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Incoming call is not working properly-VIC2 4FXO- 4 analog lines

Balaji Ezhumalai
Beginner
Beginner

Hello All

 

I have 4 PSTN line connections and it is connected to my router(2911) via VIC2 4FXO.

 

I have problem with incoming calls,

when only one line (ex. analog line1) is plugged in to the FXO I can get incoming calls,

but when I plug another 3 lines (3 analog lines) into the FXO, I can only receive the first incoming call, 

if I try to make call another PSTN number plugged to the FXO, it says that "your dialing is delayed please try again",

after this, i can not make any calls to my FXO, my FXO (my voice gateway-router 2911) is not receiving any calls from pstn network. All the lights in FXO become green keep lighting.

 

I think i have problem with my incoming voice peer or voip, but i can not figure it out.

 

The current voice configuration is attached below:

****************************

voice-port 0/0/0
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx 2050
caller-id enable
!
voice-port 0/0/1
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx 2050
!
voice-port 0/0/2
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx 2050
!
voice-port 0/0/3
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx 2050

!

!

!

dial-peer voice 1000 voip
incoming called-number .
voice-class codec 100
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fal
no vad
!
dial-peer voice 1100 pots
incoming called-number .
clid strip
direct-inward-dial
port 0/0/0
!

dial-peer voice 1200 voip
destination-pattern .T
session target ipv4:192.168.0.100
voice-class codec 100
voice-class h323 1
dtmf-relay h245-alphanumeric
fax rate disable
no vad

!

!

!

dial-peer voice 100 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/0
!
dial-peer voice 101 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/1
!
dial-peer voice 102 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/2
!
dial-peer voice 103 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/3
!

!

dial-peer voice 200 pots
destination-pattern .T
progress_ind alert enable 8
port 0/0/0
!

dial-peer voice 201 pots
destination-pattern .T
progress_ind alert enable 8
port 0/0/1
!
dial-peer voice 202 pots
destination-pattern .T
progress_ind alert enable 8
port 0/0/2
!
dial-peer voice 203 pots
destination-pattern .T
progress_ind alert enable 8
port 0/0/3
!

!

**********************************

 

The outgoing and inter communications works perfectly.

 

The picture is attached below

 

Please help and thanks in advance

3 ACCEPTED SOLUTIONS

Accepted Solutions

From the debug we can see that for your working call the inbound dial peer is 300 and the outbound is 1200. But for the none working the inbound dial peer is 101 and outbound is 100. So this loops back out to PSTN as it seems. This is very likely the cause of your issues as you have a loop in your configuration.

You need to check your dial peer configuration and figure out why it's not matched as per your preference. What is the phone number that you have in CM that should ring for inbound calls. Quite likely you need to modify the called number with a voice translation rule and also modify the destination pattern on dial peer 1200 to match what you have on the CM side. For details on how to setup voice translations please have a look at this document. Voice Translation Rules - Cisco



Response Signature


View solution in original post

I went back and looked at what you called the full configuration of your gateway and used that as a starting point for what possibly could work for you.

This should cater for your first objective, aka task 1.

voice service voip
 no allow-connections h323 to h323
 no allow-connections h323 to sip
 no allow-connections sip to h323
 no allow-connections sip to sip
! None of these are needed for what you use the gw for. These are for if the gw acts as an SBC, aka routes voip to voip call legs.
!
voice class codec 100
 no codec preference 1 g729r8
 no codec preference 2 g729br8 ! the "b" in the codec name means that it does VAD and as you turn off VAD on your dial peers there is no need to have it here.
 no codec preference 3 g711ulaw
 no codec preference 4 g711alaw
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
!

voice translation-rule 20
 rule 1 /^7222$/ /2010/
 rule 2 /^9097$/ /2011/
 rule 3 /^9096$/ /2012/
 rule 4 /^9095$/ /2013/
!
voice translation-profile PSTN-IN
 translate called 20

!
voice translation-rule 40
 rule 1 /^9\(.*\)/ /\1/
!
voice translation-profile PSTN-OUT
 translate called 40

trunk group FXO
 hunt-scheme round-robin both
 translation-profile incoming PSTN-IN
 translation-profile outgoing PSTN-OUT

voice-port 0/0/0
 trunk-group FXO
 cptone KR
 timeouts call-disconnect 2
 timeouts wait-release 2
 connection plar opx 7222
 caller-id enable
bearer-cap Speech ! voice-port 0/0/1 trunk-group FXO cptone KR timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 9097 caller-id enable
bearer-cap Speech ! voice-port 0/0/2 trunk-group FXO cptone KR timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 9096 caller-id enable
bearer-cap Speech ! voice-port 0/0/3 trunk-group FXO cptone KR timeouts call-disconnect 2 timeouts wait-release 2 connection plar opx 9095 caller-id enable
bearer-cap Speech ! voice class e164-pattern-map 1 description E164 Pattern Map for called number to CUCM e164 201[0-3] ! dial-peer voice 1000 voip description Inbound calls from CUCM incoming called-number . voice-class codec 100 fax rate 14400 bytes 48 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none dtmf-relay h245-alphanumeric no vad ! dial-peer voice 1100 pots description Inbound calls from PSTN incoming called-number . direct-inward-dial ! dial-peer voice 1200 voip description Outbound calls to CUCM subscribers no destination-pattern .T destination e164-pattern-map 1 session target ipv4:***.***.*.*** voice-class codec 100 voice-class h323 1 dtmf-relay h245-alphanumeric fax rate disable no vad ! dial-peer voice 100 pots description Outbound calls to PSTN destination-pattern 9T progress_ind alert enable 8 no port 0/0/0 trunkgroup FXO no digit-strip ! no dial-peer voice 101 pots ! no dial-peer voice 102 pots ! no dial-peer voice 103 pots ! no dial-peer voice 300 pots ! no dial-peer voice 301 pots ! no dial-peer voice 302 pots ! no dial-peer voice 303 pots

To cater for your second objective, aka task 2 where calls to any of the 4 phone numbers should ring 2010 you would need to do these modifications.

voice translation-rule 20
 rule 1 /^7222$/ /2010/
 rule 2 /^9097$/ /2010/
 rule 3 /^9096$/ /2010/
 rule 4 /^9095$/ /2010/
!
voice class e164-pattern-map 1
 description E164 Pattern Map for called number to CUCM
  no e164 201[0-3]
  e164 2010
!


Response Signature


View solution in original post

Please do not do this!

dial-peer voice 1200 voip
 destination-pattern .

You absolutely need to be more precise in your match statements on the dial peers, otherwise you run the risk of new loops, just as you did have before.

There is no magic in the operation of dial peers, they don't know anything about what you want them to do and for what direction you want to use them. That's on the person that puts in the configuration. As such you have to make sure that they match as per what you want and nothing else. Please reference the first two documents that I shared with you for details on the operation of dial peers.



Response Signature


View solution in original post

30 REPLIES 30

Nithin Eluvathingal
VIP Mentor VIP Mentor
VIP Mentor

2050, is  it a hunt Pilot  or an extension ? 

 

If you hear "your dialing is delayed please try again", I think this message is from ISP and you may also need to open a ticket with ISP.

 

after this, i can not make any calls to my FXO, my FXO (my voice gateway-router 2911) is not receiving any calls from pstn network. :- Check if the FXO  ports disconnect the calls. 

 

 

 



Response Signature


Hi Nithin,


2050 is a hunt pilot.

I will check how to open a ticket with the ISP.

I actually created a 4 hunt pilot for a 4 pstn analog line, but still the
problem exists.

But there is no problem with outgoing from cucm to pstn, it works perfectly.

I am still searching

The FXO port is not disconnecting the call, and the status led of port 0/0/0, 1 is green.

 

The voice call status says "1 active call found", but there are no active calls right now.

 

#show voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x8A6 1AD2 0x22A2CD9C 0/0/1 No DSP 9097 None 101/100
0x8A7 1AD2 0x3EDE99B8 0/0/0 No DSP *9097 None 100/101
1 active call found

FXO has a know issues of disconnection. 

 

 



Response Signature


Steve Landon
Beginner
Beginner

are these configured to hunt from the carrier?

Hi Steve,

Yes, it is configured in the hunt pilot

What Steve meant is if your carrier has configured on their end that calls should route between the 4 FXO connections? This isn’t something that you can control with configuration as it’s solely based on configuration on the carrier side.

Looking through your original post I wonder why you have dial peers that are used in the outbound direction that sends calls to CM and PSTN that both matches .T as the destination? In general you should try to be as specific as you can with dial peer matching so that you don’t end up creating a call loop, albeit not saying that you do have this issue, but with this configuration you run a real risk of having it.

To be clear, this is not related to your problem as such, just an observation on none appropriate configuration.



Response Signature


Hello Roger,

 

Thanks for your suggestion, I will reconfigure it properly.

 

I have just started learning this and I will do my best with all your help.

 

I have the following question;

 

1. My first PSTN line (ex. 00-000-1234) is connected to voice port 0/0/0  and the line group member is 2001

    My 2nd PSTN line (ex. 00-000-1235) is connected to voice port 0/0/1  and the line group member is 2002

    My 3rd PSTN line (ex. 00-000-1236) is connected to voice port 0/0/2  and the line group member is 2003

    My 4th PSTN line (ex. 00-000-1237) is connected to voice port 0/0/3  and the line group member is 2004

 

incoming calls form port 0/0/0 will be received by 2001, and ports 0/0/1, 0/0/2, 0/0/3 will be received by  2002 2003, 2004 respectively.

 

this works well when only one port is active (one analog cable plugged in to one port),

but, if I connect all the 4 lines to the to the other ports 0/0/1, 2, 3 I am facing the above mentioned problem.

 

like you said, I will check with my carrier.

 

but my question is, is this scenario will usually work? How to configure 4 analog lines to FXO?

Do i need to create separate dial-peer for incoming calls?

 

Please help

Please see the break down of your current dial peer configuration below.

*** This is your inbound dial peer for voip ***
dial-peer voice 1000 voip
 incoming called-number .
 voice-class codec 100
 fax rate 14400 bytes 48
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fal
 no vad
*** This is your inbound dial peer for pots ***
 dial-peer voice 1100 pots
 incoming called-number .
 clid strip
 direct-inward-dial
 port 0/0/0
*** This is your outbound dial peer for voip ***
dial-peer voice 1200 voip
 destination-pattern .T ! I would recommend you to use a more specific destination match than .T as you likely do not have all the worlds phone number represented in your CM ';-)'
 session target ipv4:192.168.0.100
 voice-class codec 100
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 fax rate disable
 no vad
*** This are your outbound dial peer for pots ***
dial-peer voice 100 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/0
!
dial-peer voice 101 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/1
!
dial-peer voice 102 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/2
!
dial-peer voice 103 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/3
*** This are also your outbound dial peer for pots *** 
*** These would be the once that can cause the loop as they match .T, ie the same as the match on dial peer 1200 *** *** Likely these would not be needed based on that I assume that you use 9 as the breakout code for PSTN calls *** dial-peer voice 200 pots destination-pattern .T progress_ind alert enable 8 port 0/0/0 ! dial-peer voice 201 pots destination-pattern .T progress_ind alert enable 8 port 0/0/1 ! dial-peer voice 202 pots destination-pattern .T progress_ind alert enable 8 port 0/0/2 ! dial-peer voice 203 pots destination-pattern .T progress_ind alert enable 8 port 0/0/3

On your question about inbound calls, it's been years ago since I worked with FXO, so it's not all that fresh in mind. Based on your configuration your sending all calls on all the ports inbound to 2050 by this configuration on the voice ports "connection plar opx 2050". If I understood your last message correctly you have 4 different telephone numbers on the FXOs, 2001 to 2004 on individual ports. When you make the inbound call to test your setup what number do you dial on the external phone that you make the test from?

For reference reading to learn more on this I would encourage you to have a look at this fantastic document In Depth Explanation of Cisco IOS and IOS-XE Call Routing - Cisco and also this one that is pretty good Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms - Cisco.



Response Signature


This is clearly explained and thanks you.

 

I have removed .T pots as you suggested, but still the problem exists.

 

Here is the full configuration of my router:

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

voice-card 0
dsp services dspfarm
!
!
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
ip address trusted list
ipv4 ***.***.**.* ***.***.***.*
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
!
voice class codec 100
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
voice class h323 1
h225 timeout tcp establish 2
!
!
!
!
voice-port 0/0/0
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 7222
caller-id enable
!
voice-port 0/0/1
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 9097
!
voice-port 0/0/2
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 9096
!
voice-port 0/0/3
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 9095
!
!
!
!
dial-peer voice 1000 voip
incoming called-number .
voice-class codec 100
fax rate 14400 bytes 48
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 1100 pots
incoming called-number .
clid strip
direct-inward-dial
port 0/0/0
!
dial-peer voice 1200 voip
destination-pattern .T
session target ipv4:***.***.*.***
voice-class codec 100
voice-class h323 1
dtmf-relay h245-alphanumeric
fax rate disable
no vad
!

dial-peer voice 100 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/0
!
dial-peer voice 101 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/1
!
dial-peer voice 102 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/2
!
dial-peer voice 103 pots
destination-pattern 9.T
progress_ind alert enable 8
port 0/0/3
!
!
!
!
dial-peer voice 300 pots
destination-pattern 8.T
progress_ind alert enable 8
port 0/0/0
!
dial-peer voice 301 pots
destination-pattern 8.T
progress_ind alert enable 8
port 0/0/1
!
dial-peer voice 302 pots
destination-pattern 8.T
progress_ind alert enable 8
port 0/0/2
!
dial-peer voice 303 pots
destination-pattern 8.T
progress_ind alert enable 8
port 0/0/3
!

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

 

 

************QUESTION**********

I have a question in my inbound dial peer:

Only one port (0/0/0) is configured in my inbound dial peer, but i use 4 analog lines,

do i need to create separate inbound dial-peer for other 3 analog lines connected to other three ports (0/0/1, 2, 3)????

Please help

*** This is your inbound dial peer for pots ***
 dial-peer voice 1100 pots
 incoming called-number .
 clid strip
 direct-inward-dial
 port 0/0/0

 

You would not need to have any port defined on the inbound dial peer. Port is used for the outbound direction.



Response Signature


I have removed the port 0/0/0 from inbound "dial-peer voice 1100 pots" and checked, but still its same, sir.

 

Like i said before, i got the 1st incoming call to one of my telephone number and i cut the call and tried another

telephone number but is says that "the number you have dialed is on another call"

 

There are no incoming and outgoing calls from the voice gateway but the voice port status light is green and

the voice call status is "1 active call found" as shown below:

 

#show voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x8A6 1AD2 0x22A2CD9C 0/0/1 No DSP 9097 None 101/100
0x8A7 1AD2 0x3EDE99B8 0/0/0 No DSP *9097 None 100/101
1 active call found

Those dial peers that you have in that output would indicate to me that your call might hairpin back to PSTN as dial peer 101 and 100 are your outbound dial peers to PSTN.

show voice call status
CallID CID ccVdb Port Slot/DSP:Ch Called # Codec MLPP Dial-peers
0x8A6 1AD2 0x22A2CD9C 0/0/1 No DSP 9097 None 101/100
0x8A7 1AD2 0x3EDE99B8 0/0/0 No DSP *9097 None 100/101
1 active call found

With the configuration shared previously this would have been the expected dial peers used, inbound from PSTN 1100 outbound to CM 1200.

Can you please do this?

  1. Turn on "debug voip ccapi inout"
  2. Make sure you have logg terminal set in configuration, to get a bigger log buffer if needed you could use this command "logging buffered 2000000"
  3. Turn on debug output to your terminal session by "term mon"
  4. Turn off the output by "term no mon"
  5. Collect the output by copy paste and save it in a text file
  6. Post the text file as an attached file to you answer


Response Signature


Mr. Roger, I really appreciate your response and thanks again.

 

I have turned on the "debug voip ccapi inout" and attached the "term mon" text file as per your instruction.

 

Please take a look.

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