External PSTN calls are not making it to CUE voicemail?

External PSTN calls are not being forwarded to my Unity Express Voicemail. Internall calling and forwarding to Voicemail is working like a charm. However, calls from the outside just busyout after so many rings and never make it to voicemail.
Can some one help point me in the right direction. I think it might be as simple as dial-peer problem.
Thanks in advance,
Curt

I'm assuming this is CME-integrated. Basically, if you have a PRI connection, the easiest thing to do is to do a 'debug isdn q931' and check out the called number. Or if you have an FXO connection to the PSTN, then you've got some kind of 'connection plar' under the voice port to give you the called number. Basically the called number must match a destination pattern under a voip dial-peer pointing to CUE (you can use 'debug ccsip message' and look for an INVITE message to see where it's going to). Next, it has to match a trigger ('show ccn trigger' in the CUE CLI) to envoke a particular application, such as voicemail. If you have trouble, what we'll need to see is the exact called number (either via the isdn q931 debug or connection plar configuration statement) and the configuration (show run) from the CME and the CUE.
Lastly, take a look at the following notice:
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_field_notice09186a008023cfe2.shtml
which might apply to your situation.

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Calls are not getting thru in Cisco voice GW for a particular Number

Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98396
                Exclusive, Channel 22
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
        Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
        Calling Party Number i = 0x2183, '8168911010'
                Plan:ISDN, Type:National
        Called Party Number i = 0x89, '18553808521'
                Plan:Private, Type:Unknown
        Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
        Channel ID i = 0xA98396
                Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
        Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad

Hi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson

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Hello,
I have implemented a scenerio between SAP AR -
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For SAP client is 110 for XI client is 100.
In sxmb_adm we have set SAP as application server and Xi as IS.
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"angadala" <[email protected]> wrote in
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External PSTN calls are not making it to CUE voicemail?

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