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 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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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 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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Trying to set up internet for home use with the ability to fail over to a second internet connection (such as cellular). Since my phone and security system are both internet based, I just want to see if there is a (cheap) way to ensure device connections stay up if the primary internet goes down. Since the phone and security devices are not tied to a particular external IP, having them suddenly move to a different external IP is not a cause for concern.

I know best practice for business is to fail over to a different ISP, but since this is for home use I'll say that isn't required here.

I'll figure out which ISPs later. Right now just looking for hardware recommends and how it should be configured.
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Hello,

We have an On-prem shoretel system configured running the director version 18.xx.  We also have three shoretel switches and use both softphones and deskphones.

Shoretel Switches:
SG-T1k  
SG-220T1A
SG-90

Softphone:
Ran through Communicator on PC's

HardPhones:
Shoretel IP230

Edge Firewall:
Fortigate 100D

T1 Provider:
Level3

New WAN provider:
CenturyLink Fiber

# of Users:
20-30

Currently, our phones use a dedicated T1 connection through level3.  This T1 line connects directly to the SG-T1K.  Due to increasingly high costs, we are considering getting rid of the dedicated T1 line with level3 and routing our shoretel phones through our Primary WAN(century link fiber).  The fiber is connected to our Fortigate 100D edge router.  I have worked with shoretel for several years but I have not had to made a change like this before.  

My question is, can we accomplish this with our existing equipment(phones, switches, etc)?  If we can, how do i implement the changes.  If we can not, what needs to change or be upgraded to facilitate this change?

Thank you everyone in advance for any help that you may be able to offer.
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I need to move a call manager which is a VM on a UCS C200. Can you please advise
on the proper shutdown procedure and the turn-up? It will retain IP address etc.
Just need to make sure I don't corrupt anything. I see a shutdown procedure
below for CIMC and using the power button. But should I also ssh to Call Manager
first and shut there as well? Thank you.

http://www.cisco.com/c/en/us/td/docs/unified_computing/ucs/c/hw/C200M1/install/c200M1/replace.html#wp1053068
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We are using Cisco Unified Call Manager.

Let's say John wants to call Jane. Both are corporate users but John wants to call Jane from his mobile phone to her mobile phone.

I understand Cisco have a plug in that integrates with Click-to-call API's and allows the creation of an app that performs a call back functionality the voice system will call back John, call Jane, and then bridge the two calls.

Does anyone know the name of the plug in?
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Hello

I learning up about Skype Enterprise Voice and it's potential for mobile.

Right now, we use Skype for Skype-to-skype calls. I'm wondering if this can be integrated with the telephone line in the office.  And then, if we can deploy a Skype mobile client to user's personal devices that has their office number....this way, we can :

1. A user never has to give out their personal mobile number, only their office number which will now ring on their personal phone

2. A user can make voice calls, either internally or externally, using this Skype client and reduce cost to them since it presumably uses the corporate network where possible, e.g. for international calls.

Some questions.

1. Is Enterprise Voice the term for the integration of Skype with the phone network

2. When we refer to the Phone Network, do we mean PSTN?

3. Is there a way for an enterprise Skype mobile client to have an external dialler feature so the user can phone anyone, either internal or external?

4. Are there potential cost benefits of this Skype client connecting to the corporate network

5. Is it possible for Skype mobile to have the user's office desk number so that it provides the fixed mobile convergence i talked about? Or does it need a separate number?
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Plugging a Cisco 7941 phone in to an HP switch. Phone displays "Ethernet disconnected". Phone works fine when plugged into a Cisco switch. I've always had this issue with HP switches and just avoided plugging phones into them but this time its unavoidable. I've checked my Vlan settings and everything else I can think of. Does anyone have experience using HP switches with Cisco phone systems?
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  • Have a deployment of 3 servers using dns loadbalancing.
  • we have a trunk setup via an sbc.
  • users have deskphones (CX600)


both inbound and outbound calls work as expected.


however, when a call is placed on hold after 30 seconds the call drops.

the same thing occurs when a call is parked..also after 30 seconds the call drops.



i should mention MOH IS SETUP in this deployment, however when an external call is parked there is just a beep sound.

when a call is parked between users MOH does work and the call does not get dropped.

refer and bypass are set to false on the trunk as well.  Trunk settings



in the snooper log i see references to "this call leg has been replaced"  in the same message as the BYE:
ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"

the trace from the sbc shows that the mediation server is dropping the call so i haven't mentioned that here.

have the snooper trace if needed.


any suggestions appreciated.
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How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?
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Is there a portal for AT&T SIP trunk customers? We've asked our AT&T sales rep and Tech consultant over and over for this information but they can't seem to find it. I want to be able to forward DIDs which are part of our SIP trunks. So simple - yet so complex for the behemoth.
0
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I have a Zultys MX250 and today it kicked everyone out and said password was reset logoff and log back in.

this is strange.. i am and Admin and i cannot get in to the device .. help!
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Hello,

Can anyone point me in the right direction? There are a few google hits for this problem, so it seems like a known issue.

///////Summary: On outbound calls, we are sending out caller ID, and it is showing as "restricted" on HD enabled mobile phones only. Older mobile phones and land lines display our caller ID correctly. The telco PSTN providers have all pointed to our call manager as not providing the correct 1TU-T E.164 standard. This is occurring on all 5 MGCP gateway PRI's across multiple local PSTN providers.


///////System Parameters
We have 2 call managers in our HA cluster, 1 publisher, 1 subscriber.
Cisco Unified CM Administration

System version: 10.0.1.12900-2

VMware Installation: 2 vCPU Intel(R) Xeon(R) CPU E5-2609 0 @ 2.40GHz, disk 1: 80Gbytes, 4096Mbytes RAM, Partitions aligned


///////Troubleshooting Steps
Here is the support forum posting of the same issue, I have performed the changes advised in this posting with no success:
https://supportforums.cisco.com/discussion/12746556/debug-caller-id

1) Under Service Parameters - Clusterwide Parameters - Calling Party Number Screening Indicator

Set this value to Callmanager Provides Calling Number (No success)


2) I have performed the changed to Call routing information - Outbound calls, tested with Calling Party IE type as both "national/ISDN", Calling Numbering plan "ISDN". No success.

Here are our test call DNs appearing in the debug isdnq931:
May  4 11:44:02: ISDN …
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Hi

i have a Zultys Zip 57i phone that a user is having a problem with working from home.  It says no network available,

how do i get this phone to work on a home network ?
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Hi There,

A potential new customer asked me to do some technical work to some technical work for them as they're moving to the new office. Part of the scope of the project is to run data and voice cable. I'm fine with the data aspect of it since it's something i've done for years, the question is with the voice aspect of it.

They say they have a Hybrid PBX from Verizon where they have 9 land line and it gets split into 23 (lines?), i have not done any work with Verizon hybrid pbx but i managed another network where we had a hybrid pbx from fonality with 4 land lines and the server served 20 extensions,  all phones connections were regular data runs to the voice switch the the server connect to.

My question is this: based on what they told me (the Verizon hybrid pbx) the cabling for the phone system is simply regular "data" cabling of they need to be terminated to a 66 block?

P.S, Attached is a picturecloset picture showing a Nortel and Meridial box (i'm assuming the pbx) and tons of cables terminating on a 66 block.

I greatly appreciate your help.
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Hey Guys,

How do i backup the address book on a Cisco SPA525G phone?
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Hello,

I have installed a CUCM Publisher in a UCS server, I have another UCS server and I need to install a subscriber, What do I have to configure in a subscriber?
The licenses are shared between CUCM Publisher and subscriber?

thanks
0

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.