.

 How to capture  debugs  ?
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Eye-catchers on the conference table
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Eye-catchers on the conference table

Challenge: The i-unit group was not satisfied with the audio quality during remote meetings. They were looking for a portable solution with excellent audio quality for use in their conference room but also at their client’s offices.

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 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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i have Asterisk server running on two servers as Active/Active with 1000 Clients.Now we have configured the Lync Server i need step by step Guidelines for moving the clients from Asterisk to the Microsoft Lync with same extensions and dial plan.
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I am in need to check the best solution for IP Telephony to work with Microsoft Lync Solution.any idea with logical comments will be really appreciated

already running ASterisk but want to replace it.
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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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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,

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?
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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
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On my mac I use iMessage to send text messages.

What is there on Windows to do that same thing?

Thanks.
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Hello all,

I have some Win 2012 3cx v15 phone systems and was having trouble with apple push notifications for calls to remote devices.  I've determined it to be a TLS issue.  I had used IIS Crypto to remove the less secure SSL 3.0, TLS 1.0 and 1.1, leaving just TLS 1.2 and more secure ciphers.  This breaks apple push notifications from the 3cx server/software.  I put back TLS 1.1, no luck.  Put back TLS 1.0, now push notifications work.  I find it odd that I should still need 1.0 enabled on the server.  

Is apple push still using that protocol and not 1.1 or 1.2, or might there be something else going on here.

I'm by no means familiar with protocols/ciphers, just determined what fixes the problem.
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I need for the replacement to look just like these?  Notice that they do not have the Plantronics branding on them.plantronics
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I have CUCM 10.5 and we use 9+1+(areacode) + number to dial out.  We just introduced Jabber and would like to leverage the dialing from Outlook as well as urls.  

The issue is that it is only the (AreaCode) + Number and it needs the 9+1 in the prefix.
I found something about using \+.! and the PreDot 900  https://supportforums.cisco.com/discussion/11950616/jabber-click-dial-outlook-prefix-9.

I am not familiar with Translation Patterns.  Can someone explain this a bit more.
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Hi All Expert,

Good Day.

I would like to check if there is anyway on how to check whether the company is using which phone PBX system?

Thanks!
1
I have the above phone trying to VPN with a Dell SonicWall TZ400. When I put in the VPN information, listed below, the phone fails and gives me error codes that Phase 2 no response. I will list the three error codes I also see, if anyone can point me in the right direction.

SonicWALL

SonicWall VPN Settings:

Policy Type: Tunnel Interface
Authentication Method: IKE using Preshared Secret

IPsec Primary Gateway Name or Address: 0.0.0.0

IKE Authentication:

Local IKE ID: Domain Name
Peer IKE ID: Domain Name

IKE (Phase 1) Proposal:

Exchange: Aggressive Mod
DH Group: 2
Encryption: 3DES
Authentication: SHA1
Life Time: 28800

IPsec (Phase 2) Proposal:

Protocol: ESp
Encryption: 3DES
Authentication: SHA1
Enable Perfect Forward Secrecy: Checked
DH Group: 2
Life time: 28800

In advanced tab, the only thing checked is Keep Alive.

PHONE

Server: 50.XX.XX.209
IKE ID: VPNPhone
PSK: *****
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1
VPN Start Mode: Boot

Password Type: N/A
Encapsulation: RFC
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1

Copy TOS: No
File Srvr: Blank
QTest: Disable
Connectivity Check: Never

Errors

1/3
IKE Phase1 received notify
Error Code: 3997698:18
Module: NOTIFY:305

2/3
IKE Phase2 no response
Error code: 397700:0
Module: IKMPD:353

3/3
IKE Phase2 no response
Error code: 3997700:0
Module: IKECFG:1184
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.