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.

Share tech news, updates, or what's on your mind.

Sign up to Post

Hello, I have absolutely no experience with Avaya or phone systems but i am being asked to step in and configure a few settings on an older system.

Basically, there is an Avaya IP Office 3.0 system in place managing two different offices (and two different primary numbers) in the same physical location.  One of those offices is now out of business and needs their calls to route to an external number.  There is a PRI circuit including ALL phone numbers for both offices on the same account.

Is there a way to route call intended for one number to an external phone number within the Avaya system.

I am having some difficulty locating instructions as i believe this is an old system, over 15 years now I hear.

Any advise would be greatly appreciated.  Thank you

I have an Elastix server with Grandstream 1165 IP phones. I have disabled the waiting calls. When a busy extension is called, the caller is receiving a failed tone (the display it shows "failed" too ) instead of busy tone (the display it must shows "busy" also)

Anyone knows the problem?

Best regards!
Hi Guys,
I am trying to change firmware in a Cisco CP7911G.
I dont know what version SCCP was installed in this 7911. I think I followed a wrong tutorial. I started using full factory reset (123456789*0#). I have a TFTP server and I tried many versions downloaded from Cisco Support site.
Whatever version I try, It loads term06.defaults ... and follow to the next file (jar11...).
Many tutorials tells that I will need to upload 8-5-2 firmware... I tried it too.
The last tutorial that I followed is this:

Feb 12 16:39:32 transvoip11 xinetd[30119]: START: tftp pid=28660 from=
Feb 12 16:39:32 transvoip11 in.tftpd[28662]: RRQ from filename term06.default.loads
Feb 12 16:39:40 transvoip11 in.tftpd[28783]: RRQ from filename jar11sip.9-4-2ES9.sbn
Feb 12 16:41:58 transvoip11 in.tftpd[30839]: RRQ from filename /spa2102.cfg
Feb 12 16:41:58 transvoip11 in.tftpd[30839]: sending NAK (1, File not found) to
Feb 12 16:42:45 transvoip11 in.tftpd[31555]: RRQ from filename term06.default.loads
Feb 12 16:42:52 transvoip11 in.tftpd[31555]: tftpd: read(ack): Connection refused

In this last log, I am retrying to upload 9-4-2 SIP.
I already tried 8.0.2 SCCP, 8-3-2 SCCP, 8-5-2 SCCP. 9-4-2 SIP.

I tried to Hard reset factory... (*3491672850#)... no sucess.

Any tip? Am I forgotting something?

Hi Guys

I used to Avaya commander sotware on my pc, but i lost it due to a reformat. How can i get it again?
I have done  this a number of times:  Bring the current server freepbx up to the version on the new box, then run fpbx backup, migrate, restore to the new server - done.  But this time is different.  I have Freepbx 12 running with Asterisk 1.8 on a PIAF build a few years old.  I built a new box with a Freepbx distro, Freepbx 13 and Asterisk 11.  When I try to run the freepbx "12 to 13" upgrade tool on the old box, it refuses because I don't have Asterisk 11.  But I'm sure that if I take a backup from fpbx 12 and try to load it into fpbx 13 it will either refuse or things will get flaky.  It seems the only option is to try to manually install Asterisk 11 on the old box just to get the upgrade tool to run.  So I tried  that - and the resulting Asterisk threw segmentation faults every few seconds.  And even though 1.8 was  no longer running, Freepbx still insisted I was running 1.8 and refused the upgrade.  There has got to be a cleaner way to accomplish this although googling has not helped.  What's the right way?
I have multiple lines in my firewall configuration that have our IP NAT’ing configuration in them.
When we go to our DR site and do a restore we will have a different range of external IP’s such as 66.###.###.#01-128
nat (inside,outside) static 1##.###.###.#01 dns
nat (inside,outside) static 1##.###.###.#02 dns
nat (inside,outside) static 1##.###.###.#03 dns

Is there a way of me doing a find and replace for multiple IP’sw with Multiple IP’s?
Maybe with a spreadsheet or something by the way I see as my example above that they may use the same ending numbers and just do a find and replace with the beginning numbers but that isn’t the case.
it would be 60 numbers between 129-255
and 66-128 or something along those lines, wouldn’t know until I got to the DR site.
A fellow employee always leaves a "message" in such a way that:
1- my phone MSG button blinks
2- my LCD reads "YOU HAVE 1 MESSAGE"
3- when I pick up phone and push MSG button to hear message, it auto dials the person who sent the "message"

I don't want pushing MSG to call this person back automatically

What is going on? Is the employee leaving the message in such a way that this happens?
I am setting up a help desk ticketing system and we would like to be able to pass the incoming phone number into the system.  We are using Cisco Call Manager( I can get the version if It is needed).  What I need is a program that will run on the help desk technicians computer that will pass the incoming phone number into our ticketing system.  It does this by running a command line "program.exe 999-999-9999" it then opens up a windows that has the information for the incoming call.  I have talked with our phone department and all they will say is I will have to write a program to do it.  No guidance on where to look or what commands I have to do.  This seems like something that people would want to do so I would think there must be something written already that will accomplish this.  Can somebody point me in a direction to pursue?


        I am using VOIP over the vpn. Everything is working fine but recently, I have configured a another VPN with same LAN  IP  so I have configured vertual IP and stable the vpn . But after configuring vpn with vertual IP ,I am not able to listen voice ,SIP phone is able to register with remote voip server, call are ringing but i am not able to listening the voice. I am using fortigate 200D firewall.
Hello Expert

how can someone register sip phone without using extension number as username in asterisk?
I am setting up Microsoft Skype for business 2015 with Sonus Gateway SBC 1000

 is there a way to register cisco voice conference device , 7937G  with Sonus SBC 1000 Gateway
 it doesn't support SIP  but is there any way to make it work with SCCP protocol?

  is 7937G  compatible with Cisco UCM release 10 or 11 or 9  or even 7 or 8
It seemed obvious that this problem is related to SNTP issues.

No changes have been made to switches/routers/firewall.

I verified on the phones affected that SNTP is set to (the ShoreTel server)

I rebooted the ShoreTel Win2008R2 server, no luck.  Then stopped and started W32Time service.  No change.

I stopped W32Time and installed http://www.timeutilites.com/atcs.php Absolute Time Server service and selected the setting for SNTP serving.  

No change on user's phone.  
I pulled another 560 phone out of the box which I purchased refurbished on eBay. February 5.  No luck.
After booting a 560 I went into set up and changed the phone's SNTP server to (time-c.nist.gov).  No luck.
I just found an old 530. Plugged it in and worked great with SNTP  Correct time.
I found at 230. Plugged it in and did not work. February 5.

Note that I'm not changing anything on our server between plugging all those various phones in.  All plugging in to same cable/switch port in the server room.

Any ideas are welcome. I need to step away from this for a while!
Hello Team.

I have asterisk PBX and configured everything well but phone doesn't register to certain network.

I can ping the phone from the pbx and when i shift to another network it works fine.

what could be the problem?
I hope you guys can give a hand with this scenario:

I have a PBX running on my LAN with a SIP Trunk provided thru a private circuit (ONT) by my ISP.

The remote extensions/phones support OpenVPN, so they reach my internal PBX via the OpenVPN server running on PFSense. The purpose of this OpenVPN tunnel is exclusively for SIP/RTP traffic of external phones, nothing else.

I would like to use Traffic Shaper to allocate 20% of my bandwidth to the OpenVPN/SIP/RTP traffic from my external phones to the PBX.

We are aware the tunnel traffic is encrypted, and as far as I know the Traffic Shapper would not be able to prioritize the traffic of the SIP/RTP protocols.

What options do I have to accomplish this task?

Thank you
When calling a Response Group, the work flow is followed.  But all you get is the hold music and the call is not put though to an agent.  There are agents in the group.  They are available so it can not be a presence issue.  All calls are then abandoned.  This is happening on all our Response Groups.  

I have googled the issue but have not found anything relevant.  I was wondering whether this was a known issue with Lync.

I have a site with a UC560 (UC560-T1E1-K9) running "uc500-advipservicesk9-mz.151-2.T4".  We use CME 8.6 and CCA 3.2

The client wants to know if there’s a way a user can control the options (caller-id display name, allow/deny phone numbers), etc. Is there a software that will allow them to do this? Office Manager? Enabling the GUI?

Also, Caller ID is being shown for external calls coming on site.  They display the number.  Is there a way to add a name to this number so users can see both displayed on their phones?

My apologize for the basic questions but I have not worked on Cisco Voip systems.  This is 100% out of my wheelhouse.
Trying to understand why sip calls drop at ~11 mins. The sip call log to 0756094959 is attached as an example. Any suggestions is greatly appreciated.
Hi all,

Have a customer that has purchased Vonage for a few staff members, and I am trying to get the Cisco SP504G phones working on the Sonicwall-protected LAN. The phones will provision but then go to straight amber lights (fail to register / communicate).

Found this -http://support.vonagebusiness.com/app/answers/detail/a_id/1017/~/sonicwall-firewall-configuration

Did as described within, it made no difference. 

Have tried everything that I can think of.

- Tested a phone in the DMZ, no go

- Port forwarded 5061:5013 UDP, 10000:20000 UDP to a phone, nadda

- Temporarioy enabled Allow All / Any from Any inbound & outbound

Nothing seems to work, but the phone seem to operate OK out side the LAN (on other external networks, offsite).

Anyone seen this before?
Problem - Lync Phone Edition Error: Can't download certificate because domain is not available (I think this is the message).

Recently upgraded my central system from Lync 2010 to S4B, and am in the process of upgrading my branch office SBAs (AudioCodes Mediant 800s) from Lync 2010 to Lync 2013. All seems to go well until the user tries to sign in after I've finished the update.

Configuring a PIN for the user seems to work (using PIN Authentication on the Lync Phone), client is then prompted that their phone is USB, signs out and back in using the USB method. If we try the USB authentication method from the start it doesn't work. However then PIN authentication seems to work.

Any Ideas?
I recently setup FreePBX server running on CentOS.  I have two 1120e phones hooked up right now.  Ext 101 and 102.  The server and phones are on the same network.  When I call 102 from 101, i get a DECLINE.  When I call 102 to 101 I get SERVICE UNAVAILABLE.  Any debugging that I can do to help.  I had it working at one point.
I restarted a FreePBX project for my school to get phones in about 12 classrooms that only communicate internally only.  The server is all setup and was working at one point, but now I'm getting a "Declined" when calling ext 101 to 102.  From Ext 102 to 101, I get a "SERVICE UNAVAILABLE"

Any debugging tips that I can use to see what is blocking the call.  I ran into this before, but it was related to one of the protocol settings in the extension properties.  It's not passing through any firewalls and connected all on the same network.
Hi Experts,

I need help with my server.
I have 2 issues:
1- my dhcp server doesn't start automatically when I reboot my freepbx
2- my sntp doesn't work

for issue number 2 here is what's happening:
I have a freepbx Distro and everything works on my phones besides the time and date (it keep flashing).
I changed the configuration on my tftp configuration file of a phone and changed the sntp to point to a different server , then I rebooted the phone and the time and date worked. when I change it back to point to my freepbx it doesn't work anymore.
I need help on configuring it or to check if something wrong on the config file of the sntp.

I installed webmin to help me configure my dhcp . for the sntp I didn't configure anything I assumed it is installed and working by default,but I guess I am wrong.

can someone help me with my 2 issues?

Hi, I inherited a Mitel 3300 system that can host both digital and VOIP phones. There is a need to add some phones on the system and I'm not sure which way to go. I know VOIP is easier to support but I don't see any other benefits in order to justify the price. Basically VOIP phones add more expenses with the headset, license and the bandwidth consumed compared to a much cheaper digital phone. I know that the main advantage of VOIP is going through a SIP trunk rather than the traditional PRI but it's not the case for us and I'm worried that by the time we can move on to SIP, the phone system and the phones will be obsolete. So, as of now the price of a VOIP phone would be close to $600 + Internet, while a digital phone costs $150 tops.

Any insight into this would be greatly appreciated.

I have a weird situation with one of my phone extensions. For some reason, when you dial extension 422, it doesn't dial it out. It just sits without any system prompts or error messages as if it's waiting for me to add more numbers to it or something. Not sure where to start troubleshooting this to be honest. Never experienced this before. It works when you call it from an outside line (although it takes a while to route the call than usual) but it doesn't work internally.
Hello, I have a client that wants his auto attendant to ring during business hours directly to a blast group and during non business hours have the auto attendant kick in. Any Advice to get this config done. Its a Cisco 2921 router with CUE 8.0.6 installed.

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.