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DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

2 edits

DaSneaky1D

MVM

SOLVED - SIP Trunk to Analog phone on FXS port using CME

For the life of me, I can't seem to get what I thought to be a simple concept to work...

I have a SIP trunk with an ephone right now. That works perfectly inbound/outbound. I've added a FXS card and DSP to my 3825. When I plug in my analog phone, I can make outbound calls and I can call between extension. However, I cannot find a way to simply allow an inbound call to reach this phone.

I've look into the sccp/stcapp config (which you apparently need an analog gateway to make properly work...) and tried a creating a new dial-peer to route inbound calls from my DID to the voice port, but it never worked. Any help or suggestion would be greatly appreciated!

voice-port is 0/0/0

rsaturns
join:2004-12-06
Beaverton, OR

rsaturns

Member

Re: [HELP] SIP Trunk to Analog phone on FXS port using CME

Not seeing your config here it's a bit of a shot in the dark. But you'll need a new pots dial-peer pointed at voice-port 0/0/0 with destination-pattern of XXXX. XXXX being the DNIS coming in from the SIP trunk that you want to ring on this analog port.

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

You know, I didn't know exactly what to add from the config... that's why I wanted to leave it open... What you suggested is the one thing I didn't try! I'll build it now, but I'll have to check it later. Thanks for the reply!

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

sk1939 to DaSneaky1D

Premium Member

to DaSneaky1D
said by DaSneaky1D:

For the life of me, I can't seem to get what I thought to be a simple concept to work...

I have a SIP trunk with an ephone right now. That works perfectly inbound/outbound. I've added a FXS card and DSP to my 3825. When I plug in my analog phone, I can make outbound calls and I can call between extension. However, I cannot find a way to simply allow an inbound call to reach this phone.

I've look into the sccp/stcapp config (which you apparently need an analog gateway to make properly work...) and tried a creating a new dial-peer to route inbound calls from my DID to the voice port, but it never worked. Any help or suggestion would be greatly appreciated!

voice-port is 0/0/0

When you get a chance, post you dial-patterns and/or a show tech-support sanitized.

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

Well, I tried variations of rsaturns' suggestions, but nothing is working.

"DID is 314.555.1234"

I've tried:

dial-peer voice 20 pots
incoming called-number 1234
direct-inward-dial
port 0/0/0

dial-peer voice 20 pots
incoming called-number 3145551234
direct-inward-dial
port 0/0/0

- and -

dial-peer voice 20 pots
destination-pattern 3145551234
port 0/0/0

The main parts of the FXS config is as follows:

voice-port 0/0/0
timeouts ringing 20
caller-id enable
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
dial-peer voice 102 pots
destination-pattern 102
port 0/0/0
DaSneaky1D

DaSneaky1D

MVM

I've tried applying the config (suggestions) listed at both of these links, to no avail. I'm completely lost as to what I'm missing:

»supportforums.cisco.com/ ··· -h323-gw

»www.techexams.net/forums ··· unk.html

All in all, I have a perfectly working system with my ephone. I've tried removing the ephone-dn's from the config to see if that made a difference, too. Unfortunately it didn't.

I don't mind posting the whole config, but everything else is just outbound dial-peers, COR lists and translation rules for outbound calls...nothing that should impact getting an inbound call to a FXS port.

sk1939
Premium Member
join:2010-10-23
Frederick, MD

sk1939

Premium Member

You do realize that to receive incoming calls via POTS you need an FXO card right? You cannot use an FXS card for incoming calls. Sorry I missed that earlier.

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

Correct, but I'm trying to receive the call via the SIP trunk, but have the call ring to a phone plugged into the FXS port. Currently I can only get calls to ring into my IP phone.

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

sk1939

Premium Member

Your dial pattern is off if your going from VoIP to POTs then I believe. The POTs command you have listed indicates a dial peer using basic telephone service, not VOIP

Try:

dial-peer voice 20 voip
destination-pattern 3145551234
port 0/0/0

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

Yeah, I tried that earlier... didn't work:

3825-1(config)#dial-peer voice 20 voip
3825-1(config-dial-peer)#port ?
% Unrecognized command
3825-1(config-dial-peer)#port

sk1939
Premium Member
join:2010-10-23
Frederick, MD

sk1939

Premium Member

Are you sure of your port numbers?

WIC 0 ports: 0/0/0, 0/0/1
WIC 1 ports: 0/1/0, 0/1/1
WIC 2 ports: 0/2/0, 0/2/1
WIC 3 ports: 0/3/0, 0/3/1

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

They're 0/0/0 thru 0/0/3

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

3 edits

sk1939

Premium Member

If Direct Inward Dialing (DID) is configured in the inbound POTS dial peer, the router uses one-stage dialing, which means that the full dialed string is used to match outbound dial peers.

You also need to have your destination pattern, port, VoIP, and session-target in the command. You can also try the answer-address command.

dial-peer voice 1 pots
incoming called-number (phone number)
voice-port (port)
direct-inward-dial
dial-peer voice 2 pots
destination-pattern (phone number)
session target (VoIP Provider)

dial-peer voice 4 pots
answer-address (Phone Number)
session target (VoIP provider)

(Session Target for IP clients and Port X/X/X for POTS.)

Edit: Please post your config, it is very hard to troubleshoot with only partial pictures.

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

Thanks for your continued assistance! I'll try those tonight when I get home.

voice service voip
ip address trusted list
ipv4 208.100.39.53
ipv4 208.100.39.54
ip address trusted call-block cause not-in-cug
gcid
clid substitute name
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
asserted-id ppi
midcall-signaling passthru
no call service stop
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
voice translation-rule 1
rule 5 /^10[1-5]$/ /[ACTUAL DID]/
!
voice translation-rule 2
rule 5 /^\([2-9]......\)$/ /314\1/
!
!
!
voice translation-profile 314-7-to-10-Digit
translate calling 1
translate called 2
!
voice translation-profile Caller-ID
translate calling 1
!
voice-port 0/0/0
timeouts ringing 20
description Cordless Phone
station-id name Cordless
caller-id enable
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
dial-peer cor custom
name 911
name Local
name LD
!
!
dial-peer cor list call-Local
member Local
!
dial-peer cor list call-LD
member LD
!
dial-peer cor list Ext-Allowed-Calls
member Local
member LD
!
dial-peer cor list All-Allowed-Calls
member 911
member Local
member LD
!
dial-peer cor list EMERGENCY
member 911
!
!
dial-peer voice 5 voip
corlist outgoing EMERGENCY
description Dialing Plan for 911
preference 1
destination-pattern 911
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10 voip
corlist outgoing call-LD
description Dialing Plan for LD
translation-profile outgoing Caller-ID
preference 1
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad
!
dial-peer voice 15 voip
corlist outgoing call-Local
description 7-digit Local for 314
translation-profile outgoing 314-7-to-10-Digit
preference 1
destination-pattern [2-9]......
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad
!
dial-peer voice 102 pots
description All Analog Phones
destination-pattern 102
port 0/0/0
!
telephony-service
no auto-reg-ephone
max-ephones 5
max-dn 6
ip source-address 10.11.1.1 port 2000
timeouts interdigit 3
load 7960-7940 P00307020200
max-conferences 3 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 31 2014 23:58:13
!
!
ephone-dn 1 dual-line
number [ACTUAL DID] no-reg primary
label Home
description Bat Cave
corlist incoming All-Allowed-Calls
!
!
ephone-dn 2 dual-line
number 101 no-reg primary
label Bat Cave - 101
corlist incoming Ext-Allowed-Calls
!
!
ephone 1
device-security-mode none
mac-address 0017.5976.86D4
type 7960
button 1:1 2:2
DaSneaky1D

DaSneaky1D to sk1939

MVM

to sk1939
This isn't going to work, as the pots dial-peer cannot have a session target, let alone one to an IP address:

dial-peer voice 2 pots
destination-pattern (phone number)
session target (VoIP Provider)

**tried and the router really didn't like that!

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

sk1939

Premium Member

said by DaSneaky1D:

This isn't going to work, as the pots dial-peer cannot have a session target, let alone one to an IP address:

dial-peer voice 2 pots
destination-pattern (phone number)
session target (VoIP Provider)

**tried and the router really didn't like that!

I had mentioned below that pots = port and VoIP = session target (might have forgotten that).

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

DaSneaky1D

MVM

Tried applying the below, but the call didn't route to the voice port. Only ephone-dn 1 rang.

dial-peer voice 1 pots
incoming called-number [DID NUMBER]
port 0/0/0
direct-inward-dial

dial-peer voice 2 pots
destination-pattern [DID NUMBER]
port 0/0/0

dial-peer voice 3 voip
preference 1
destination-pattern [DID NUMBER]
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

sk1939

Premium Member

It seems like your missing some stuff. Note VoIP.ms's sample configuration:

»wiki.voip.ms/article/PBX ··· isco_IOS

Your doing something that is increasingly rarely done (going from VoIP to POTS) instead of the other way around. It is frankly easier to use an ATA and configure CME as a strict gateway/PBX imo, and most Cisco-suggest configurations reflect that. However, your router should be able to do what your trying to do.

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

1 edit

DaSneaky1D

MVM

Well.... if I could have just paid attention to that page 5 days ago, I could have got it ringing!

It's working! Thanks for your help and suggestions. Both got me to where I needed to be!

### The Fix ###

I've trimmed down the dial-peers that I strictly needed to get this working:

dial-peer voice 1 pots
destination-pattern [FULL DID]
port 0/0/0
!
dial-peer voice 2 voip
description Main Line Inbound
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
incoming called-number [FULL DID]
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad

The problem arose because I had an ephone-dn that had my DID directly assigned as the "number" on it. That ephone-dn was taking precedent and accepting all the calls. Once I added the above dial-peers and removed it, calls ran perfectly.

Can't thank you enough for your assistance!

sk1939
Premium Member
join:2010-10-23
Frederick, MD
ARRIS SB8200
Ubiquiti UDM-Pro
Juniper SRX320

sk1939

Premium Member

said by DaSneaky1D:

Well.... if I could have just paid attention to that page 5 days ago, I could have got it ringing!

It's working! Thanks for your help and suggestions. Both got me to where I needed to be!

### The Fix ###

I've trimmed down the dial-peers that I strictly needed to get this working:

dial-peer voice 1 pots
destination-pattern [FULL DID]
port 0/0/0
!
dial-peer voice 2 voip
description Main Line Inbound
session protocol sipv2
session target dns:chicago3.voip.ms
session transport udp
incoming called-number [FULL DID]
voice-class codec 1
no voice-class sip block 180
no voice-class sip block 183
no voice-class sip block 181
dtmf-relay rtp-nte
no vad

The problem arose because I had an ephone-dn that had my DID directly assigned as the "number" on it. That ephone-dn was taking precedent and accepting all the calls. Once I added the above dial-peers and removed it, calls ran perfectly.

Can't thank you enough for your assistance!

Your welcome, glad that it is finally working. I assume that your VoIP line continues to work, correct?

DaSneaky1D
what's up
MVM
join:2001-03-29
The Lou

1 recommendation

DaSneaky1D

MVM

I can make inbound/outbound calls just fine. The ephone is actually the least of my concern. I really wanted to get the analog phones working. If I could get both ringing inbound and answering successfully, then that's a plus, but not a major concern.