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New Member

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Hi all,

I have this issue with an FXO port where if i make an inbound call to the port, the IP phone rings for less than a second, cuts off, rings for less than a second, cuts off, and so on.  On the mobile i'm making the call from, it sounds fine.

Config i have at the moment is -

voice-port 4

translation-profile incoming Inbound_Call

translation-profile outgoing Outbound_Call

cptone GB

impedance complex3

description *** PSTN Fallback ***

caller-id alerting line-reversal

I can't find anything online describing this issue - could it be something to do with the BT wall socket being either a master or a secondary, or a supervisory-disconnect issue?

Thanks in advance,

Jason


1 ACCEPTED SOLUTION

Accepted Solutions
Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Jason,

010050: *Nov 25 15:17:40.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected

JMHRAY-VG01#

010051: *Nov 25 15:17:40.396 GMT: htsp_timer2 - 1000 msec

010052: *Nov 25 15:17:40.396 GMT: htsp_timer_stop

010053: *Nov 25 15:17:41.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc

010054: *Nov 25 15:17:41.396 GMT: htsp_timer_stop

010055: *Nov 25 15:17:41.396 GMT: htsp_timer_stop2

010056: *Nov 25 15:17:41.396 GMT: [4] set signal state = 0x4 timestamp = 0

010057: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnected:

   Cause Value=34, Interface=0x868B0B70, Call Id=185

It appears that there is no dial tone on the line

Can you check that a POTS phone works on the wiring.

Can you also make sure that the wiring is correct

FXO --RJ11 ---3 -------  BT PSTN JACK  2

FXO --RJ11 ---4 -------  BT PSTN JACK  5

You must make sure that these are the only wires connected.

Also try a pair reversal

HTH

Alex

Regards, Alex. Please rate useful posts.
9 REPLIES
Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Jason,

Can you change the voice port

!

voice-port 4

no caller-id alerting line-reversal

caller-id enable

caller-id alerting dsp-pre-alloc

supervisory disconnect dualtone mid-call

!

Let us know if this helps

Regards

Alex

Regards, Alex. Please rate useful posts.
New Member

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Hi Alex,

Tried the above, still not working.  Config is below -

voice-port 4

translation-profile incoming Inbound_Call

translation-profile outgoing Outbound_Call

supervisory disconnect dualtone mid-call

cptone GB

impedance complex3

description *** PSTN Fallback 020 7235 1623 ***

caller-id alerting dsp-pre-allocate

!

!

!

sh voice port 4

Foreign Exchange Office 4 Slot is 0, Sub-unit is 3, Port is 0

Type of VoicePort is FXO

Operation State is DORMANT

Administrative State is UP

The Last Interface Down Failure Cause is Administrative Shutdown

Description is *** PSTN Fallback ***

Noise Regeneration is enabled

Non Linear Processing is enabled

Non Linear Mute is disabled

Non Linear Threshold is -21 dB

Music On Hold Threshold is Set to -38 dBm

In Gain is Set to 0 dB

Out Attenuation is Set to 3 dB

Echo Cancellation is enabled

Echo Cancellation NLP mute is disabled

Echo Cancellation NLP threshold is -21 dB

Echo Cancel Coverage is set to 64 ms

Echo Cancel worst case ERL is set to 6 dB

Playout-delay Mode is set to adaptive

Playout-delay Nominal is set to 60 ms

Playout-delay Maximum is set to 1000 ms

Playout-delay Minimum mode is set to default, value 40 ms

Playout-delay Fax is set to 300 ms

Connection Mode is normal

Connection Number is not set

Initial Time Out is set to 15 s

Interdigit Time Out is set to 10 s

Call Disconnect Time Out is set to 60 s

Power Denial Disconnect Time Out is set to 1000 ms

Ringing Time Out is set to 180 s

Wait Release Time Out is set to 30 s

Companding Type is u-law

Region Tone is set for GB

Analog Info Follows:

Currently processing none

Maintenance Mode Set to None (not in mtc mode)

Number of signaling protocol errors are 0

Impedance is set to complex3 Ohm

Station name None, Station number None

Caller ID Info Follows:

Standard ETSI

DSP is pre-allocated

Translation profile (Incoming): Inbound_Call

Translation profile (Outgoing): Outbound_Call

Voice card specific Info Follows:

Signal Type is loopStart

Battery-Reversal is enabled

Number Of Rings is set to 1

Supervisory Disconnect is dualtone mid-call

Answer Supervision is inactive

Hook Status is On Hook

Ring Detect Status is inactive

Ring Ground Status is inactive

Tip Ground Status is inactive

Dial Out Type is dtmf

Digit Duration Timing is set to 100 ms

InterDigit Duration Timing is set to 100 ms

Pulse Rate Timing is set to 10 pulses/second

InterDigit Pulse Duration Timing is set to 750 ms

Percent Break of Pulse is 65 percent

GuardOut timer is 2000 ms

Minimum ring duration timer is 125 ms

Hookflash-in Timing is set to 600 ms

Hookflash-out Timing is set to 400 ms

Supervisory Disconnect Timing (loopStart only) is set to 350 ms

OPX Ring Wait Timing is set to 6000 ms

Thanks,

Jason

Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Jason,

Normally we need to use PLAR to direct the call to a specified number.

I do not see any PLAR here

What does the Translation profile (Incoming): Inbound_Call

contain

Regards

Alex

Regards, Alex. Please rate useful posts.
New Member

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Hi Alex,

See below.  4004 is an internal extension i'm using for testing, but the PSTN line is supposed to be for local 999 breakout, so incoming working is not essential.

In CUCM, i have a route group configured to send any emergency numbers to the 881, but when i try an outbound call i get a Disconnect Cause=34, which i believe means no cct/channel available.  Some debug below as well -

sh deb

CCAPI:

  debug voip ccapi error call is ON (filter is OFF)

  debug voip ccapi error software is ON

  debug voip ccapi inout is ON (filter is OFF)

Voice Port Module signaling debugging is on

#

010013: *Nov 25 15:17:40.136 GMT: //-1/00FD1AFE1000/CCAPI/cc_api_display_ie_subfields:

   cc_api_call_setup_ind_common:

   cisco-username=Office

   ----- ccCallInfo IE subfields -----

   cisco-ani=4004

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=9123

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=-1

   cisco-rdnplan=-1

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

010014: *Nov 25 15:17:40.136 GMT: //-1/00FD1AFE1000/CCAPI/cc_api_call_setup_ind_common:

   Interface=0x863795A8, Call Info(

   Calling Number=4004,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=9123(TON=Unknown, NPI=Unknown),

   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,

   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=184

010015: *Nov 25 15:17:40.136 GMT: //-1/00FD1AFE1000/CCAPI/ccCheckClipClir:

   In: Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

010016: *Nov 25 15:17:40.136 GMT: //-1/00FD1AFE1000/CCAPI/ccCheckClipClir:

   Out: Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

010017: *Nov 25 15:17:40.140 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010018: *Nov 25 15:17:40.140 GMT: :cc_get_feature_vsa malloc success

010019: *Nov 25 15:17:40.140 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010020: *Nov 25 15:17:40.140 GMT:  cc_get_feature_vsa count is 1

010021: *Nov 25 15:17:40.140 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010022: *Nov 25 15:17:40.140 GMT: :FEATURE_VSA attributes are: feature_name:0,feature_time:2264524184,feature_id:184

010023: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/cc_api_call_setup_ind_common:

   Set Up Event Sent;

   Call Info(Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=9123(TON=Unknown, NPI=Unknown))

010024: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/cc_process_call_setup_ind:

   Event=0x8686EED0

010025: *Nov 25 15:17:40.140 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:

   Try with the demoted called number 9123

010026: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/ccCallSetContext:

   Context=0x84F424B4

010027: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 184 with tag 1 to app "_ManagedAppProcess_Default"

010028: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

010029: *Nov 25 15:17:40.140 GMT: //184/00FD1AFE1000/CCAPI/ccCallSetupRequest:

   Destination=, Calling IE Present=TRUE, Mode=0,

   Outgoing Dial-peer=9, Params=0x84F433C4, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)

010030: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccCheckClipClir:

   In: Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

010031: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccCheckClipClir:

   Out: Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)

010032: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccCallSetupRequest:

   Destination Pattern=9T, Called Number=123, Digit Strip=TRUE

010033: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccCallSetupRequest:

   Calling Number=4004(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=123(TON=Unknown, NPI=Unknown),

   Redirect Number=, Display Info=

   Account Number=Office, Final Destination Flag=TRUE,

   Guid=00FD1AFE-8BB0-F1EC-1000-12010A05050E, Outgoing Dial-peer=9

010034: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/cc_api_display_ie_subfields:

   ccCallSetupRequest:

   cisco-username=Office

   ----- ccCallInfo IE subfields -----

   cisco-ani=4004

   cisco-anitype=0

   cisco-aniplan=0

   cisco-anipi=0

   cisco-anisi=1

   dest=123

   cisco-desttype=0

   cisco-destplan=0

   cisco-rdie=FFFFFFFF

   cisco-rdn=

   cisco-rdntype=-1

   cisco-rdnplan=-1

   cisco-rdnpi=-1

   cisco-rdnsi=-1

   cisco-redirectreason=-1   fwd_final_type =0

   final_redirectNumber =

   hunt_group_timeout =0

010035: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccIFCallSetupRequestPrivate:

   Interface=0x868B0B70, Interface Type=6, Destination=, Mode=0x0,

   Call Params(Calling Number=4004,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),

   Called Number=123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=9, Call Count On=FALSE,

   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

010036: *Nov 25 15:17:40.144 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010037: *Nov 25 15:17:40.144 GMT: :cc_get_feature_vsa malloc success

010038: *Nov 25 15:17:40.144 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010039: *Nov 25 15:17:40.144 GMT:  cc_get_feature_vsa count is 2

010040: *Nov 25 15:17:40.144 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

010041: *Nov 25 15:17:40.144 GMT: :FEATURE_VSA attributes are: feature_name:0,feature_time:2264523960,feature_id:185

010042: *Nov 25 15:17:40.144 GMT: //185/00FD1AFE1000/CCAPI/ccIFCallSetupRequestPrivate:

   SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1

010043: *Nov 25 15:17:40.144 GMT: //185/00FD1AFE1000/CCAPI/ccCallSetContext:

   Context=0x84F43374

010044: *Nov 25 15:17:40.144 GMT: //184/00FD1AFE1000/CCAPI/ccSaveDialpeerTag:

   Outgoing Dial-peer=9

010045: *Nov 25 15:17:40.144 GMT: htsp_timer_stop3 htsp_setup_req

010046: *Nov 25 15:17:40.144 GMT: htsp_process_event: [4, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup

010047: *Nov 25 15:17:40.144 GMT: [4] set signal state = 0xC timestamp = 0

010048: *Nov 25 15:17:40.148 GMT: htsp_timer - 1300 msec

010049: *Nov 25 15:17:40.148 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_proceeding:

   Interface=0x868B0B70, Progress Indication=NULL(0)

010050: *Nov 25 15:17:40.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected

JMHRAY-VG01#

010051: *Nov 25 15:17:40.396 GMT: htsp_timer2 - 1000 msec

010052: *Nov 25 15:17:40.396 GMT: htsp_timer_stop

010053: *Nov 25 15:17:41.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc

010054: *Nov 25 15:17:41.396 GMT: htsp_timer_stop

010055: *Nov 25 15:17:41.396 GMT: htsp_timer_stop2

010056: *Nov 25 15:17:41.396 GMT: [4] set signal state = 0x4 timestamp = 0

010057: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnected:

   Cause Value=34, Interface=0x868B0B70, Call Id=185

010058: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=34, Retry Count=0)

010059: *Nov 25 15:17:41.396 GMT: //184/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:

   release reserved xcoding resource.

010060: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/ccCallSetAAA_Accounting:

   Accounting=1, Call Id=185

010061: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/ccCallDisconnect:

   Cause Value=34, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=34)

010062: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/ccCallDisconnect:

   Cause Value=34, Call Entry(Responsed=TRUE, Cause Value=34)

010063: *Nov 25 15:17:41.396 GMT: htsp_process_event: [4, FXOLS_ONHOOK, E_HTSP_RELEASE_REQ]fxols_onhook_release

010064: *Nov 25 15:17:41.400 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x868B0B70, Tag=0x0, Call Id=185,

   Call Entry(Disconnect Cause=34, Voice Class Cause Code=0, Retry Count=0)

010065: *Nov 25 15:17:41.400 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

010066: *Nov 25 15:17:41.400 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

010067: *Nov 25 15:17:41.400 GMT: :cc_free_feature_vsa freeing 86F9E4B0

010068: *Nov 25 15:17:41.400 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

010069: *Nov 25 15:17:41.400 GMT:  vsacount in free is 1

010070: *Nov 25 15:17:41.400 GMT: //184/00FD1AFE1000/CCAPI/ccCallDisconnect:

   Cause Value=34, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

010071: *Nov 25 15:17:41.400 GMT: //184/00FD1AFE1000/CCAPI/ccCallDisconnect:

   Cause Value=34, Call Entry(Responsed=TRUE, Cause Value=34)

010072: *Nov 25 15:17:41.400 GMT: //184/00FD1AFE1000/CCAPI/cc_api_get_transfer_info:

   Transfer Number Is Null

010073: *Nov 25 15:17:41.404 GMT: //184/00FD1AFE1000/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x863795A8, Tag=0x0, Call Id=184,

   Call Entry(Disconnect Cause=34, Voice Class Cause Code=0, Retry Count=0)

010074: *Nov 25 15:17:41.404 GMT: //184/00FD1AFE1000/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

010075: *Nov 25 15:17:41.404 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

010076: *Nov 25 15:17:41.404 GMT: :cc_free_feature_vsa freeing 86F9E590

010077: *Nov 25 15:17:41.404 GMT: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

010078: *Nov 25 15:17:41.404 GMT:  vsacount in free is 0

voice translation-rule 30

rule 1 /^$/ /4004/

voice translation-rule 40

rule 1 /^1/ /901/

rule 2 /^2/ /902/

rule 3 /^3/ /903/

rule 4 /^4/ /904/

rule 5 /^5/ /905/

rule 6 /^6/ /906/

rule 7 /^7/ /907/

rule 8 /^8/ /908/

rule 9 /^9/ /909/

voice translation-profile Inbound_Call

translate calling 40

translate called 30

Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Jason,

010050: *Nov 25 15:17:40.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected

JMHRAY-VG01#

010051: *Nov 25 15:17:40.396 GMT: htsp_timer2 - 1000 msec

010052: *Nov 25 15:17:40.396 GMT: htsp_timer_stop

010053: *Nov 25 15:17:41.396 GMT: htsp_process_event: [4, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc

010054: *Nov 25 15:17:41.396 GMT: htsp_timer_stop

010055: *Nov 25 15:17:41.396 GMT: htsp_timer_stop2

010056: *Nov 25 15:17:41.396 GMT: [4] set signal state = 0x4 timestamp = 0

010057: *Nov 25 15:17:41.396 GMT: //185/00FD1AFE1000/CCAPI/cc_api_call_disconnected:

   Cause Value=34, Interface=0x868B0B70, Call Id=185

It appears that there is no dial tone on the line

Can you check that a POTS phone works on the wiring.

Can you also make sure that the wiring is correct

FXO --RJ11 ---3 -------  BT PSTN JACK  2

FXO --RJ11 ---4 -------  BT PSTN JACK  5

You must make sure that these are the only wires connected.

Also try a pair reversal

HTH

Alex

Regards, Alex. Please rate useful posts.
New Member

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Cannibalised the RJ11-BT wiring, and it's now working...  Thanks!!!

So if i'm buying the correct RJ11 - BT male wire, what am i looking for?

Cheers

Jason

Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Jason

I normally use any old RJ11 cord , cut off the other end and Krone the wires from Rj11 3 & 4 (ONLY)

on to the BT wiring 2 & 5 in a jack.

In bigger offices etc the FXO lines appear on the Krone strip or an RJ45 panel

Another note is if it is RJ45 then the pair should be on RJ45 -4 & 5 (Usually the blue/blue-white)

HTH

Alex

Regards, Alex. Please rate useful posts.
New Member

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

I have a simple question, the 881SRTS router can be register with a CUCM 7 and work has a remote gateway (SCCP,MGCP or H323) with FXO lines ?

Green

SRST881 FXO Port (UK) - phone keeps disconnecting during ringing

Hi,

Looks like you are correct

http://www.cisco.com/en/US/docs/routers/access/800/860-880-890/software/configuration/guide/overview.html

Voice Features

The Cisco 880 voice and data platforms (C880SRST, C880SRSTW, C881-V, C887 VA-V, and C887VA-V-W) support the following voice features:

Signaling protocols: Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and H323

Real-time transfer protocol (RTP), Cisco RTP (cRTP), and secure RTP (SRTP) for these signaling protocols

Fax passthrough, Cisco fax relay, T37 fax store-and-forward, and T.38 fax relay (including T.38 gateway-controlled MGCP fax relay)

Dual tone multifrequency (DTMF) Relay—OOB and RFC2833

Silence suppression/comfort noise

G.711 (a-law and u-law), G.729A, G.729AB, G.729, G.729B, G.726

Support of SRST failover to a Foreign Exchange Office (FXO) or BRI backup port connected to PSTN in case of WAN failure on C880SRST and C880SRSTW.

Support for SRST and CME requires user license, but only a 5-user license is supported on C881-V, C887VA-V, and C887VA-V-W routers.

Direct inward dialing (DID) on FXS

Regards

Alex

Regards, Alex. Please rate useful posts.
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