fxo port connected to pstn

[sarahr202]

Hi everybody.

Let say we have a phone number assigned by our pstn as : 555-458-6523.

We bought a voice gateway with two fxs ports and one fxo port.

analog phone1( 20001)-------fxs-Voice-gateway voice port 0/1/0 (fx0)------------PSTN.

                                                             | (fxs)

                                                             |

                                                    analogphone2(20002)

We want our analog phones to call any number on PSTN in the world.

The question is how do we configure our dial peer on voice gateway to acheive that goal?

thanks and have a great weekend.

[Ayodeji Okanlawon · Accepted solution]

Sarah,

Its as simple as configuring a pots dial-peer..

dial-peer voice 10 pots

destination-pattern 9T

port 0/1/0

The 9T assumes you use 9 as an access code for the PSTN.

Have a look at this excellent document to understand dial-peers and call legs

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_ovrvw.html

You can also have pots dial-peer that are specific to avoid inter digit time out

eg..for UK mobile calls

dial-peer voice 11 pots

destination-pattern 907[5-9]........

port 0/1/0

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"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[sarahr202]

Thanks Aokanlawon.

analog1---fx0-----voice-gateway fx0------pstn,

When a user at analogphone1, picks up the handset, and dial 9 to dial any number on pstn, user will get the dial tone. Who is providing this 2nd dial tone, pstn or voice-gateway ?

[Ayodeji Okanlawon · Accepted solution]

The gateway provides the secondary dial tone, not the PSTN.

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

[sarahr202]

Thanks Aokanlawon

analogphon1--------fxs  --voice-gateway---fx0-------DTMF-----pstn

our analogphone1 is old and uses pulse dialing while pstn understands only dtmf.

Suppose analogphone1 dials a number on pstn, what will voice gateway do? Will voice gateway convert that signals to dtmf so pstn can understand it? or we must replace our analogphone1 by different analogphone that uses dtmf ?

thanks and have a great weekend.

[Harmit Singh · Accepted solution]

Hi Sarah,

If you wish to use pulse dialing for that analog phone, then you must instruct the FXO port to use pulse dialing as well. The default is DTMF for FXO and E&M ports.If you configure the voice port to support pulse dialing, the different  voice port signaling types handle pulse in the following way:

•FXO—Transmits but does not receive pulse dialing

•FXS—Does not transmit, but receives pulse dialing

•E&M—Transmits and receives pulse dialing

Here is how to change to pulse dialing for the FXO port:

router(config-voiceport)# 
dial-type {pulse | dtmf}

However, please remember, this applies to all calls going out this FXO port. So if you have some other analog phones or IP phones which are using dtmf method, then this FXO port may not understand it since it is set to pulse. My advice is - if it is just one old analog phone which does pulse dialing only, change it to a newer one which supports dtmf.

HTH.

Regards,

Harmit.

[sarahr202]

Thanks Harmit.

Just want to confirm the following:

FXS:  can only receive dtmf and pulse dialing. But it can not transmit dtmf or pulse dialing

FXO  can receive and transmit dtmf.  But it can only transmit pulse dialing.

is my understanding correct ?

------------------------------------------------------------------------------

router(config-voiceport)# dial-type {pulse | dtmf}

However,  please remember, this applies to all calls going out this FXO port. So  if you have some other analog phones or IP phones which are using dtmf  method, then this FXO port may not understand it since it is set to  pulse. My advice is - if it is just one old analog phone which does  pulse dialing only, change it to a newer one which supports dtmf.

Please correct me if I am wrong

You are saying since our fxo port is configured for pulse dialing, therfore it can only transmit pulse dialing. It will not understand if it receives a dtmf dialing on its fxo from ip phone.

Did I understand you right ?

=====================================================================================

Does Ipphone use dtmf signalling?

I remembered a while ago I performed a lab as decribed below:

ipphone1------CME-router--------ipphone2

I dialed ipphone2's extension using ipphone1 and performed a packet capture between Ipphone1 and cme.

I noticed that all the dialed digits were carried in sccp messages " button pressed". I don't see any dtmf signaling

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Assuming ipphone uses dtmf signaling , I can not think of any example where our fxo port receives a dtmf signaling from ippphone. Could you please give me an example?

Because my understanding is fxo connects to pbx or pstn.

Thanks and have a great day.

[Harmit Singh · Accepted solution]

Hi Sarah,

I'll start answering your questions top down:

1>     The following statements only apply to pulse dialing and not DTMF:

FXO—Transmits but does not receive pulse dialing

FXS—Does not transmit, but receives pulse dialing

2>     What I was trying to say was that both parties (FXS and FXO or IP Phone and FXO) should either do pulse dialing or both should do DTMF, else you may see that you get an unbreakable dialtone.

3>     Regarding IP Phone and DTMF dialing, there are several methods of transporting DTMF between endpoints. In general terms, these methods can be classified as out-of-band (OOB) and in-band signaling. In-band DTMF transport methods send either raw or signaled DTMF tones within the RTP stream, and they need to be handled and interpreted by the endpoints that generate and/or receive them. Out-of-band (OOB) signaling methods transport DTMF tones outside of the RTP path, either directly to and from the endpoints or via a call agent such as Cisco Unified CM, which interprets and/or forwards these tones as required. In your case, since you were using a skinny IP Phone, it had to report the digits entered to the UCM via SCCP protocol.

4>     Sorry about the confusion, maybe I didnt frame what I wanted to say very clearly before :-) IP phone would talk to the call agent (CUCM in our case) and FXO would either be controlled by the call agent (in case MGCP protocol is used) or by the gateway (in case H323 or SIP trunk is used between gateway and CUCM). Hence, IP phone would not send DTMF directly to the FXO.

IP Phone (SCCP/SIP) --- CUCM --- (MGCP/H323/SIP) --- GW (FXO) --- PSTN/PBX

HTH.

Regards,

Harmit.

[sarahr202]

Thanks Harmit.

I am going to spend next few days reading my book. See you guys soon.

[Harmit Singh]

Anytime Sarah. See you around :-)

Regards,

Harmit.