Restrict FXO ports on the Outgoing Calls.

I need to configure a restriction on the Outgoing Calls, i need the extension that start whit 1 (100,101,102,etc..) to only use the FXO ports 0/1/0 0/1/1 0/1/2 and the extensiones that start whit 2 (200,201,202,etc...) to only use the FXO ports 0/2/0 0/2/1 0/2/2, how can i make this happen ?

Hello Mario,
First off, this is not possible with CCA, this can only be done though CLI.
1. Setup seperate PSTN Trunk Groups. A: 0/1/0, 0/1/1, 0/1/2 B: 0/2/0, 0/2/1, 0/2/2
2. Setup Dial-Peers for each Trunk Group.
3. Setup COR list to restrict which users can use either Trunk Group.
That is the simple explaination, the actual setup can get quite complicated and lengthy.
Thank you,
Darren

Similar Messages

Last night, I saw my iPhone on the table in front of me and twice it made phone calls on its own. I just watched it make the outgoing call as it sat in front of me. Should I think that it has a virus? What should I do about it?

Last night, I saw my iPhone on the table in front of me and twice it made phone calls on its own. I just watched it make the outgoing call as it sat in front of me. Should I think that it has a virus? What should I do about it?
Also, it's been a little quirky lately: slow to respond to key pad touching and making it impossible to delete Apps?
Should I try wiping it and restoring it to factory settings?
Thank you.
Ben

It isn't a virus.
I would try the Restore from backup first and if that does not help try a Restore as a new iPhone.

The outgoing call icon has just appeared in the menu bar at the top of the screen on my iPhone 5 and i cant seem to get rid of it.

The outgoing call icon has just appeared in the menu bar at the top of the screen (next to the wifi icon) on my iPhone 5 and no matter what i do i cant seem to get rid of it.
I have rebooted and and cleared all running apps and still on luck.
The phone still works ok and i can make and receive calls.... any ideas on how to get rid of it?
Thanks

Here's what it looks like.
Mine did the same, just randomly overnight

Uc560 fxo port not answering incoming calls

Hi,
My customer is facing problem for incoming calls in uc560 fxo port.They have 12 PSTN lines which is connected to UC system.System is configured with Auto-Attendant also. almost all days they are facing this major issue of incoming call is not getting answered by UC560 and caller can hear the line is ringing.While the time of this problem I can see some of the FXO port status LED is UP and not disconnecting even if no one is on call also.Once remove the cable from the FXO port and connect it back the problem will solve for time being.What will be the reason for this issue of line getting held.Is there any configuration needs to change in FXO module? Below is the configuration I done on all 12 FXO ports. Please check and
suggest me a solution.

HI Paolo,
Thank you provoding the proper documentation .
On the system side I made the change by keeping companding type from a-law to u-law and enabled battery reverse.This setting works fine for last three days and now again the customer is facing the same problem of FXO port get held and incoming calls are just ringing and system is not answering even.
How to get proper solution for this issue????
Please help me............
Regards,
Rinchuraj

UC560 showing no dsp on FXO ports when making a call.

Hi Everyone,
We are having 2 customers who are facing this problem on 2 differenct model one is UC560 and another one is UC540.
This is happening with almost all the ports that when ever we are making a call the voice ports hangs showing no dsp and it hangs in that state until we
issue the shut & no shut the ports.
This is now becoming really annoying why this thing is happening repeatedly.
i have attached the configuration, show dsp-group all, show voice-call status and show voice port summary outputs.
Any help will be really appreciated.

mohd abdul malik wrote:the smartnet for this device has already expired.We do not have other option than support community i guess
It's customer choice to renew contracts or not, but don't expect freely contributed forums to fix Cisco bugs or serious issues.

3825 FXO Port remains in off-hook after call

Hello,
I have a 3825 router with 8 FXO ports running Cisco IOS Software, 3800 Software (C3825-SPSERVICESK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2). The problem we are facing is that after a call is placed through any of the FXO ports and the call is ended by the user, the port remains in off-hook till a reset of the port is done or someone restarts the router. Only then is the port accessible again.
I am thinking of changing the cards, but i do not want to invest in replacing the cards and then find out that this doesnt solve the problem.
The wierd thing is that this issue started on its own accord not too long ago.
Comments and suggestions please!
Regards,
Femi

Hello,
I do not want to change the FXO card till I am sure that is the problem and I did state that I always have to reboot the router when the problem starts. Rebooting clears the problem but it is back immediately I attempt a call again and hang up that call.
I have timeouts call-disconnect already configured, see below:
voice-port 0/0/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 2626878
caller-id enable type 1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
Regards,
Femi

Fax line used for outgoing calls when incomming fax not coming in?

So is there a way to do this? Can I route the 1 FXO to FXS for incomming calls (and plug fax into FXS) or something like this and then use that line as an outgoing line when not in use?
I don't need step but steps but the basics of how something like this could be configured would be awesome!!!

I'm not sure if I understand. I set this up, so I understand phone systems enough to do this, but I don't really don't know the terminoligy.
Let me back up.
Currently our business has 2 lines and a fax line. Right now the two lines are plugged into 2 FXO ports and the fax line is just plugged into the fax machine--not the UC320. We're in mixed mode, not pbx specifically. We get one or two faxes a day and we send two or 3 faxes a day. We'd like to be able to use the fax line for outgoing calls when it's not being used for fax.
So I need to be able to control the in and the out of the fax line. I need all incomming calls for the fax number routed to what I assume would be the FXS port (with the fax machine plugged into the FXS port), and none of the other lines routed in this way (the other lines would go to the attendant).
Then I need if we send a fax, we need for when the fax machine opens the line for it to automatically be routed to the fax line. Then the fax machine can dial and send faxes. I think we send few enough faxes that we're not too worried about what happens if we try to send a fax and the phone is busy, but it would be nice to be able to look at a phone and tell if the fax line is in use.
I don't really care if the Fax line is part of a shaed FXO, it can be a seperate choice on the phone for making outgoing calls (one of the buttons next to the screen or something).
Is all this possible? Is it only possible in PBX style or is mixed ok?

FXO not giving the dial tone

Hi,
We have FXO ports on the router and SIP trunk towards ITSP. People used to dial into FXO and get the dial tone to callout using the SIP trunk. But after the upgrade of the IOS, this functionality has stopped. Now if you call the FXO, you get busy tone. The IOS has been upgraded from 12.4(11) to 15.1(M4) to basically its a big leap.
I strongly believe that we need to make some configuration to make it work like before.
Please advise.
Attachached are the logs from the "debug vpm all"

[+] for Calro
Here is the new feature complied in 15.X release
http://www.cisco.com/en/US/docs/ios/15_1/release/notes/151-2TNEWF.html
Toll Fraud Prevention
In Cisco IOS Release 15.1(2)T, the Toll Fraud Prevention feature is supported as below:
•Source IP address authentication is enabled on incoming IPv4 H323/ or SIP trunk calls. The source IP address of any incoming IPv4 H323 or SIP trunk calls will be authenticated based on:
–Manually configured IP address trusted list.
–VoIP dial-peer session target (the state of a VoIP dial-peer must be in "Operation State = UP")
Incoming IPv4 H323 or SIP trunk calls will be rejected if the authentication fails and the default cause-code call-reject (21) disconnects the call.
Execute the show ip address trusted list command to display IP address trusted data and a list of valid source IP addresses. The default behavior can be disabled as shown in the example below:
voice service voip
no ip address trusted authenticate
•Secondary dial-tone is disabled for a call initiated from a FXO port. No secondary dial-tone causes the outgoing call setup to fail if the called number is NULL. The default behavior can be disabled as shown below:
voice-port
secondary dialtone
•Direct-inward-dial is enabled to prevent the toll fraud for incoming ISDN calls. Two-stage dialing is disabled for incoming ISDN calls by default. The incoming called number will then be used for outgoing call setup. The default behavior can be disabled as shown in the example below:
voice service pots
no direct-inward-dial isdn
For more information, see the Cisco Unified Communications Manager Express System Administrator Guide at the following URL:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm.
html
Br,
Nadeem
Please rate all useful post.

Issue connecting SPA3102 FXO port to paging interface

Hello,
I was referred to this community after a call to small business support, we were unable to come to a solution.
SPA3102 v5.2.13 FW
Viking Electronics PI-1 Paging Interface
We were told from Viking that the interface only supports an FXO port, all the connections were verified with their support and we're able to page by connecting an analog phone straight to the interface. Plugging into the FXS port on the SPA3102 also results in a dial tone playing over the paging system.
All settings on the SPA3102 are default minus SIP registration information, we see the unit as registered within the web interface as well as within our Asterisk CLI.
Trying to ring the extension results in constant ringing if voicemail is disabled, and straight to voicemail if enabled. Within the Asterisk CLI we see a SIP response 503 "Service Unavailable."
I can get the SPA3102 to respond by lowering the "Line In Use Voltage" down from its default value of 30. However all I hear is a hum and I'm not able to speak over the paging system.
Any insight into this issue would be greatly appreciated, we believe it's just settings on the SPA3102 unit itself that need to be changed.
Thanks!

Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE. The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.
I would try the following:
Viking Pl-1:
Talk Battery Switch (PT): On
Audio In: Cabled to FXO port of SPA3102
SPA3102:
Interface Analog Phone attached to SPA3102:
Line 1 Tab
Line Enable: YES
Dial Plan: (S0<:@gw0>)
PSTN Line Tab
Line Enable: Yes
VoIP-to-PSTN Gateway Enable: Yes
One Stage Dialing: Yes
Line 1 VoIP Caller DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: xx
The Line In Use Voltage needs to be lower than the Talk Battery voltage supplied by the Viking Pl-1. Usually it is set about half way between the on-hook and off-hook voltage level. Measure the FXO (PSTN Line) on-hook voltage by reading it on the SPA3102 INFO Tab. Set the Line In Use Voltage substantially below the on-hook talk voltage.
If you lift the phone with the above settings you should be connected to the paging system.
Interface to an asterisk pbx system:
PSTN Line Tab
Setup Registration on PSTN Line Tab to Asterisk system.
Register: Yes
Proxy: xxx
UserID: xxx
Password: xxx
VoIP-to-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: None
One Stage Dialing: Yes
VoIP Caller Default DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: same comments as above
If you call the extension on the asterisk PBX you should be attached to the Viking unit.

How can I pass a # sign to my MGCP FXO port?

CUCM 7.1.5 and a 2951 with IOS 15
Goal: I need to be able to dial 50# and have that passed out the FXO port, including the # symbol.
**I cannot use and FXS card or ATA because the intercom system cannot "answer" the call. It stays off-hook at all times.
I have an MGCP-controlled FXO port tied to an intercom system. All I need is for the FXO port to pick up and pass "50#" down the line to the paging system. No matter what I do, I cannot get CUCM to pass the # to the gateway, only the "50". So the user then has to manually hit # after connecting to the paging system. This is for emergencies so we really don't want to require them to remember this. I have verified with CCAPI debugs that the called number does not contain a # when it arrives at the gateway.

There is a CM service parameter that controls whether CUCM will leave or strip the "#" from the called party string. By default this parameter is set to "true". Which means the "#" is stripped. Go to CM service parameters and search for 'strip'.
Note that this is a system-wide setting and could affect other call flows. You want to pay particular attention to any route patterns (or translations) where you rely/allow inter-digit timeout.
HTH.
-Bill (@ucguerrilla)
http://ucguerrilla.com

SPA8000 Problem with outgoing calls from trunk

Hi all,
I have SPA8000 with the latest firmware 6.1.12.
Tried the first time to configure the trunk groups. With incoming calls it caused no problems. But with the outgoing calls cannot solve the problem:
When dialing, the SPA8000 try found number to itself.
Port 8 in trunk group 1.
10.120.0.67 - ip of SPA8000.
172.17.1.1 - ip of PBX
302 - number of trunk.
444 - other number on PBX.
INVITE sip:[email protected]:6060 SIP/2.0
Via: SIP/2.0/UDP 10.120.0.67:5361;branch=z9hG4bK-e9bbff35
From: <sip:[email protected]>;tag=1e668866b76f0b0do1
To: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]:6060>;screen=yes;party=calling
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:5361>
Expires: 240
User-Agent: Linksys/SPA8000-6.1.12
Allow-Events: talk, hold, conference
Content-Length: 202
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 13756 13756 IN IP4 10.120.0.67
s=-
c=IN IP4 10.120.0.67
t=0 0
m=audio 19469 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
SIP/2.0 404 Not Found
To: <sip:[email protected]>;tag=63d33124-0
From: <sip:[email protected]>;tag=1e668866b76f0b0do1
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.120.0.67:5361;branch=z9hG4bK-e9bbff35
Server: Linksys/SPA8000-6.1.12
Allow-Events: talk, hold, conference
Content-Length: 0
What is wrong? What must I do to outgoing call?
Screenshots of configuration:

Hey!
Thank u soo much ana_bidi! It worked and my phone is back to life
I even sent an email to Nokia and I got the weirdest reply ever! They were telling me to check if my sim was inserted correctly I was like whaaaat? And I did mention to them that I can receive calls and send/recieve SMS! I seriously don't know what made them suggest that thing
Anyways thanks again for the solution!
Have a great day!
Walid Shaaya

UC520 SIP trunk unable to make outgoing calls, incoming calls are ok

I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843CE50C
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 0777777777 <- main sip number
Called Number            : 0888888888 <- called number
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name         :
001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 3
Disconnect Cause (SIP)   : 200

Hi Emil,
I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
but
UC_520#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl

VCS-E unable to make outgoing call with search rule not found error

Hi
I deploy telepresence device registered to CUCM and connect with VCS-C and VCS-E. The issue now is I am not able to make outgoing call and my search rule show not found, request time out.
I use DNS zone to make the outgoing call and my VCS-E is able to resolve DNS SRV record. However search rule result show not found.
Any advise are appreciated.

Hi
I managed to make a call to address that I am trying to reach. But it supposed to come up with PIN number to enter instead I saw the message that no incoming video.
Can you please advise what could be the issue.
tvcs: Event="Call Connected" Service="SIP" Src-ip="10.84.83.101" Src-port="27224" Src-alias-type="SIP" Src-alias="sip:[email protected]" Dst-alias-type="SIP" Dst-alias="sip:[email protected]" Call-serial-number="24a63c27-42b4-4950-ba02-6a548b279a09" Tag="f060faef-441f-47a2-9c75-e1b0623b5e14" Protocol="TCP" Call-routed="YES" Level="1" UTCTime="2015-02-25 14:38:32,604"

Limit outgoing calls?

I have rented my basement to a family that has abused the privilege of using my phone line, and they won't stop! Can I have a password added somehow for outgoing calls? Is there another way to stop this (other than removing the phone line which is too permanent for me, or cancelling the number)? Any suggestions would be appreciated.
Thanks,
Ann

CALL GATENote: This feature is available in DC, DE, MD, NJ, PA, VA and WV.
You now have greater control over the outgoing calls dialed from your home phone.
Call Gate offers a range of options. Decide how it can work best for you, and then
program your phone. Anytime you want to change your Call Gate features, you may
access the service from any touch-tone phone.
It pretty much lets you allow and restrict certain numbers one can call. Hopefully that is a step in the right direction! If you need more information on phone features, I would call our Customer Support line at 1-800-688-2880.
Message Edited by CharlesH on 12-27-2008 10:24 AM
Message Edited by CharlesH on 12-27-2008 10:25 AM

SRST - Outgoing calls.

Hi,
I have some questions regarding outgoing calls when WAN outages occurs.
We have a 2801 router with SRST 3.3 connected to PSTN through 2x VIC2-2BRI.
I am not sure how to get the outgoing calls to work, i want everyone to hit the 0 to get a secondary dial-tone if they want to make calls through PSTN.
I guess i need dial-peers to make this work? Should i setup one dial-peer for each voice port? with a destination pattern 0.T?
I have got the incoming calls to work, here is a small post from the config.
voice translation-rule 1
rule 1 /2992/ /982/
rule 2 /^.*\(...\)/ /\1/
voice translation-profile SRST-INCOMING
translate called 1
call-manager-fallback
secondary-dialtone 0
max-conferences 4 gain -6
ip source-address 10.20.12.10 port 2000
max-ephones 12
max-dn 15
system message primary Fallback, only ext. calls.
keepalive 60
translate called 1
translation-profile incoming SRST-INCOMING
time-format 24
date-format yy-dd-mm
How do the dial-peers work? I would be glad if someone help me back on the right track again.

If this was a MGCP gateway under normal wan Up/Up condition, you will need following commands to fallback to H323 mode. Thi swill be in addition to the dial-peers suggested by Brandon.
call application alternate default
or
service alternate default.
ccm-manager fallback-mgcp
And you may use the same dial peers that are used for mgcp.
DIAL-PEER VOICE 999101 POTS
PORT 1/0/0
SERVICE MGCPAPP
destination-pattern 9T
incoming called-number .T
HTH
Sankar.
PS: please remember to rate all posts!

Restrict FXO ports on the Outgoing Calls.

Similar Messages

Maybe you are looking for