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CDelapena2
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Calling from analog phone to VoIP phone?

Thu Mar 14, 2013 1:02 am

Hello,

Is it possible to connect an analog phone to an FXS port on a CME router and a VoIP phone to a switch connected to said router and have voice connectivity between the phones? Also, is it possible to connect an FXO port on that same CME to a RJ-11 wall jack to connect to the PSTN and be able to call that VoIP phone as well as the analog phone from my cellphone?

Is any of this possible or am I missing something? These are just loose ends in my mind I'm trying to tie as I read the CCNA Voice OCG.

Thanks
Yo, —embriagado de mis penas,— me devoro,
Y mis miserias lloro,
Y buitre de mí mismo me levanto,
Y me hiero y me curo con mi canto
Buitre a la vez que altivo Prometeo.—

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gbarnas
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Re: Calling from analog phone to VoIP phone?

Thu Mar 14, 2013 5:56 am

Short answer: yes and sort-of.

IP to POTS is supported; I have parts of a cheap speakerphone in a box at the front door, configured for a PLAR to operate as a door-phone, and have a POTS phone/fax as well.

Calling in from POTS will ring all designated phones - you can't call the analog or IP phone directly from outside unless you have a DID trunk.

Glenn
Lost: Rocket, appx 8' tall, Green w/ yellow fins, nose, and computer bay.. Last seen streaking skyward in NY

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CDelapena2
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Re: Calling from analog phone to VoIP phone?

Thu Mar 14, 2013 7:35 am

gbarnas wrote:Short answer: yes and sort-of.

IP to POTS is supported; I have parts of a cheap speakerphone in a box at the front door, configured for a PLAR to operate as a door-phone, and have a POTS phone/fax as well.

Calling in from POTS will ring all designated phones - you can't call the analog or IP phone directly from outside unless you have a DID trunk.

Glenn

Thank you for your input.

How would I go about getting a DID trunk? Is that something I'd have to pay for/negotiate with a PSTN provider?
Yo, —embriagado de mis penas,— me devoro,
Y mis miserias lloro,
Y buitre de mí mismo me levanto,
Y me hiero y me curo con mi canto
Buitre a la vez que altivo Prometeo.—

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gbarnas
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Re: Calling from analog phone to VoIP phone?

Thu Mar 14, 2013 7:57 am

A DID trunk (typically a T1 or PRI) is not cheap, and are usually sold in blocks of 10, 20, or 100 numbers. With a DID, the FXO receives the Called number And the Calling number. The Called number allows the PBX/VoIP router to route the call to a specific station set, the Calling number presents Caller-ID. Most larger businesses use this - it allows each employee to have a unique public phone number without paying for a phone line per employee. A trunk with 100 DIDs might only be capable of carrying 23 calls, but this might be sufficient based on average call load.

You could accomplish this fairly inexpensively with a SIP trunk. By design, SIP provides the called number and calling number.. Many SIP providers offer DID blocks, or you could simply register two individual DID lines or DID numbers. At home, I have a single trunk with two DID numbers registered, one for business and one for personal. Calls are $1.30 per month for each phone number plus $0.009 per minute out and $0.012 per minute in. With the two DIDs, e911 service on one of them, and my typical calls, it costs me less than $10 per month. I currently use FlowRoute, but have also tested others including VoipVoip.com. I selected flowroute because the monthly cost of a registered number was lower, allowing me to have two for less fixed cost. Both cost about the same per minute and have comparable quality. I do like VoipVoip because they allow calls between customers at no cost. This is nice for small businesses with remote offices and work-at-home employees. There are many providers with various costs and features. I'm about to publish a How-To on my web site related to setting this up in CME for a small business or home lab.

Glenn
Lost: Rocket, appx 8' tall, Green w/ yellow fins, nose, and computer bay.. Last seen streaking skyward in NY

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CDelapena2
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Re: Calling from analog phone to VoIP phone?

Sat Mar 16, 2013 2:15 am

gbarnas wrote:A DID trunk (typically a T1 or PRI) is not cheap, and are usually sold in blocks of 10, 20, or 100 numbers. With a DID, the FXO receives the Called number And the Calling number. The Called number allows the PBX/VoIP router to route the call to a specific station set, the Calling number presents Caller-ID. Most larger businesses use this - it allows each employee to have a unique public phone number without paying for a phone line per employee. A trunk with 100 DIDs might only be capable of carrying 23 calls, but this might be sufficient based on average call load.

You could accomplish this fairly inexpensively with a SIP trunk. By design, SIP provides the called number and calling number.. Many SIP providers offer DID blocks, or you could simply register two individual DID lines or DID numbers. At home, I have a single trunk with two DID numbers registered, one for business and one for personal. Calls are $1.30 per month for each phone number plus $0.009 per minute out and $0.012 per minute in. With the two DIDs, e911 service on one of them, and my typical calls, it costs me less than $10 per month. I currently use FlowRoute, but have also tested others including VoipVoip.com. I selected flowroute because the monthly cost of a registered number was lower, allowing me to have two for less fixed cost. Both cost about the same per minute and have comparable quality. I do like VoipVoip because they allow calls between customers at no cost. This is nice for small businesses with remote offices and work-at-home employees. There are many providers with various costs and features. I'm about to publish a How-To on my web site related to setting this up in CME for a small business or home lab.

Glenn

So then what's the difference between a DID trunk and a SIP trunk if SIP also provides DID? Is it that SIP allows you to get 2 DIDs instead of having to go by tens for a DID trunk? I guess I'll get to all this eventually in my CCNA Voice book. Thanks a ton for your input. I'd be more than happy to read that guide whenever you get it up. Thanks again.
Yo, —embriagado de mis penas,— me devoro,
Y mis miserias lloro,
Y buitre de mí mismo me levanto,
Y me hiero y me curo con mi canto
Buitre a la vez que altivo Prometeo.—

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gbarnas
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Re: Calling from analog phone to VoIP phone?

Sun Mar 17, 2013 8:43 am

Well, logically, there isn't much difference between a DID trunk and a SIP trunk. Physically, they are quite different - one being a copper (or fiber) line from your local Telco that has the capacity to handle multiple concurrent calls AND the technology to deliver both the caller and calling numbers while the other is an IP network interface. Most commonly, the DID trunk is delivered by T1 or ISDN PRI service. You can usually get fractional service, where only a few (usually 4 or more) channels are turned up of the 23 that are available. Still, it's big bucks for a home or lab situation.

The SIP trunk can also deliver multiple call sessions, limited by your Internet bandwidth and not an arbitrary number of channels in a T1 or PRI. Some vendors may artificially limit the number of concurrent sessions, however. The configuration within a Cisco CME is also different in how the lines are terminated, but actual routing of inbound calls is quite similar. The called number is routed to a specific station or set of stations.

For you to route incoming calls on multiple numbers to specific station sets, the SIP trunk method will be your least expensive choice, both in terms of equipment and ongoing service cost.

You can get the document off of this link to my web site: www.innotechcg.com. The document outlines the configuration I built to configure a SIP trunk in my lab, including the ideas and reasons behind the choices. I also review a couple of alternative configurations that I tried.

This document is an initial draft, so I'd greatly appreciate any comments, suggestions, and feedback.

As for the DIDs, my provider (flowroute.com) sells DIDs one at a time or in sequential blocks of 10, 20, or 50. I was able to purchase two separate DIDs, one for the business and one for my personal use and associate both with the single SIP trunk.

Glenn
Lost: Rocket, appx 8' tall, Green w/ yellow fins, nose, and computer bay.. Last seen streaking skyward in NY

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Re: Calling from analog phone to VoIP phone?

Thu Nov 05, 2015 1:35 am

VoIP is the communication that discover the best calling from analog phone OnetvoIP.com is the leading voice over IP services provider which enables organizations to reduce the complexity and cost of international telecommunications.

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Re: Calling from analog phone to VoIP phone?

Thu Nov 05, 2015 1:36 am

OneTvoIp is the leading voice over IP services provider which enables organizations to reduce the complexity and cost of international telecommunications.

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Re: Calling from analog phone to VoIP phone?

Sat Mar 05, 2016 1:17 am

A VoIP Adapter, also known as an Analog Telephone Adapter (ATA) will convert a VoIP signal to an analog tone so that you can use existing analog devices such as your phone or fax machine with VoIP service. This allows you all the awesome cost-savings of VoIP without having to replace your existing analog equipment. VoIP Supply ATA's will work with most SIP service providers and some Obihai VoIP Adapters even work with Google Voice.
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gymprathap
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Re: Calling from analog phone to VoIP phone?

Sun Jun 26, 2016 11:48 pm

A VoIP adapter is a device that converts analog voice signals into digital IP packets for transport over an IP network. A VoIP adapter also converts digital IP packets in analog voice streams.

Standard VoIP adapters connect to analog telephones via an FXS port. The VoIP adapter then connects to your Local Area Network (LAN) via an Ethernet cable and an RJ45 port. Some VoIP adapters also feature an FXO port so you can connect to the PSTN.

FXS and FXO Ports

FXS and FXO ports are important to know and not confuse.

An FXS port is an interface that connects station devices such as your phones or PBX to a VoIP adapter. An FXO port is an interface that connects your POTS line to a VoIP adapter.

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