Thank you for your response..

LPT Administrator · ‎05-24-2016

Current phone setup:

(old system)

PSTN -> ISDN PRI -> Cisco 2821 router -> Cisco CallManager -> Phones

(new system)

PSTN -> SIP Trunk -> AT&T-provided router -> Cisco 2921 router (CUBE) -> CallManager -> Phones

Some of our blocks of DID numbers were ported from the PRI to the SIP.

The new system has a very simple, basic configuration on the router and the CallManager.

I can make calls out of the SIP to any number.

I can make calls in to the SIP from my cell phone.

Calls from the PRI to the SIP fail with a fast busy tone, if the number being called is one of the ported numbers.

AT&T set up 2 test DID numbers on the SIP when it was installed. I can call these numbers from the PRI.

The dialed-number-analyzer on the old system shows the numbers should be routed.

I included the ISDN Q931 debug results from the old system when a call is made to a good number on the new system.

------------------------------------------DEBUG LOG-----------------------------------------------------------------------------------------------------------------

MSN-2821-1#terminal monitor

MSN-2821-1#debug isdn q931

debug isdn q931 is ON.

MSN-2821-1#

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0B33

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98394

Exclusive, Channel 20

Display i = '4621'

Calling Party Number i = 0x0081, '6082684621'

Plan:Unknown, Type:Unknown

Called Party Number i = 0xA1, '16082753696'

Plan:ISDN, Type:National

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8B33

Channel ID i = 0xA98394

Exclusive, Channel 20

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0190

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA18392

Preferred, Channel 18

Facility i = 0x9F8B0100A10F02012006072A8648CE1500040A0100

Protocol Profile = Networking Extensions

0xA10F02012006072A8648CE1500040A0100

Component = Invoke component

Invoke Id = 32

Operation = InformationFollowing (calling_name)

Name information in subsequent FACILITY message

Calling Party Number i = 0x2180, '6082684621'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '6082753696'

Plan:ISDN, Type:National

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8190

Cause i = 0x8081 - Unallocated/unassigned number

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x8B33

Cause i = 0x829F - Normal, unspecified

Progress Ind i = 0x8088 - In-band info or appropriate now available

.Apr 27 14:53:07: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x0B33

Cause i = 0x8090 - Normal call clearing

.Apr 27 14:53:07: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x8B33

.Apr 27 14:53:07: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x0B33

MSN-2821-1#no debug all

All possible debugging has been turned off

MSN-2821-1#exit

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

Suresh Hudda · ‎05-24-2016

Sorry , I thought it is incoming call :-(

Edit- Getting unallocated number error, is this (6082753696) dialable from mobile phone ? And also verify the ISDN plan and type as well with service provider.

Suresh

LPT Administrator · ‎05-24-2016

Thank you for your response...

Yes, 6082753696 can be called from my cell phone, just not from the PRI.

Can you give me a little more detail about what I should ask the service provider (AT&T)?

Outgoing and incoming calls on the PRI to any other numbers work fine...just calls to the numbers that were ported to the SIP.

I have talked to their help lines several times and they keep telling me they don't know what the problem is.

Suresh Hudda · ‎05-24-2016

Can you please tell me where pri exist in below dial plan ?

PSTN -> SIP Trunk -> AT&T-provided router -> Cisco 2921 router (CUBE) -> CallManager -> Phones

Seems call try to go out using channel 20 on pri and same call came back again on channel 18, do have proper dial peer configured on gateway ? can you attach debug voip ccapi inout also.

Suresh

LPT Administrator · ‎05-24-2016

The PRI and the SIP are completely isolated, separate systems. They are not connected to each other in any way.

Attached is the debug...hopefully it will help.

Dennis Mink · ‎05-26-2016

I would take it up with your PRI provider, sound like an unfinished number port to be honest.

Whoever provides the PRI, somehow can't route to AT&T sip service,

its obviously not your SIP cube, as you can dial into it via mobile and your test numbers work.

Please remember to rate useful posts, by clicking on the stars below.

LPT Administrator · ‎05-26-2016

The PRI and the SIP are both from AT&T.

I agree it appears to be on the telco's end. The keep denying it and telling me everything is ok so I want to be sure I am not missing something on my end.

Deepak Mehta · ‎05-24-2016

Does it work when you send prefix 1 with called party ,,,, in first example called number is '16082753696' and it seems first call was normal call clearing.

Failed call is where you are sending 10 digits? You may want to check this will Service provider how many digits they are expecting for national calls or try prefix 1 for each call .Also check plan and type info as per Suresh advise.thanks

Calling Party Number i = 0x2180, '6082684621'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '6082753696'

Plan:ISDN, Type:National

.Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8190

Cause i = 0x8081 - Unallocated/unassigned number

LPT Administrator · ‎05-26-2016

I have tried prefixing a 1 to the called number and tried changing the plan and type to every combination and nothing works.

Sudheer Shenoy · ‎05-24-2016

Hello,

In your debug can see that the service provider is sending your SETUP back to your PRI gateway, instead of sending it as INVITE to your CUBE.

Note "Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0B33" followed

by Apr 27 14:53:01: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0190 with different call ref number, however with same calling party number and called party number.

Best Regards,

Sudheer

LPT Administrator · ‎05-26-2016

What would cause that to happen? Is it a telco issue or a configuration problem?

Do they bounce the setup back if there is a problem with the sent call data?

I attached a q931 debug that shows 2 calls.

The call above the dashed line in the file is to one of the test numbers that AT&T gave me when the SIP trunk was installed. It is set up on the CallManager exactly like all the other numbers. As you can see it works fine.

The call below the dashed line is to one of the ported numbers from the PRI to the SIP... it fails every time.

Sudheer Shenoy · ‎05-27-2016

Yes, This looks like configuration problem from provider end. Open a ticket with them see what they say.

Best Regards,

Sudheer Shenoy

Cannot make outgoing calls to numbers from PRI ported to new SIP trunk