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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.

Can sombody help me to understand the option feature in the dhcp pool configuration for VoIP (see sample below)?

ip dhcp pool Data
   network x.x.x.x
   default-router x.x.x.1
   dns-server x.x.x.x y.y.y.y.y
   option 242 ascii L2Q=1,L2QVLAN=5
   option 176 ascii L2Q=1,L2QVLAN=5
ip dhcp pool VoIP
   network x.x.x.x
   default-router x.x.x.1
   option 242 ascii MCIPADD=x.x.x.x,x,x,x,x,MCPORT=1719,HTTPSRVR=x.x.x.x,TFTPSRVR=x.x.x.x,VLANTEST=0
   option 176 ascii MCIPADD=x.x.x.x,x,x,x,x,MCPORT=1719,HTTPSRVR=x.x.x.x,TFTPSRVR=x.x.x.x,VLANTEST=0

Anyone has a sample VoIP packet Tracer lab? I'd like to see how the VoIP phone boot up process and how the whole thing is setup. Thanks
I have several NEC DT700 series ITL-8LD-1 VoIP phones and I have the following questions:
- what is the codec used by this model?
- Do I configure my Cisco switch with "mlqs os trust dscp" or "mls qos trust cos" on the interface?


I need to modify a script and add an option for when it is a statutory holiday to play our closed message even though a user is logged on.

What happened is that we had a few persons call in during a holiday and they got no closed message. I would like to prevent this for next time.

I have checked the script and can add or modify weekdays and weekends and time of day but did not find a way to add statutory holidays.

Can this be done? if so how.

Thanks in advance for your replies
I am 90% sure it was working and belive this is a director issue..  However:

Tested over a VPN and also locally on the LAN and get a line not in service error on the communicator.
Check HQ/Director and foudn that the SG90 @ HQ site is showing the softphone as out of service and is assigned to a MAC address of a device i cant trak (assyume a laptop):

D4-BE-D9-23-A1-EC Headquarters D4-BE-D9-23-A1-EC Unknown Out of Service (Operational)     Soft Phone


We are making a sip to sip call from one sip instrument to another sip instrument via a hosted server on a fixed IP address.

We find that intermittently the calls between the 2 soft phones will drop at about 19-20 seconds - on the other hand on some occasions the call will last its full duration without any problems.

We are using codec G.729.

Can you suggest what could be causing this issue & a method on how we can solve it?

I'm trying to get my router ready to install ciscu unified express.  I'm trying to follow some instructions at http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel8_0/install/installsre.html#wp1121849 but when i try to add an ip address to my ism0/0 interface i get the error message that the ip address may not be configured on L2 links.  I also don't see the commands unnumbered under the ism0/0 interface either.  The card i'm using is  ISM-SRE-300-K9
Hi, we are moving from a Nortel/Avaya CS1000 to IPO Ver. 8. I have about 15 users that will be getting 9640's with button modules. ( they currently use  Nortel m3904's with button modules) we have not been able to figure out how to set up the user programmable autodial buttons like we can do on the CS1000. These guys are on the phone all day and change the buttons pretty frequently. sometimes daily !

This would be a MAJOR headache if we cant do this

Thanks for your help
We currently have a Comdial FXII PBX System w/ about 90 digital phones & 2 PRI's.  We have a separate location that uses around 20 VoIP Phones we own that we pay a monthly fee to 8x8 for service.  I inquired with my PBX vendor, and our system does not support VoIP phones so we'd need to upgrade to a newer system.  I've seen some SIP to T1 gateways online, so my question is the following...

If we get another T1 card for the PBX, and add a SIP to T1 gateway (such as the G100 from Digium), would we be able to point the VoIP phones to the SIP gateway, and then have it in turn interface w/ the PBX so the users can make outbound calls using the existing PRI's?

Or would the gateway only be for a T1 connection to a carrier?
I am running Trixbox with a Cisco 7961 handset. Our upstream provider is about to restrict our outbound Caller ID to the incoming number ranges we have with them.

If I use the CFwdALL button on the 7961 handset to forward all calls to a mobile the Caller ID shown on the receiver's mobile is the originating caller's.

1. All calls to 0299222222 (handsets external number) are forwarded to 0411111111
2. External caller, 0433333333 makes a call to 0299222222.
3. Call is forwarded to 0411111111 and shows caller ID of 0433333333.

Once our upstream provider starts restricting our outbound caller id this call will be rejected by them as we don't own the number 0433333333.

Is there a way to force the forwarded call to use the forwarded extension's external number (0299222222) instead of the calling party's (0433333333) for the outbound Caller ID?
The title says it all, but I will try and expand although my knowledge of Voip is beginner at best.  

PBX is an Avaya IP500 with v9 software.  The Internet Telephony Provider is also the Internet Provider and the internet itself is a fibre leased line.  The connection to SIP is direct through the ethernet demarcation device which handles QoS.  There is no firewall or NATing.  I am porting a 200 DDI block of numbers from ISDN to SIP.  I have some sample numbers from the ISP to get the system up and running and tested before the porting date.  A few hiccups and learning curves later, I have calls working inbound and outbound with audio both ways at very high quality.  The last piece I need to get working is the Ringback Tone.  This is the tone the outside caller should hear before the phone is answered.  Currently there is silence. Help.

Is this a PBX or ISP configuration problem?  If it is ISP what should I instruct them to do to in order to get the tone?


If I have missed out some information, please ask me. Thank you in advance for your answers.
Hello.  Recently implemented a voip PBX and am having poor performance.  Have not implemented any QoS or VLans.
Have Comcast for out ISP with mid tier which is about 50/20.
Have plenty of switches at my disposal.  several Procurve 2924s and one 5406zl.  All support POE.
I am not an IT professional.  Just manage the best I can.  Have the ability to set Vlan tags and do most configuration changes if I know what to do.  The last time I tried Vlans, I found it cumbersome and problematice with DHCP, DNS and the few locations where the phone shares the same connection for both the phones and the PC.
My Question.
Given that we are a small shop with plenty of hardware.  Can anyone provide me some recommendations that do not require an in-depth study of SIP and networking.  Just want to phones to work.

My only other option is to segment the entire network physically and put the phones on a different subnet.  Do not like the idea of maintaining two DCHP servers.  DNS is not a big problem.  To manage the PBX have planned to configure two NIC.  one to manage it from the regualr network and the other with no gateway for the Voip network.

Any recommendations will be a big help.
I see this quote:

"digging for IP data (embedded image on a link) posted on some website.

This implied that data that can identify an IP address is embedded in images.
How does this work and how to avoid being identified.
Businesses and the like often have a need for anonymity.
I am researching to replace my present Panasonic 6.0 Plus for the home cum office, which has aged and is not working properly.   The idea is to be able to have my hands free to use the pc when answering the telephone call.  
I am looking for a telephone set where I could connect a headset, maybe like Plantronics, or possibly one that comes with a headset as an accessory.  
Can the experts please share their experiences they have with any of these.  Of course the budget is less than $50.  Thank u.
I have an internal customer who purchased a Cisco Linksys E1200 router to replace his old one and now his Cisco Proxy Phone can not regisister to the the Call Manager (via ASA).  
I suspect some default security setting is messing this up.  Anyone know what all needs to be permitted outbound for this to work?
I am trying to access Elastix using LINUX commands. I need to view the directories using the ls command. How do i do this? View screenshot.
Hello Experts,

We have Cisco Switch 3560 Series. We are going to deploy 300 phones.
My question Is it recommended to configure same port for voice and data vlan

Its Ericsson phone with 2 ports.

Thanks in advance.
POTS trunk (OBi110 device) returns "Answered" ruining a ringall strategy --FreePBX. I am trying to ring some local IP phones and 3 trunks in a ringall stratedgy. But as soon as a call goes out on the OBi110 it reports "answered" and the call gets connected. It isn't really "answered". The ringall stops. This happens even though I have specified "Confirm Calls" in the ring group.
We have an allworx 24x system running in our office. We just moved from a very expensive T1 connection with 5 digital lines to Comcast 100mb down/20mb up with 6 regular analog lines. How can I set up the analog lines and routing in allworx? i'm new to allworx and I only see 3 analog ports on allworx system. Sorry for dumb question, Comcast promised me one thing and I got a different connection from what I expect.
I am trying to set up a Call Handler that all forward-facing directory numbers will forward to.  So, no matter what number is called, they will be redirected to this Call Handler.  The purpose of the Call Handler is to inform callers that which days we are closed for the holidays, so it is not a permanent change.  Because of this, I want to do it as simply as possible with no major changes to the existing structure.

Here's what I have so far, a Call Handler with an alternate greeting on the extension 3199.  3199 is a DN that was created for this purpose and I can contact it from the phone at my desk (I hear the greeting that was recorded).  I created a Forwarded Routing Rule, set to Dialed Number equals 6* (I have also tried "In" 6000-6999 and "equals" 6934) and sent to the Call Handler/Go Directly to Greetings (status is active).  

When I try calling any number in that range, it does not forward the call to the Call Handler, it ignores it and rings on the phone as usual.  I have looked into CTI, but I do not know how to set it up to forward every DN in that range to 3199.  Is there something I'm missing?  Thanks.

I am trying to setup Microsoft Lync 2013 on Windows Server 2012 and am using the following document to guide my setup of this http://windowspbx.blogspot.no/2012/07/step-by-step-installing-lync-server.html.

I have verified I have internet connectivity and disabled IE Enhanced Security in Windows Server 2012 but when I get to the Choose a 'Certificate Authority (CA): Select a CA from the list' nothing appears in the list.

Any assistance will be much appreciated.

Thank you.
I have a client who has me looking into Hosted VOIP solutions.  They have four sites, about 15 phones total, and nothing too unusual in their requirements.

My concern has to do with getting internet service from an internet provider that is not the same company that provides the Hosted VOIP.  More specifically, the ISPs with whom I have spoken can't guarantee QoS all of the way to the Host site.

The Host companies tell me that it usually works just fine.  They will do tests from our site to theirs for a few days or longer to ensure that communication integrity is reasonable to expect.

The big concern that I have is that there is no assurance that there won't be a problem with traffic prioritization in the future.  The ISP will take the stand that they never guaranteed it in the first place and the Host vendor will say that there is nothing they can do about it.

I will ensure that all of the internal switches and routers and such will support QoS.

One solution for this is to have the ISP do the VOIP Hosting.  We are getting a quote for that, but I am expecting a substantial price difference.

Should I have these concerns?  Is anyone experiencing these problems?
We are using Elastix, and i need to add an announcement at an extension. The extension #400 belongs to the reception. The announcement should be configured at extension 400. How do i do this?
Can i also configure voicemail after the announcement?
I will tend a wedding in a few minutes and have been tasked with providing streaming video to the grandparents of the groom, and others. So I decided to use Skype.

I tested everything and it works fine. BUT how do I play back that video later? Is this possible?

I do not have a Skype group video license.

I have misplaced a cell phone in a room and I am looking for a free service to make internet phone calls.

This way I will be able to use my computer to call my cell phone number and I will be able to locate my cell phone when it rings.

Please provide me with a list of free services to make internet phone calls.

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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IP Telephony