05-24-2016 09:20 AM - edited 03-17-2019 07:01 AM
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
-----------------------------------------------------------------------------------------------------------------------------------------------------------
05-24-2016 09:43 AM
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
05-24-2016 09:57 AM
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.
05-24-2016 10:15 AM
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
05-24-2016 03:09 PM
05-26-2016 06:56 PM
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.
05-26-2016 07:02 PM
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.
05-24-2016 04:17 PM
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
05-26-2016 06:26 PM
I have tried prefixing a 1 to the called number and tried changing the plan and type to every combination and nothing works.
05-24-2016 04:33 PM
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
05-26-2016 06:33 PM
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.
05-27-2016 02:28 AM
Yes, This looks like configuration problem from provider end. Open a ticket with them see what they say.
Best Regards,
Sudheer Shenoy
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