EFTPOS No Response X0 Using FXS Port
Hi All,
Trying to get an EFTPOS machine working connected into an FXS Port using SCCP/STCAPP Protocol for call control.
The port is registered in CUCM, I can connect an analogue phone to the port and successfully call the Bank's 1800 Number etc, I can make other external calls no problem.
When I try the EFTPOS Machine, the call setup runs through normally and I see the call is then in a connected state. Approx about 10 or seconds later the call is cleared by the local device with an 0x8090 code Normal Clearing. The call is then disconnected and released as per any other call.
The EFTPOS Machine prints a error for "Declined No Response X0"
I have attached the following debugs.
debug isdn q931
debug voip vtsp all
debug vop ccapi inout
debug voip rtp session named-event
Calling Number is 0245604627
Called Number is 1800509183
General environment setup is as follows.
EFTPOS connect to FXS Port -> Registered to CUCM (9.1.2) -> SIP Trunk -> IOS 15.x -> ISDN Pri.
Any feedback i smuch appreciated.
Ben
Not so sure if the service code being utilize by the FXS is the same as what is mentioned in VSA.Then again, the manual says "dial the corresponding * code on the client station" and if you look at an SPA9xx phone, it has the following codes defined for call park and call unpark code.
*38
*39
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Call park using fxs port on spa9000
I have seen this question asked several times but no seems have answered it.
Is it possible to enter a service code to park a call using an analogue phone. If it is, please share what is the service code and the procedure.
I have a spa9000 with pstn (spa3002, spa400) and exts spa-942,xlite and spa2102/pap2. Everything works fine on the 942 phones and the xlite softphones but cant get more that the *98 transfer feature working on the analog phones.Not so sure if the service code being utilize by the FXS is the same as what is mentioned in VSA.Then again, the manual says "dial the corresponding * code on the client station" and if you look at an SPA9xx phone, it has the following codes defined for call park and call unpark code.
*38
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HO Router (FXO card)
Current configuration : 1718 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
resource policy
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
ip subnet-zero
ip cef
no ip dhcp use vrf connected
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.1 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.1
ip classless
no ip http server
control-plane
voice-port 2/0
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2001
description Remote PSTN#:35296913
music-threshold -70
voice-port 2/1
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2002
description Remote PSTN#:35296914
music-threshold -70
voice-port 3/0
voice-port 3/1
dial-peer voice 2000 voip
destination-pattern 200.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.2
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
dial-peer voice 1321 pots
description line 1
huntstop
destination-pattern 1321
port 2/0
dial-peer voice 1322 pots
description line 2
huntstop
destination-pattern 1322
port 2/1
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end
Remote Router (FXS Card):
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
no aaa new-model
ip subnet-zero
ip cef
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.2 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.2
interface Ethernet1/0
no ip address
shutdown
half-duplex
ip classless
no ip http server
control-plane
voice-port 2/0
description PSTN#:
voice-port 2/1
description PSTN#:
voice-port 3/0
voice-port 3/1
dial-peer voice 2001 pots
description Remote
huntstop
destination-pattern 2001
port 2/0
dial-peer voice 2002 pots
description Remote
huntstop
destination-pattern 2002
port 2/1
dial-peer voice 1320 voip
destination-pattern 132.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.1
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
endIn your voice class codec 10 there aren't any codecs declared.
Add G.711 codec in this way:
voice class codec 10
codec preference 1 g711alaw
If the fax communication fails again try to disable T.38 and try fax passthrough mode:
no fax rate
modem passthrough nse codec g711alaw
fax protocol pass-through g711alaw
Regards. -
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vy MarkusSince there has been no response to your post, it appears to be either too complex or too rare an issue for other forum members to assist you. If you don't get a suitable response to your post, you may wish to review our resources at the online Technical Assistance Center (http://www.cisco.com/tac) or speak with a TAC engineer. You can open a TAC case online at http://www.cisco.com/tac/caseopen
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Cisco 2432-24FXS Calls won't ring FXS ports when placed in Huntgroup
Hi,
I am having an issue getting calls to ring through on the FXS ports when using the "trunkgroup" command under the POTS dial-peers. Calls will ring through fine if I use the "port" command under the pots dial-peer. What I am trying to accomplish is getting a single number to hunt through 13 FXS ports in a round-robin fashion. With the below setup I am getting a SIP 404 not found message in the SIP debug info. I am not sure if I am missing something here or what the deal is. If the below information is not enough I would be happy to provide additional info. Any idea's why this configuration doesn't seam to work?
trunk group HuntGroup1
hunt-scheme round-robin both up
dial-peer voice 1 voip
translation-profile outgoing 7to11
destination-pattern .T
progress_ind setup enable 3
no modem passthrough
voice-class codec 1
session protocol sipv2
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dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
no vad
dial-peer voice 300 pots
trunkgroup HuntGroup1
description TEST NUMBER
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voice-port 2/0
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 1
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/1
trunk-group HuntGroup1
timeouts interdigit 3
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station-id number XXXXXXXXXX
caller-id enable
voice-port 2/2
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 3
station-id number XXXXXXXXXX
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station-id number XXXXXXXXXX
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trunk-group HuntGroup1
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trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 8
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caller-id enable
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timeouts interdigit 3
description HuntGroup 1 Line 9
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/9
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 10
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/10
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 11
station-id number XXXXXXXXXX
caller-id enable
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trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 12
station-id number XXXXXXXXXX
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trunk-group HuntGroup1
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Message was edited by: Thomas Schmidt
Added CCSIP Debug and Router Config.Hi.
Can you please attach the complete config and a debug ccsip message?
Thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App -
I am using Call-manager 6.0.1b, a MGCP controlled Gateway.On the Gateway i have installed a NM-HD-2V with a Vic2-2FXS module. On this module i connect 2 analog phone with capability to display the Caller-ID. When i call the analog port from a ip phone or from the other analog phone the Dn is not shown. When i connect the phone directly to the PSTN and dial this nr via my cell phone the nr is shown so i expect that the nr format received is not correct. How can this performed that the correct format is shown to the FXS port connected phone ?
Hi,
Yes - you are correct. Looked at this one too quickly.
You will want to make sure the CPTONE defined on the port is for the country the phone is manufactured for, and that the voice-pport has the 'caller-id enable' command.
If those are both correct, and I'm guessing that they are since some caller-id works, then you need to inspect the gateways that take the calls to begin with.
Do you have caller-id trouble for internal calls also?
How do these calls come into your network?
hth,
nick -
Is there a way to make an analog phone connected to an FXS port a part of a call pickup group that contains both analog phones & IP phones? I setup a lab and used MGCP to add the gateway and I was able to add the DN associated with the FXS port to a call pickup group. However, I am unable to figure out how to answer the call from the analog phone when another IP phone in the call pickup group is ringing.
Thanks in advanceHi
You are going down the right track with this.
http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1dde5372/0#selected_message
See this other post I made (for a different purpose, but the principal is the same - it just opens up features available to IP phones for FXS ports by registering them using SCCP).
Regards
Aaron
Please rate helpful posts... -
Restricting FXS ports to internal calls only
Hi,
I have recently intalled Call Manager 8.5.1 with a H323 Gateway. On the gateway, I have a number of FXS ports for lobby phones. Is there a way I can restrict these phones to allow only internal calls only? Do I need to use COR lists like in Call Manager express or is there a more practical way?
Thanks,
DerekHi Derek,
It sounds like these FXS ports are NOT registered to CUCM 8.5.1? If they aren't, then yes COR lists is the most pratical way to do this. You could do a "connection plar xxxx" to a receptionist extension and her her/him forward the calls to the correct extension. -
Hi,
We are running VoIPovFR to a remote location, which has 2 x 4FXS/DID module installed into a C2621XM chassis. We are using IOS c2600-is-mz.122-15.
Out of the 8 ports, two continually go off-hook, FXSLS_WAIT_OFFHOOK before finally going into a FXSLS_PARK state. I have moved the cabling and the error occurs on which ever port the two lines plugs into.
Has anyone ever see this before. To my mind it looks like a cabling issue?
Any ideas?
ThanksTry wiring all FXS ports like this
FXS PORT 0 RJ11 3-----------------------A ANALOGUE
FXS PORT 0 RJ11 4-----------------------B PHONE
FXS PORT 1 RJ11 3-----------------------A ANALOGUE
FXS PORT 1 RJ11 4-----------------------B PHONE
FXS PORT 2 RJ11 3-----------------------A ANALOGUE
FXS PORT 2 RJ11 4-----------------------B PHONE
FXS PORT 3 RJ11 3-----------------------A ANALOGUE
FXS PORT 3 RJ11 4-----------------------B PHONE
Ensure that all other wires are left disconnected -
Transmission Volume on a FXS Port
G'Day,
We have a fax machine connected to an FXS port in a 2851 (CME 4.1). Running tests on the machine reveal that the transmission volume is set too high, causing distortion on faxes and an inability to fax to some destinations.
Is there a way to turn the transmission volume down at the port-level? Apparently it cannot be done on the fax machine (? this sounds odd to me...but the technician for the machine is adament)
Thanks,
-marti-This depends if the faxing is being done via fax relay or fax passthrough.
If fax relay is being used, the fax traffic originates from the egress voice port connected to the PSTN, not from the FXS port that has the fax machine on it. The output level of the fax relay codec is fixed - at this stage it is not possible to alter the level of the outgoing audio on the 2800/3800 ISR's.
If you use fax passthrough, the audio is transmitted via a G711 codec. You can use the voice port 'input-gain' or 'output-attenuation' command to boost or reduce the levels in different directions.
If the level FROM the CME system towards the PSTN is too high (for example 3dB), you would use the command 'input-gain -3' on the FXS port to reduce the inwards audio level by 3dB, so the level SENT to the PSTN is correspondingly reduced by 3dB.
Most fax machines can set the outgoing audio levels - they may have a special config menu that is not normally directly accessible by users. -
Hi,
I have a VIC3-2FXS/DID in Switerzland and my customer has asked for it to be connected to a fax machine through Strucuted cabling and they would like to know what type of cables they required.
They want RJ11 (FXS Port) to RJ45 (Patch Panel) then RJ45 (Patch Panel) to RJ11(Fax Machine).
Could someone explain the pins out of the cables, as it is going to be two RJ11 to RJ45 cables do they need to rollover?
Thanks for any help.All cabling on Cisco voice cards is straight through from the port to the device. RJ11 has 2 pins, for tip and ring. That's blue and white/blue. RJ45 has 8 pins, but if you want to stick with the 568B wiring standard on the RJ45 jacks,you would use this.
Blue: RJ11(pin1)----RJ45(pin 4)--------(pin4)RJ45-----(pin 1)RJ11
White/Blue: RJ11(pin2)----RJ45(pin 5)--------(pin 5)RJ45-----(pin 2)RJ11
If you actually have RJ14 jacks (4 pins), its pins 2 and 3 (the middle pins) which are used. If it is RJ25, it's still the middle pins, which would be 3 and 4. -
Hello,
I have a problem with my SPA3102's FXS port.
In last April I tested my friend's two PAP2T adapters with four phones. The adapters were connected to same switch with different fix IP addresses and phones were able to ring each other. Then I got one of adapters from him as a gift and on 2nd of January I bought an SPA3102 - so I have two adapters.
I connected my adapters to same switch. Their parameters are:
PAP2T IP: 192.168.0.221
FXS #1:
SIP port: 5060
display name: 1
user ID: 1
FXS #2:
SIP port: 5061
display name: 2
user ID: 2
SPA3102 IP: 192.168.0.222
FXS:
SIP port: 5062
display name: 3
user ID: 3
Dialplans:
PAP2T Line1: (<2:>S0 <:[email protected]:5061>|<3:>S0 <:[email protected]:5062>|<[x*]:>S0 <:[email protected]:5060>)
PAP2T Line2: (<1:>S0 <:[email protected]:5060>|<3:>S0 <:[email protected]:5062>|<[x*]:>S0 <:[email protected]:5061>)
SPA3102 FXS: (<1:>S0 <:[email protected]:5060>|<2:>S0 <:[email protected]:5061>|<[x*]:>S0 <:[email protected]:5062>)
Well there is problem with third phone (SPA3102 FXS port). Phones with PAP2T work perfectly, they can call each other. The 3rd phone has dialtone sometimes only but it is unable to call the other two phones. "1" and "2" can call third phone sometimes but another time I get reorder tone when I dial "3".
I have made backups from adapters' settings. Does somebody have any idea about this problem?
Thanks in advance,,,Hi,
Problem is solved...
SPA3102 has two RJ45 outlets: LAN and WAN (internet). I have used LAN only because of switch and the INTERNET terminal was empty. It appears to me this situation drives SPA3102 crazy... I had to make a "loopback" plug (I linked contact 1 to contact 3 and contact 2 to contact 6) and insert it into INTERNET outlet. So all of phones work well...
If you want to use a PAP2T lonely (without internet access) as a "micro-PBX" with two extensions, you will have to use same plug otherwise phones won't work. -
Hi all,
I would like to know how i can see if i have any traffic in a fxs port.
thanks.Hi,
You can check the status of the calls using the following command.
show call active voice brief
To check the status of the port, use the following:
show voice port status
show voice port summary
HTH,
Jagpreet Singh Barmi -
Installing an analog polycom soundstation 2 on FXS port in CUCME
I apologize if this is a stupid question, I'm an Avaya voice (cisco data) guy, I'm still learning Cisco voice.
I've installed an analog polycom soundstation 2, I can make internal and external calls. However I can only receive one incoming call at at time (second call receives a busy signal) and I can't conference a second call.
From researching I think I need to change the FXS port from MGCP to SCCP (I have the license for it) but I'm not 100% sure that's correct and if it is I'm not sure how to do it.
Any advice would be much appreciated.This should give you an idea where to start
http://www.icciev.com/1/post/2011/09/adding-vg224-to-cucm-80-as-sccp-or-mgcp-gateway-differences-and-configurations-part-2.html
Jorge Armijo
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