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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I have been using a CIsco 3845 for trunking SIP traffic to my PSTN carrier. It's been working fine in this role and it has dual power supply which i love.  And it can pass quite a bit of traffic as I have it doing some routing as well. Am I missing out on anything by not upgrading to the CUBE?
I am using CUCM 7.1.5 and I have two H.323 voice-gateway routers which are Cisco 1760's. The first router has two VIC2-4FXO cards with 8 analog phone lines plugged into it, and the second router has one VIC2-4FXO card with 2 analog lines plugged into it, which gives me a total of 10 phone lines. We are growing out of 10 phone lines, so I just ordered a voice PRI (T1). I will be activating this circuit next week. I am going to install the PRI on a Cisco 2821 Router, and the 10 analog lines that are plugged into the 1760 routers will be ported over to the PRI, and I will then decommission the 1760's. My question is can I add this Cisco 2821 router to CUCM as a gateway now, even though there is no PRI connected to it at the moment because activation isn't until next week? I want to do everything I can ahead of time to minimize downtime during cut over day. Thanks

I have been told it is possible in Lync server 2013 to set up call forwarding / Simring, both server side and client side.

So that if user X is calls, it will also always call a number set server side. But still allow them to set up their own call forward through the client.

Any one know how this is done.

For example I want it so that if you call user A, it will also always ring the reception phone. However user A can still forward his number to another phone such as his mobile if he wishes, But calls will still be forward to reception as well.


Im looking at trying to get hold of some traffic stats off our hp switch that drives our 2x vlans for data and voice.

The console is pretty pants.  Any pointers?

I have nothing running in the background and yet when I run SKYPE, I get static, funny noises and the Skype message says that I need to quit any file sharing applications.

What can I do so as to improve my Skype calls.  
I'm using the free service.
Hello all,

I have looked at some of these posts, but none really address what I am trying to do.

Several sites on seperate subnets on default native vlan



Voice vlan ip as follows:



All default gateways are


We are putting in Shoretel system and are having trouble getting the vlans set up

All switches are Cisco 2950 and 2960 switches.

I have vlan 1 for the 192.168.x.x with subnet of
I have voice vlan for the 10.10.x.x with subnet of

The switches currently have the following commands on the ports:
switchport mode access
switchport voice vlan 5
no cdp enable
spanning-tree portfast

The port that will have the Shoregear switch plugged into it is configured as such:
description Uplink to Shoretel Switch
switchport trunk native vlan 5
no cdp enable
spanning-tree portfast

Uplink ports are:
switchport mode trunk

Now, the issue that I have at the moment is that most switches when issuing the no shut command on voice vlan 5, it will shut down vlan 1 (default native) and I lose remote access on the switch. What is wrong with the config? All ports need to be configured so that a PC and phone can be plugged into it. Also, the Shoretel server that will use both vlans as well.

Note...there will be no DHCP or auto config needed as the phones will be manually programmed.

We have a small number (10) grandstream phones that connect directly to sipgate VOIP provider.  These setup is a little tempramental but we have realised that a reboot of the phones fixes the vast majority of the problems.  They work for at least 24 hours without any intervention so are looking for a script that we can run to reboot the phones every 24 hours.

Is anyone aware of such a script?  At the moment we are connecting through HTTP to the phones IP, authenticating and selecting reboot from the configuration menu.
I will be activating a PRI in the coming days.  My fax number will be ported into the PRI.  How do i configure the FXS ports to allow for inbound and outbound fax?  I looked at many links from Cisco, but am still unsure.  Can someone send me an example of a configuration they have used that works?  I'm using CUCM 7.1.5, just an FYI.  I will attached my configurations.  

My call routing plan is as follows:

Any calls in the 610 range, 866 range and 888, range are the main numbers to the office which get routed to the auto attendant via 2001 (CTI Route Point)

My fax is 610-xxx-xxxx

All calls in the 484 range are for DID.
I posted a very similar question recently. But on further investigation I need to ask again. I need to track when and for how long various Cisco phones ring. I thought I found this in the Call Detail Reporting.  BUT it turns out that this information is only available for calls that were picked up. The problem I am analyzing is trying to figure out why the phone is NOT being picked up. So I need ringing information to correlate with data from our call distribution system. I looked also at the call traces. But that only give me the SIP code for "ringing" and not more specific information as to when the phone started ringing and when it stopped. Any other thought about how I could track this? One thought I had was that for an individual I could add their line to my phone and I would be able to see whenever it rang. Kind of cumbersome but perhaps better than nothing. Any other thoughts??
Hi All,

Hope everyone is well. I wonder if somebody could help me with this.

Its driving me a little mad to say the least.

We have a Cisco CUCME Box with a FXO Card in. If i plug a phone

direct into the wall and call the PSTN Number Caller ID shows fine.

If i plug into the Cisco FXO i dont get any caller ID.

Below is a copy of the Diag Capture:-

000334: Sep  2 16:47:10.899 AST: %SYS-5-CONFIG_I: Configured from

console by sysadmin on vty1 (
000335: Sep  2 16:47:31.339 AST: htsp_dsp_message: SEND_SIG_STATUS:

state=0x0 timestamp=52947 systime=328539
000336: Sep  2 16:47:31.343 AST: htsp_process_event: [0/1/0,

FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
000337: Sep  2 16:47:31.343 AST: htsp_timer - 125 msec
000338: Sep  2 16:47:31.471 AST: htsp_process_event: [0/1/0,

FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
000339: Sep  2 16:47:31.471 AST: htsp_timer - 10000 msec
000340: Sep  2 16:47:31.471 AST: htsp_timer3 - 5600 msec
000341: Sep  2 16:47:31.471 AST: [0/1/0] htsp_start_caller_id_rx:Mode

BELLCORE. Alerting 0x1
000342: Sep  2 16:47:31.471 AST: htsp_start_caller_id_rx create

000343: Sep  2 16:47:31.471 AST: [0/1/0] htsp_dsm_create_success  

returns 1
000344: Sep  2 16:47:32.479 AST: htsp_dsp_message: SEND_SIG_STATUS:

state=0x4 timestamp=54085 systime=328653
000345: Sep  2 16:47:32.479 AST: htsp_process_event: [0/1/0,

I need help as I'm comparing 2 vendors as we look at a Cisco Phone system and have the following

69 Analog Lines
11 Public Space Phones
336 Advanced Phones
17 Conference Phones

The phone models we are looking at are 7841, 8831 (Conf) and 8841

One vendor is quoting:
BE6K-UCL-BAS            11
BE6K-UCL-VM            288
BE6K-UCL-ENH            305
BE6K-UWL-STD            25

The other #2:
BE6K-UWL-STD            312
BE6K-UCL-ENH                        27
BE6K-UCL-ESS                            65

I am told Essential is Analog.  I am told ENH is for conference and Public at 28.  I am unclear what Basic is and VM and would like input on what anyone with experience with cisco phone licensing would say about either of these vendors based on this as I am not sure who is closer and even more confused now.
Hey everyone.  I am having some really weird issues with my Metro Ethernet sites and Shoretel.  Please note I am not the greatest at this as I have never had to deal with VoIP issues at my last job.  But here is the breakdown and what I have done.
Our company strictly runs HP ProCurve Routers/Switches and a few older 4500/4510g 3com switches.  We have 4 locations using Metro Ethernet 100mbps links and the phones (regardless of type) will randomly go into "Failover" mode and reboot.  The Shoretel tech said this was because they were losing connectivity to their ShoreGear T1k switches at our main location and our network was the cause.  I have ran Wireshark and captured VoIP traffic from our most troublesome site and could find no such occurrence of dropped packets.  Everyone conversation I analyzed had minimal jitter and seemed fine.  But then again I am not a wireshark professional.  I enabled SNMP and have been running Solarwinds Engineer's toolset using NPM, Interface Monitor and monitoring all the routers and switches in between the locations up to the Shoretel switch.  None of them show any dropped packets, errors or discards what so ever.  Their logs are all clear of any out of the norm errors.  The CPU stays low on all the units as well as the memory and the switches have more than enough buffer for the network traffic.  I checked to make sure the ports connecting the Shoretel switches were 100mpbs full as I had seen that could be an issue.  I also switched the HP …
Is there a way to link or integrate 2 separate phone systems? I have Shoretel phones with a Shoretel server in one office, and Cisco phones in the other. My goal is to make it so that any phone is reachable by just their extension, no matter which office is called...
Dear All,

 I have a core switch 4507 connected to all my fiber location's ( Loc 1, Loc 2, loc 3, Loc 4, etc). I have Cisco 3560 in my location's.  I need to implement QOS for MY lycn 2013 voice traffic between the loc 1 and the main core switch.

can someone explain to me what commands I need to implement in my global and interfaces configuration. in my loc 1 I have 2 Vlan ( Vlan 50 for Data and Vlan 55 for Voice ).


Cisco Iso version
Cisco IOS Software, Catalyst 4500 L3 Switch Software (cat4500-IPBASEK9-M),                                                                     on 12.2(54)SG, RELEASE SOFTWARE (fc3)

Cisco IOS Software, C3560E Software (C3560E-UNIVERSALK9-M), Version 12.2(55)SE5,

I have some customer service people who will say that they didn't answer a particular call because their phone never rang.  Is there a way to track when and if a particular Cisco 7945 phone rang?
Hi all,

We are currently using a VOIP Provider that has some great reporting and feature sets. The downside is the outages are never at a convenient time, like an outage is ever good. If we switch to another provider we will lose the reporting and pay more then double for the same features thru another provider. with the advancing telco market, i know there is a way to Route calls from a trunk to a provider.

Here is what i mean:

If we own 217-555-0123 and 217-555-1233 and 888-555-1233 as our main number.

If i use company FAILOVER for all 3 numbers. If a customer calls any of the numbers above they are routed to company MAIN that then rings our office. If for any reason company MAIN is not able to take calls company BACKUP will start to ring calls to our office.

Basically i want a redundant VOIP providers that if they are down for any reason the backup company can take calls.

i have the exe but can't find the msi...

Can anyone help?

Hello, looking for some feed back on best way of tackling/looking at a situation.  IP500 Avaya behind a WG xtm520 bovpn s2s to another loc(s) behind XTM505.  9620L VOIP phones at remotes(where ip500 is not). Users say will reboot and can see on display screen where phone is rebooting looking for IP. When on phone calls will drop and can see when rebooting.  No "jitter" is being reported as in "you're breaking up" "can't hear you" just phone rebooting throughout day.  I am thinking this is within the LAN as DHCP is local to the LAN but going back to the IP500 over tunnel at primary site, but I may be missing the best way to think about this/behavior.  

I've applied Qos on the voip policy and Traffic mgmt. as of last night on bovpn in and out policy preference five.  Wondering if I'm not looking at this issue correctly or what else I can try? Other input on troubleshooting.

No Avaya support but considering opening a case or looking for a local consultant but these things never have definitive answers, while I like to fix issues myself to understand and know it's done right.  Finally word has it AVAYA's official stance is that HW VPN tunnels aren't supported is their disclaimer while resellers do sell it b/c not all can have MPLS and MetroE?  

Hoping for some feedback/ideas.  

Thank You.
We have SIP VoIP connectivity between Kuwait and Dubai on Avaya Avaya IPO 500 which is Installed in Dubai to Kuwait which is running over Sonus Gateway through a VPN configured on FortiGate Firewall on both ends.

Everything seems to be working but all the sudden staff are able to dial in from Kuwait but the outgoing call from Dubai to Kuwait are not working.

I have the logs if someone can help me finding the problem it would be great.
Dear experts,

This is a fresh (and first) installation of freebpx on a virtual server, downloaded here http://schmoozecom.com/distro-download.php. It's the default installation and I haven't installed any "fancy" modules yet. The version of FreePBX is 5.211.65.

All outgoing calls work great. Randomly, we will not here our correspondent. On their side, they hear the phone ringing, but they don't know somebody has picked up the call. On our side, we hear absolutely nothing.

I'm new to this but I did my homework first. I opened all the recommended ports (http://www.freepbx.org/support/documentation/howtos/howto-resolving-audio-problems) but some of the instructions on that link are too advanced for me. The fact that the other person doesn't know that we took the call indicates (in my regards) that it's not really an audio problem (or codec), but something else.

Any pointers ? What could I test ?

Thank you for your help
Hi All,
     We are having issues with dropped calls form our Teleworker system, Below are the logs from the Mitel system
Mitel Error Log
Our support company are not overly helpful with this issue hence the post here. This morning i received confirmation that call's in and out were working but on the 2nd attempt not even a dial tone was heard.

Some time only 1 side can not hear, this will happen with both internal & External calls, UK and international Calls.

The Mitel system is running via a Watchguard unit in XTM510 in an active/Passive Cluster. If you could prompt with any questions you have as i know the above information is very sketchy at best....

Image looked poor quality so i may amend if required.
What do you recommend
that meets the below specs ?
  1. STYPE = cordless
  2. TYPE = Phone or Headset
  3. USE = warehouse
  4. RANGE = 10,000+ square foot
  5. EXTRAS = works with optional Range Extender
  6. INTEGRATES = with 3CX VOIP phone system
I already have been testing and have a local 3CX reseller, but want some more Windows 2012 server specs.

What recommends do you have for 32 simultaneous calls on a DELL server, without Hyper-V, that is only RUNNING your software ?
1.      MEMORY  better to have two 8GB RDIMM sticks or one 16GB RDIMM stick ?
2.      # of PROCESSORS  how many ?
3.      # of CORES per PROCESSOR  how many CORES per processor ?
4.      PROCESSOR SPEC  Is Intel® Xeon® E5-2403 v2 1.80GHz, 10M Cache, 6.4GT/s QPI, No Turbo, 4C, 80W, Max Mem 1333MHz OK ?
5.      RAID  OK with software RAID, or do you recommend hardware RAID ?
6.      HARD DRIVES  better to have two 7.2K RPM hard drives or one 15K RPM hard drive for VoiceMail saving/etc ?
7.      TAPE DRIVE  any recommendation for each backup/restore to another server if this server crashes ?  it is as easy as just restoring onto the same make/model machine or do I need to re-register/etc ?  basically if something happens to the PBX I need to recover in under 30 minutes
We have a mailbox-only ext covering a hunt group for busy or no answer.
We ring the hunt group 4 rings, then it goes to voice mail.

We would like the voice mail ext. not to ring 4 times before the greeting.  If no answer in 4 rings, transfer and immediately play the greeting.  So the user would get 4 rings not 8. (Yet give the hunt group 4 rings to grab the call.)
Not interested in using the call center or work groups for this. (no agents)
Can this be set on the class?
Thanks in advance.
trying to get outlook to dial out using the shoretel tapi drivers that i have installed and failing.
outlook accesses the shoretel and makes the call but its not sticking in the 9 to make an external call.

i have 9 in the phone and modem options but not working.


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).