PSTN Simulation in CCIE voice lab

kethucbk

Hi all,

 

I am setting up the PSTN simulation for lab. The topology is :

IP Phone(2001) ---SCCP--CUCM---H323---GW1----E1 PRI----GW2---FXS--Analog Phone(6001)

 

My configuration :

 

GW1:

 

controller E1 0/2/0

 pri-group timeslots 1-12,16

 

interface Serial0/2/0:15

 no ip address

 encapsulation hdlc

 isdn switch-type primary-net5

 isdn incoming-voice voice

 no cdp enable

 

dial-peer voice 201 voip

Description Match Incoming call from CUCM 

voice-class h323 1

 incoming called-number .

 dtmf-relay cisco-rtp h245-alphanumeric h245-signal rtp-nte

 codec g711ulaw

 ip qos dscp cs3 signaling

 no vad

!

dial-peer voice 100 pots

description PSTN_Simulation

 destination-pattern 9T

 direct-inward-dial

 port 0/2/0:15

!

!

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

GW2:

controller E1 0/1/0

 clock source internal

 pri-group timeslots 1-12,16

interface Serial0/1/0:15

 no ip address

 encapsulation hdlc

 isdn switch-type primary-net5

 isdn protocol-emulate network

 isdn incoming-voice voice

 no cdp enable

dial-peer voice 100 pots

 destination-pattern 9T

 direct-inward-dial

 port 0/1/0:15

!         

dial-peer voice 6001 pots

 destination-pattern 6001

 port 0/3/0

!         

voice-port 0/3/0

 station-id name Thuc_Analog1

 station-id number 6001

!     

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

When I make call from the IP Phone to the analog phone, the output of debug isdn q931 is :

 

*Oct 28 02:46:50.663: %CONTROLLER-5-UPDOWN: Controller E1 0/1/0, changed state to down (AIS detected)

*Oct 28 02:46:50.663: ISDN Se0/1/0:15 Q931: L3_ShutDown: Shutting down ISDN Layer 3

*Oct 28 02:46:50.667: ISDN Se0/1/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2

*Oct 28 02:46:52.663: %LINK-3-UPDOWN: Interface Serial0/1/0:15, changed state to down

*Oct 28 02:47:41.663: %CONTROLLER-5-UPDOWN: Controller E1 0/1/0, changed state to up

*Oct 28 02:47:43.663: %LINK-3-UPDOWN: Interface Serial0/1/0:15, changed state to up

*Oct 28 02:50:45.791: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0080 

        Bearer Capability i = 0x8090A3 

                Standard = CCITT 

                Transfer Capability = Speech  

                Transfer Mode = Circuit 

                Transfer Rate = 64 kbit/s 

        Channel ID i = 0xA9838C 

                Exclusive, Channel 12 

        Calling Party Number i = 0x0081, '2001' 

                Plan:Unknown, Type:Unknown 

        Called Party Number i = 0x80, '6001' 

                Plan:Unknown, Type:Unknown

*Oct 28 02:50:45.791: ISDN Se0/1/0:15 Q931: Received SETUP  callref = 0x8080 callID = 0x0032 switch = primary-net5 interface = Network 

*Oct 28 02:50:45.799: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8080 

        Channel ID i = 0xA9838C 

                Exclusive, Channel 12

*Oct 28 02:50:45.803: ISDN Se0/1/0:15 Q931: TX -> PROGRESS pd = 8  callref = 0x8080 

        Progress Ind i = 0x8188 - In-band info or appropriate now available 

*Oct 28 02:50:45.855: ISDN Se0/1/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x0080 

        Cause i = 0x80AF - Resource unavailable, unspecified

*Oct 28 02:50:45.855: ISDN Se0/1/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x8080

*Oct 28 02:50:45.867: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x0080

 

Please give me an advise.

 

Thanks

Thuc

michael.thompson

do 2 things please.  On the receiving gateway, add "incoming called ." to dial-peer voice 100.  That will assure you're hitting THAT dial peer instead of dial peer 0.  Also, please attach a show hardware from the receiving gateway.  I'm questioning if it has enough DSPs.

Also, if this is for the voice lab.  Analog ports have been removed for quite some time now.

On Mon, Oct 28, 2013 at 10:12 PM, kethucbk <[email protected]> wrote:

Hi all,

 

I am setting up the PSTN simulation for lab. The topology is :

IP Phone(2001) ---SCCP--CUCM---H323---GW1----E1 PRI----GW2---FXS--Analog Phone(6001)

 

My configuration :

 

GW1:

 

controller E1 0/2/0

 pri-group timeslots 1-12,16

 

interface Serial0/2/0:15

 no ip address

 encapsulation hdlc

 isdn switch-type primary-net5

 isdn incoming-voice voice

 no cdp enable

 

dial-peer voice 201 voip

Description Match Incoming call from CUCM 

voice-class h323 1

 incoming called-number .

 dtmf-relay cisco-rtp h245-alphanumeric h245-signal rtp-nte

 codec g711ulaw

 ip qos dscp cs3 signaling

 no vad

!

dial-peer voice 100 pots

description PSTN_Simulation

 destination-pattern 9T

 direct-inward-dial

 port 0/2/0:15

!

!

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

GW2:

controller E1 0/1/0

 clock source internal

 pri-group timeslots 1-12,16

interface Serial0/1/0:15

 no ip address

 encapsulation hdlc

 isdn switch-type primary-net5

 isdn protocol-emulate network

 isdn incoming-voice voice

 no cdp enable

dial-peer voice 100 pots

 destination-pattern 9T

 direct-inward-dial

 port 0/1/0:15

!         

dial-peer voice 6001 pots

 destination-pattern 6001

 port 0/3/0

!         

voice-port 0/3/0

 station-id name Thuc_Analog1

 station-id number 6001

!     

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

When I make call from the IP Phone to the analog phone, the output of debug isdn q931 is :

 

*Oct 28 02:46:50.663: %CONTROLLER-5-UPDOWN: Controller E1 0/1/0, changed state to down (AIS detected)

*Oct 28 02:46:50.663: ISDN Se0/1/0:15 Q931: L3_ShutDown: Shutting down ISDN Layer 3

*Oct 28 02:46:50.667: ISDN Se0/1/0:15 Q931: Ux_DLRelInd: DL_REL_IND received from L2

*Oct 28 02:46:52.663: %LINK-3-UPDOWN: Interface Serial0/1/0:15, changed state to down

*Oct 28 02:47:41.663: %CONTROLLER-5-UPDOWN: Controller E1 0/1/0, changed state to up

*Oct 28 02:47:43.663: %LINK-3-UPDOWN: Interface Serial0/1/0:15, changed state to up

*Oct 28 02:50:45.791: ISDN Se0/1/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0080 

        Bearer Capability i = 0x8090A3 

                Standard = CCITT 

                Transfer Capability = Speech  

                Transfer Mode = Circuit 

                Transfer Rate = 64 kbit/s 

        Channel ID i = 0xA9838C 

                Exclusive, Channel 12 

        Calling Party Number i = 0x0081, '2001' 

                Plan:Unknown, Type:Unknown 

        Called Party Number i = 0x80, '6001' 

                Plan:Unknown, Type:Unknown

*Oct 28 02:50:45.791: ISDN Se0/1/0:15 Q931: Received SETUP  callref = 0x8080 callID = 0x0032 switch = primary-net5 interface = Network 

*Oct 28 02:50:45.799: ISDN Se0/1/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8080 

        Channel ID i = 0xA9838C 

                Exclusive, Channel 12

*Oct 28 02:50:45.803: ISDN Se0/1/0:15 Q931: TX -> PROGRESS pd = 8  callref = 0x8080 

        Progress Ind i = 0x8188 - In-band info or appropriate now available 

*Oct 28 02:50:45.855: ISDN Se0/1/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x0080 

        Cause i = 0x80AF - Resource unavailable, unspecified

*Oct 28 02:50:45.855: ISDN Se0/1/0:15 Q931: TX -> RELEASE pd = 8  callref = 0x8080

*Oct 28 02:50:45.867: ISDN Se0/1/0:15 Q931: RX <- RELEASE_COMP pd = 8  callref = 0x0080

 

Please give me an advise.

 

Thanks

Thuc

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RichardJSimmons

From the looks of it you are doing a back-to-back E1, having you configured a clocking source  for this?

Also can you provide the following outputs on both sides

show controllers e1 status

kethucbk

Hi all,

After checking, I find out that the codec is missmatched between IP Phone and GW.

 

Thanks all,

Thuc