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).

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

Sign up to Post

What ports do I need to open on my organization's firewall so that users outside the office will be able to communicate with the Lync 2013 server and with other Lync users both inside and outside of the office?

We have a Lync Edge server and a Lync front end server installed.

We will be using the Lync chat, computer to computer phone calls, and Lync video conferencing.

My organization uses just one firewall (a SonicWALL Pro 3060 with the Enhanced 4.2.1.7-17e firmware) and both the Lync Edge server and the Lync Front End server are Server 2012 Hyper-V virtual machines installed on the same Hyper-V base server.

Additionally both the Lync Edge server and the Lync front end server are on the same IP subnet.

Please tell me which ports need to be opened up on my organization's firewall and which protocols need to be forwarded onto the Lync edge and Lync front end server.
0
Can someone recommend a VOIP service that operates similarly to Google Voice?

I want to get a virtual phone number that can auto-forward all incoming calls to my cell phone. In addition I would like to be able to make outgoing calls (from my smart phone or computer) that will display that phone number in the caller ID of anyone receiving that outgoing call.

Also, it's critical that this number be able to receive text messages and forward those to my smartphone or email inbox. Will need to send text messages from my smart phone that display as coming from this virtual number.

I see there are companies like Grasshopper.com that offer this functionality, but they all seem to charge between $12 and $20 per month. Would prefer a cheaper alternative. Or free would be even better, but ultimately I realize I will probably have to pay something.

Thanks in advance.
0
Greetings,

I'm having an issue with configuring the V1910.  We have a lot of ProCurve 2500 and 2600 units that I have configured and have no issues.  These V1910 units are another beast.  Anyhow, for this device, I have VLANs of 1(default), 59, 202-204.  59 is voice, 202 is data, and 204 is management.  I have ports 23-24 tagged for 59, 202-204 and untagged for 1.  Ports 1-24 are untagged for 202 and tagged for 59.  At this point, on the ProCurves (25xx/26xx), I'd assign voice and qos to the 59 vlan, but the V1910 CLI is pathetic.  Anyhow, at that point, I would be able to connect a phone to any port (1-22) and it would get an IP for the 59 network (via DHCP).  I would then be able to connect a computer to that phone and get an IP for the 202 vlan (via DHCP).  That is not the case.  The only way that I get the phone to connect to the 59 vlan is by having the PVID for the port assigned 59.  The problem with that is that the computer behind the phone also then gets an address on the 59 vlan.  I've looked at the routing table.  I've done some messing around with the Voice VLAN and QoS settings on the web interface of the V1910, but no luck.  Frankly, I think this device is crap, but it could just be that I'm not a fan of web-only options for configuring a switch.  As such, I'm hoping that someone out there has successfully used the V1910 in this setting.

I appreciate any assistance.

Thanks,

Jeremy
0
I'm new to Asterisk and have a small Asterisk 11.x + FreePBX setup and am trying to diagnose random call quality issues. I've googled quite a bit and have found a number of tools, but haven't had much luck w/the ones I've tried (some of it due to my own inexperience I'm sure ...)  Can anyone recommend a good tool for assessing/monitoring QoS? Preferably one that doesn't require 6 mos of training to configure.

ANY assistance would be greatly appreciated.

TIA
0
I was reading about SIP and in the phones setup process, they have to go through the SIP proxy server. Now if I look at the Cisco Call Manager, does the SIP proxy server part of the Cisco CM? Thanks
0
We have AD integration setup on our shoretel director.


Numbers starting 019 are showing up fine in director under the Work and Fax columns.  All DID numbers under the DID column are showing correctly e.g. +4401912345678 or +4402012345678
Numbers to 020 or 07 under the work column and fax are showing as +2012345678 and +7912345678

Then in communicator you obviously cant make a call by clicking you need to manually dial the number!

Why, and how to fix?
0
I have a CUCM 8.6 setup at my main site.  We are moving one of my remote locations and right now we have a dedicated POTS line just for a fax machine which I was hoping to get rid of.  The remote site has a Cisco router which is a H323 gateway with a PRI going into it.  Anyone have any idea how I can port my POTS number into the PRI and get my physical fax machine to communicate with the router for faxing passthrough?  It looks like the Cisco ATA 187 might be used to do this?
0
I am trying to implement time of day routing. I have created the time periods and schedule. I created a partition with this after hours time schedule. I added it to the CSS at the top, so it looks at it the after hours time first. However, it still rings my hunt pilot instead of my translation which goes to my after hours number. They are both the same DN. Any ideas? Thanks.
0
looking for the ip addresses of reliable SNTP servers for time sync on our Avaya IP Office
system
currently using: 129.6.15.28
receive sporadic errors about no response from time server with above ip address
all help is greatly appreiciated
delebute
0
Hello Experts,
I am have serious issues trying to get Cisco Unified Call Manager v 9.0 (1) installed into a test lab and am looking for assistance.

I am using VMPlayer v10 (latest available as of this time), configured with 4gb memory, 180gb scsi hard drive, 1 processor, network bridged adapter, OS set as Red Hat Linux enterprise v5. I have also tried Red had v4 and then v6 as well.

I continue to get prompted several times for a logon after the installation. I enter my information three times and on the third time I get in as admin (so I know my credentials are correct). But, I also get "Permission Denied" errors at "/usr/local/platform/bin/cliscript.sh:" at various lines such as 135, 145, 17, and 18. I also get an IOException: hardware API:Cannot run program "/usr/local/bin/base_scripts/ihardware_iml": java.io.ioexception: error=13, Permission denied"

I've searched high and low on the Internet for some answers and am usually pretty successful with research, but am truly stuck this time. Any assistance on this is appreciated. I am very new to CUCM and am trying to get it going in a lab.

Thank you.
0
So far after looking online, I am thinking of buying Logitec HD Pro webcam (http://www.amazon.com/Logitech-Webcam-Widescreen-Calling-Recording/dp/B006JH8T3S/ref=sr_1_1?ie=UTF8&qid=1393619162&sr=8-1&keywords=logitec+webcam) for video, then Jabra SPEAK410 USB Speakerphone for Skype (http://www.amazon.com/Jabra-SPEAK410-Speakerphone-Skype-other/dp/B007SHJIO2/ref=pd_cp_pc_1) for voice.

What do you use for Skype conference call at your conference room?
And how do you mount the camera and use what mounting device for camera?
0
We have 30 cisco phones in use, 30 licenses in use with the Cisco phone manager, and 2 extra phones (already configured for the phone network).  When we try to deploy one of the extra phones, the phones fail registration with the server ... someone told us that is because of the licensing issue (too few licenses for the number of phones), which makes perfect sense.  

However, one of our phones has been giving us problems ... we would like to swap out the problem phone with one of the extras, but simply unplugging the problem phone doesn't allow the extra/replacement to register.  Is there some way to "clear" or "force" the problem phone off the server to free up a spot to add the extra in its place?

I realize the easiest solution is to simply purchase extra licenses, but you know how management can be sometimes - that's what we are dealing with here - but if we simply must buy the licenses, so be it :)
0
there is always the struggle of measuring for bandwidth needs and type:

guaranteed(4.5mb dedicated 600-$900) vs best effort(comcast business cable 100mb for approx $200 a month)

New small office with hw vpn tunnel back to main office hour away.

onsite AD replicating across tunnel.
VOIP phones back to IP phone system pbx at main office.
eight users (mostly sales)
conference room with tv and want to do some type of tel/Vid conference back to main office.
moving of 30/40mb art/pic files from and to main office.
of course will wan to prioritize voice traffic and vid traffic vs data traffic over tunnel.

enough/Thoughts?

Thx
0
I have an Vertical Xcelerator IP VOIP phone system behind a cisco RV220w router/firewall.  All toll free calls are being dropped after 16 minutes.  toll calls do not experience this behavior.  The trace shows that the carrier is sending a DPD re-invite but the PBX is not responding until 7 seconds later.  By this time the call is being torn down.  The cisco RV220W has the SIP ALG active but there are no settings for this.  Has anyone experienced this and is there a fix?(trace items below)

Dead peer detection INVITE
Feb 4 12:42:42 gw1 /usr/local/sbin/kamailio[9963]: INFO: [R-MAIN:101afc0c-a0002be-17ac-52f0dcbb-920dfef1-70bf3d82@10.0.2.190] -> Other sequential INVITE received from 74.80.226.164:5060

Reply from PBX
Feb 4 12:42:49 gw1 /usr/local/sbin/kamailio[9952]: ERROR: [R-DEFAULT-STATELESS-REPLY:101afc0c-a0002be-17ac-52f0dcbb-920dfef1-70bf3d82@10.0.2.190] !> Received stateless reply 200 (INVITE) from 108.32.57.50:6060
0
Hello Experts,

The owner of the company that I work at purchased another franchise two states away from where I am now.  They are running Verizon FIOS as their isp and we have comcast here as our isp, not sure if that matters or not... They will be running VOIP entirely for their phone system.  I am the only I.T. person here and what I need to know is how do I connect the VOIP phones at that location to our telephone system here?  What information do I need?  Basically how do I do this?
0
Customer has four companies using the same Cisco Communications Manager (CM) cluster (8.6(2)). Each company wants to mask their outbound DID block with the company name/number and not the specific user's name/number.  They have a system in place where they appear to be masking each individual line - they maintain a several hundred line spreadsheet mapping each Company A user to the same outgoing number, again with Company B and Company C. (See the attached pic, but expand logically to 400 users.)

Is there a simpler way of doing this?  Is there a way to translate every outgoing number for Company A to XXX, Company B to YYY and Company C to ZZZ with only three entries to the dial peer?

Thank you
Capture.JPG
0
I have a Polycom SoundPoint IP 335 voip phone.

It is hard-wired to my network and it works fine.

I was hoping to move the phone to a location that currently does not have any cabling.

I found that IOGEAR makes a product called -
Model - GWU627
http://www.iogear.com/product/GWU627/
Universal Wi-Fi N Adapter which is supposed to give any networked component wifi capability.
You simply plug a cable from the component's LAN port to the LAN port on the adapter and the component then is able to connect to your network via WIFI.

I bought this adapter and it works fine with a laptop.
I was able to use WPS to set it up and get it connected.

Then I moved it over to my voip phone and the phone is not able to establish a connection to the service provider (8x8 is the voip provider).

Do you think this is possible?
1
I just purchased and implemented a new CUCM with UCCX.  I moved away from Avaya IP Office.  The Avaya system was configured to pass the SMDR (CDR) call information to my SQL server.  My DBAs then used SQL Reporting to present detailed reports of in and outbound calling.

Cisco has a similar function using CDR records.  The problem is when using CTI Route points the inbound calls dialing specific DIDs are being translated into CSQs and the CSQ extensions are being logged and NOT the dialed DID.  This forced us to have the Gateway Router dump the call records to a file and deliver it via SFTP.  

So this is kinda messy but it is working.  My last piece missing is the Outbound calling.  It seems that UCCX has built in reporting that has this information.  How can I link or connect into the UCCX dastabase to collect this information?

I am a newbie to the world of Cisco Voice and CUCM.  I purchased this system because I was told I could get the reporting I needed and so far it has been a nightmare.

Please help!!
0
I am attempting to deploy CiscoJabberSetup.msi v9.6 via our Windows 2008 R2 Group Policy as part of a new CUCM installation. An initial manual install of the msi from a share proved successful.

I have built and assigned a package via a respective Group Policy object as per the Cisco deployment doc, and set the scope to include a security group (read + execute access) of test PC's.

I can browse to the package/msi at the software distribution point and install manually, but attempts through GPO fail. There are no indications in event viewer at all, no indications of file or folder creation on respective PC's. Running GPresult /scope computer /r on a respective indicates that the GPO is applied.

When running manually  I receive dialog prompts, so I'm wondering if the package needs to be modified to run 'quiet' (although Cisco docs do not indicate as such).

1) Has anyone deployed successfully/have any thoughts that might help?
2) Has anyone done so and created a transform using Orca to configure for presence server/log in etc?

I feel I am missing some key element here, so help would be appreciated.
0
Dear experts,

I cannot add a phone extension to a users phone in our CUCM 8.5.

She needs to have someone else's extension in her/him phone for her to pick it up.

I know is easy. Please help.

Thanks!
0
Dear All,

I have Lync 2013 Implemented in my environment, i need to configure the enterprise voice for users, we already have Cisco Call Manager in Place, what i want is to configure Lync to forward all calls happen from Lync client to Cisco, that's all, i don't need Lync to do anything here, how i can configure this??

please help.
0
Can the police trace a skype call or record it?
0
Hi,

I'm having issues login into CUE
System details:
UC-520
CUEv3.0
CCUME v4.2(0)

Basic error = "Cisco Unity Express has lost contact with the host router"

Attached screen shots may be more helpful

Screenshot1, I'm able to login to cue but it then takes me to screenshot2 where it states that cue has lost contact with the host router" I tried the same user/pass word I used in screenshot1 but no luck

I tried these steps with no success.
Enabling the HTTP Server
Enabling GUI Access for the System Administrator

Any help will be greatly appreciated
screenshot1.png
screenshot2.png
0
I have a digital Nortel CS60 switch.  I have been using TM3 (CS1000) for billing purposes.  Does anyone have any opinions as to what would be the best replacement for this dated software?  Would a current version simply be the best move, or is there something else out there that is better?  My TM3 build is going to need a reinstall anyway, so now would be a good time for me to change.
0
What VOIP providers do you recommend ?
  ** http://www.vitelity.com/services_voip, etc ?
0

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).