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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Is there a converter or way to utilize a very nice slightly used Panasonic phone system that does not have VOIP technology built in to the phone system. It's perfect for a small business. Bought it a few years ago as the VOIPs were coming in. Wish I had not now. Unfortunately I have a cell phone but would like to utilize the home phone system NOT VOIP again because its conveniently setup to two different work stations. So I was wondering if the phone system I have can still be utilized with a phone line though some kind of a converter or am I forced to purchase all new phone system that works with VOIP? Because I am under contract with AT&T cellular. I unfortunately cannot switch and AT&T does not have landline services in my area so that means I have to get all new services with another provider? It just mean redundancy I just do not want to put everything through my cell phone. when I have a super nice phone system in house. Your thoughts appreciated.
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Hi,

We have a Skype account for our company.

We sign in with our company email address to access our Skype account on our Windows 10 Pro PC.

For people to get in contact with us, they usually request a Skype ID.

Our Skype ID displays as something like the following: live:" "_(numbers)

We would like to change this as it does not look professional.

How do we change the Skype Name/ID for our Skype account?

We are aware we can change the display name for the account but the Skype ID is needed for people to search for our company on Skype.

Thanks,
Robbie
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I have a DHCP server with two IP scopes.  We also have VOIP phones that use a separate VLan and receives their IP addresses via a DHCP server on my router.  We had some problems with the VOIP Vlan and was forced to put all the phones onto the default DHCP server with my computer devices.  Once we fixed the VLAN issue, we changed the configuration in the phone which put all the phones back on their own VLAN, thus releasing all the IP's leases from my default DHCP server.  This morning, I got a warning that my DHCP server was using 88% of the IP's, looked at the server and all the phones leases were back.  I know these phones are not using these IP's because the change is manual.  Each phone had to be put back onto their VLan through a manual setup.

My lease time is 6 hours, so I deleted all the phone devices that showed up, but does anyone know why after a week, they showed up and started pulling IPs?
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+18777682265

when I call Santander bank using skype
"welcome to skytel system"

but I can call using a cell phone

Please explain why you think I am blocked
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We have a router in front of SBC that blocks ICMP Packets, how can we check the ping delays to the SBC from customers' SBC?
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Hi,
 
 My customer has been using TextBox (a service of Line1 Communications, Inc. https://www.businesstextbox.com/) to send text messages directly from their computers in the past.
 Since they switched their phone system to VoIP system, TextBox failed to run. The tech support of TextBox software company told me that the VoIP phone carrier is the problem.
 TextBox software utilizes main business phone number to send/receive texts from recipients cell phones.
 Having said that, is there a software out there to replace this TextBox software?
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Hi,
As I am new to Cisco UC, I have the following design question which I would like to ask on what is required for it to work. There is an existing Cisco Call manager in HQ at Country A.

a. There is a requirement to setup a branch office in country B.
B. There is a secure vpn connection from the branch office in country B to country A
C. When call are initiated from the ip phones in country B locally, it should be called locally and not via country A call manager to avoid overseas call charges
D. When call are initiated from the ip phones in Country B for country A; it should be via country A local call tolls.
E. When call are between the ip phones, it will go via the secure vpn connection.

As I am new to Cisco UC, I would like to ask, is CME required for the router at the branch office or UC license would be sufficient? May I also ask on how this is done/design?

Any suggestions is greatly appreciated
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We have a DHCP server with multiple scopes (VLANs). We have option configured for VoIP phones too. It seems to be straight forward to configure as per MS, but anything can happen while we try to configure!. Any of you had any problem of making to work?.  Do we have to do anything on the network equipments to make this work?
I am looking for EE/KB article . I would appreciate if you could help me on this.
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I am using Freepbx 13 and want to block outbound calls on 911 no only.
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Hello All,

We had a MPLS connectivity between our HQ and branch,  we have recently migrated to IPSec VPN and i have enabled all ports between the HQ Soretel server and the branch Office. We can do a ping and access the Shoretel Branch swithc but Swith fails to Pair up as its sending RST packets back to HQ server.

Will SHoretel works fine in IPsec ? do i need to do anything to avoid this issue.

Shorele guys says there is a network issue Port 5452 seems Filtered intermittently. I have checked the config with Cisco Support, they verified and confirmed ports open and nothing found in packet capture.

can any one suggest over my case ? below are the Logs

tmsncc log shows hq cannot communicate to this switch

00:00:12.321 ( 5072: 6528) cco_cmd: clnt_call error: status= 3 (RPC_CANTSEND)
00:00:12.321 ( 5072: 6528) ncc_connect_setup: --> -20 (RPC_ERROR)
00:00:12.321 ( 5072: 6528) ncc_connect_to_switch: "10.104.6.7", 5452
00:00:12.327 ( 5072: 6528) 9, (19.47.5900.0) "10.104.6.7", "00-10-49-3D-2D-5B", 25(SG4-30) Flash, "en-US", (1.1.3.27),"3 d + 08:12:45",0
00:00:12.327 ( 5072: 6528) ncc_event_connect (2018/06/11 07:00:12.327, +7)
00:00:22.328 ( 5072: 6528) readtcp wfmo timeout
00:00:22.328 ( 5072: 6528) sw_cmd: clnt_call error: status= 5 (RPC_TIMEDOUT)
00:00:22.328 ( 5072: 6528) nec_event_connect_ex: --> -20 (RPC_ERROR)
00:00:26.314 ( 5072: 6528) ncc_connect_to_switch: "10.104.6.7", 5452

since the ip address changed to a different network, this …
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Dear Experts,

I need recommendations of an Open Source telephone system Solution.

I need to let go of an old Cisco Unity Express. There is no budget for that office.

Is there a reliable product out there that I can set up in a Server machine?

Thank you
0
Hi,
As I am new to Cisco UC and I have a scenario whereby there is a requirement to have cisco ip phone in a branch. The IP phones would be connected back to HQ call manager.  I would like to check on the advantage and disadvantage on the edge router whereby on using dual WAN connectivity links for high availability against a single WAN connectivity link and SRST.

For SRST, It seems that there is a regular SRST and using CME as SRST. May I check if there may be any reference documents on how it works/requirements and consideration.

Thanks.

Appreciate for any suggestion!
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Hello,

I would like to know if it is possible to effectively use voip with private vlan edges, and how.

I have private vlan edges configured, essentially with the switchport protected, switchport block unicast and switchport block multicast, on all my user's workstation ports on the distribution switches. This is to prevent lateral movement in case of compromise. I would like to configure the ports for VoIP in the usual chained jack-to-phone-to-computer format. These catalyst switches are connected to the core catalyst switch via fiber.

I understand that all traffic on a switchport protected interface will be sent to the uplink and that this includes all voice and data traffic from that particular interface. But, I would prefer not to have to disable protected ports to allow phone to phone voice traffic.

Please help.
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We are installing a number ATA's either Cisco SPA's or Grandstream ATA's at a number of different locations. Each location is on a private network, however we do have static WAN IP's for the Firewalls. Each ATA has ethernet Available, but not Copper 2 pair.  Our goal is to pay a VOIP provider for about 5 VOIP Lines, that would be shared by 25 ATA's. Most of the traffic will be outbound dialing, and we would want the ATA's to rollover to the first available outbound line. We do ideally, want to be able to inbound dial to the ATA's using extensions.

Where would we get started in finding a VOIP provider that can help us with this, or would this be best done with on some sort of On Premise Gateway, which we would host at our NOC?

All comments and suggestions are appreciated.
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The current Network setup is customer site connected one SIP trunk each in US and Europe respectively
over MPLS Network. The customer is asking for cross region resiliency in SIP Trunks, is it possible? I'm not sure
if inter continent trunking will cause any issues? Please provide pros and cons.
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Hi
Were experiencing issues with the Shoretel VoIP breaking up.
This is occurring predominately at one of our remote sites, however the main site is being affect, if slightly less.
Internal calls from the remote site to head office along with external calls are frequently causing problems.

Each site has 100/100mb link,
Shoretel switch at both sites.
Director and E1k and Ingate at main site,

Diffserve 467 enabled on HP POE switches
dedicated vlan for voice in place across the sites
sites connected by site to site VPN

Ideas?

Thanks
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I had this question after viewing Microsoft access database make skype call.

Skype4Com API can only be used with the classic Skype version. The new Microsoft Skype for Desktop (the 8.X version) is not compatible with this API.

 

You should also be aware that several functions included in Skype4Com API will now not work even with the classic Skype. You will not find any new documentation of this API. The last available Help file is from 2011, so it's entirely up to you to check if applications based on Skype4Com API are still working or not.

Is there a solution that will allow 'doze 10 users to make VoIP calls from an access database?  I am developing a portable contact solution and I need it to be portable and work from remote locations.
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Hello,

Are cloud phones better than an on premise solution?
We currently have a Siemens 4K phone system used with a PRI.
We have about 230 voice over IP phones and about 15 digital line phones.
Most of the ip phones have a basic setup with add on module to configure additional lines.

The 15 digital lines are for our sales department, the have a ACD routing setup(similar to a hunt group). Each sales person has their prime line and 2 secondary lines. We have it configured so that that each sales person is able to answered each others lines. Each sales person has about 50 lines configured on two phones at their desk that they can answer when any line rings. We it setup this way because our president wants sales to try and answer all calls and not have them go to voicemail.

We recently started to look at cloud phones and providers like ring central, 101voice and others. Our main concern is QOS and if our setup in sales will be supported.

We are not sure if cloud phones will be more reliable and if we will have the same quality of service?

Thank You


Let me know if you need any clarification.
0
VOIP Assuming that one T1 line can take 42 calls  and per call is 38k.
I have a site with 62 users using 2 T1 lines, how do you calculate if the lines are enough for the site?
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Having issues with VoIP, we have Vonage and there's some buzzing going on when people make calls, the current environment we have is:

ISP to >>> ASA 5505 >>>> 3560E to the distribution switch for phones 2960X (the phone type is Yealink)


Any ideas?

Thanks so much for your help.
0
We are trying to locate serial number of Cisco UCS, we have access to Cisco unified communications manager, we have lost password for Cisco Enterprise License Manager ( CLI doesnt work, because passwords dont work).

Is there anyways to find out the serial number of the device through CUCM?
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I have a Panasonic tda50 system and tva50 voicemail.

1. If I’m on a call on line 1 and a second call comes in on line 2, the stations alerts me of the incoming call but doesn’t show the caller id, I constantly have to look on the display of a phone that’s not in use.

2. If the auto attended answers a call, the call does not get logged in my call log button for the icd group - how do I fix.

3. When I press hold the call is laced on hold but then I hear a dial tone - I have to either hang up or shut the speakers phone off - why doesn’t it just put the call on hold and go on hook?
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We are just starting to use Skype For Business for Internal IM.   We have setup Skype For Business to not allow External Contacts to be added. We didn't want Employees to start chatting with Friends and Family while at work.

We are looking to get rid off WebEx for Presentations and wanted to use Skype For Business, though it looks like we can't Invite Customers to the Meetings since we blocked External Access.

Is there a Way to Block External Users for IM, though Allow Inviting External Customers for Presentations?

Thanks!
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Look for outsource call center that they will have to use my five9 subscription to do in and out bounds calls.
https://www.five9.com/

Do u know any places that can provide those services?
1
Is there a Microsoft tool for testing Skype for Business video performance?  There exists for audio NetworkAssessmentTool.ext which allows you to make hundreds of simulated skype audio calls and gain insight into network preparedness for audio. Is there any similar tool that looks at Skype for Business online video?
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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).