"No matching outgoing dial-peer" on ISR 4K for outgoing calls to PSTN

[Liv_Liv]

Hi, I am testing the following setup: 

CUCM -- SIP trunk -- ISR 4331 -- PRI

Like I said this is a test environment so the PRI are not yet plugged in the NIM E1 cards. So the inbound/outbound dial-peers tied to the PSTN are down by now. 

So Im testing outbound calls from CUCM (destinations starting with 999 and 024) and Im getting an 404 "No matching outgoing dial-peer" error message from the gateway, but I don´t see the reason why. Im starting to suspect this is due to the PSTN dial-peers are currently down but from my perspective I should get a dial-peer match and a 5XX Service Unavailable message instead. 

Here is my config snippet: 

Spoiler

!
card type e1 0 1
!
!
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
!
!
voice class uri CUCMs sip
host ipv4:XX.XX.XX.4
host ipv4:XX.XX.XX.1
host ipv4:XX.XX.XX.2
host ipv4:XX.XX.XX.3
host ipv4:XX.XX.XX.5
!
!
!
voice class server-group 10000
ipv4 XX.XX.XX.4 port 5060 preference 1
ipv4 XX.XX.XX.1 port 5060 preference 2
ipv4 XX.XX.XX.2 port 5060 preference 3
ipv4 XX.XX.XX.3 port 5060 preference 4
ipv4 XX.XX.XX.5 port 5060 preference 5
description *CUCMs*
!
voice class sip-options-keepalive 200
down-interval 60
up-interval 61
transport tcp
!
!
!
!
voice translation-rule 200
rule 1 /^1024/ /024/
rule 2 /^1999/ /999/
!
!
voice translation-profile REMOVES_1_CALLED
translate called 200
!
!
!
!
voice-card 0/1
dsp services dspfarm
no watchdog
!
controller E1 0/1/0
framing no-crc4
pri-group timeslots 1-31
!
interface Serial0/1/0:15
no ip address
encapsulation hdlc
no cdp enable
isdn switch-type primary-net5
isdn incoming-voice voice
!
dial-peer voice 100 voip
description *Incoming from CUCMs*
session protocol sipv2
incoming uri via CUCMs
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 200 voip
description *Outgoing to CUCMs*
translation-profile outgoing REMOVES_1_CALLED
destination-pattern 1T
session protocol sipv2
session server-group 10000
voice-class sip options-keepalive profile 200
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
!
dial-peer voice 300 pots
description *Incoming from PSTN - Central*
incoming called-number 024......$
port 0/1/0:15
prefix 1024
!
dial-peer voice 301 pots
description *Incoming from PSTN - Corp mob*
incoming called-number 024.....$
port 0/1/0:15
prefix 1024
!
dial-peer voice 400 pots
description *Outgoing to PSTN - Noncorp mob*
destination-pattern 999.........$
port 0/1/0:15
prefix 999
!
dial-peer voice 401 pots
description *Outgoing to PSTN - Corp mob*
destination-pattern 024.....$
port 0/1/0:15
prefix 024
!

And this is the output of debug voip ccapi inout & debug ccsip messages: 

Spoiler

*Sep 21 13:09:09.186: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:999677777777@XX.XX.XX.15:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.5:5060;branch=z9hG4bK35b7c4dd73a4e
From: "333 Name Surname" <sip:333@XX.XX.XX.5>;tag=4194162~0316432d-bda5-4ef0-a846-8b9c084232c0-98253025
To: <sip:999677777777@XX.XX.XX.15>
Date: Tue, 21 Sep 2021 13:10:09 GMT
Call-ID: 3ddbd400-1491d9b1-2fdab-5ff020a@XX.XX.XX.5
Supported: timer,resource-priority,replaces
Min-SE: 180
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:XX.XX.XX.5:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Cisco-Guid: 1037816832-0000065536-0000009712-0100598282
Session-Expires: 1800
P-Asserted-Identity: "333 Name Surname" <sip:333@XX.XX.XX.5>
Remote-Party-ID: "333 Name Surname" <sip:333@XX.XX.XX.5>;party=calling;screen=yes;privacy=off
Contact: <sip:333@XX.XX.XX.5:5060>
Max-Forwards: 70
Content-Length: 0

*Sep 21 13:09:09.189: //-1/3DDBD4000000/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=333
----- ccCallInfo IE subfields -----
cisco-ani=333
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=999677777777
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFFFFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0

*Sep 21 13:09:09.190: //-1/3DDBD4000000/CCAPI/cc_api_call_setup_ind_common:
Interface=0x7F84596086F8, Call Info(
Calling Number=333,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=999677777777(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=100, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=10123
*Sep 21 13:09:09.190: //-1/3DDBD4000000/CCAPI/ccCheckClipClir:
In: Calling Number=333(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Sep 21 13:09:09.190: //-1/3DDBD4000000/CCAPI/ccCheckClipClir:
Out: Calling Number=333(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Sep 21 13:09:09.190: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Sep 21 13:09:09.190: :cc_get_feature_vsa malloc success
*Sep 21 13:09:09.190: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Sep 21 13:09:09.190: cc_get_feature_vsa count is 1
*Sep 21 13:09:09.190: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*Sep 21 13:09:09.190: :FEATURE_VSA attributes are: feature_name:0,feature_time:140206458329420,feature_id:9
*Sep 21 13:09:09.190: //10123/3DDBD4000000/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=333(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=999677777777(TON=Unknown, NPI=Unknown))
*Sep 21 13:09:09.191: //10123/3DDBD4000000/CCAPI/cc_process_call_setup_ind:
Event=0x7F845C2705A8
*Sep 21 13:09:09.191: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
Try with the demoted called number 999677777777
*Sep 21 13:09:09.191: //10123/3DDBD4000000/CCAPI/ccCallSetContext:
Context=0x7F845D5EC260
*Sep 21 13:09:09.191: //10123/3DDBD4000000/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 10123 with tag 100 to app "_ManagedAppProcess_Default"
*Sep 21 13:09:09.192: //10123/3DDBD4000000/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Sep 21 13:09:09.192: //10123/3DDBD4000000/CCAPI/ccCallDisconnect:
Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Sep 21 13:09:09.193: //10123/3DDBD4000000/CCAPI/ccCallDisconnect:
Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
*Sep 21 13:09:09.194: //10123/3DDBD4000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.XX.XX.5:5060;branch=z9hG4bK35b7c4dd73a4e
From: "333 Name Surname" <sip:333@XX.XX.XX.5>;tag=4194162~0316432d-bda5-4ef0-a846-8b9c084232c0-98253025
To: <sip:999677777777@XX.XX.XX.15>
Date: Tue, 21 Sep 2021 13:09:09 GMT
Call-ID: 3ddbd400-1491d9b1-2fdab-5ff020a@XX.XX.XX.5
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-17.3.4a
Session-ID: 00000000000000000000000000000000;remote=510dd16b411555798a9ba84eee12b465
Content-Length: 0

*Sep 21 13:09:09.194: //10123/3DDBD4000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.XX.XX.5:5060;branch=z9hG4bK35b7c4dd73a4e
From: "333 Name Surname" <sip:333@XX.XX.XX.5>;tag=4194162~0316432d-bda5-4ef0-a846-8b9c084232c0-98253025
To: <sip:999677777777@XX.XX.XX.15>;tag=53AEE2D-8EE
Date: Tue, 21 Sep 2021 13:09:09 GMT
Call-ID: 3ddbd400-1491d9b1-2fdab-5ff020a@XX.XX.XX.5
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 399 XX.XX.XX.15 "No matching outgoing dial-peer"
Server: Cisco-SIPGateway/IOS-17.3.4a
Reason: Q.850;cause=1
Session-ID: 510dd16b411555798a9ba84eee12b465;remote=3a508c47794256d18a4b3784762db15c
Content-Length: 0

*Sep 21 13:09:09.196: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:999677777777@XX.XX.XX.15:5060 SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.5:5060;branch=z9hG4bK35b7c4dd73a4e
From: "333 Name Surname" <sip:333@XX.XX.XX.5>;tag=4194162~0316432d-bda5-4ef0-a846-8b9c084232c0-98253025
To: <sip:999677777777@XX.XX.XX.15>;tag=53AEE2D-8EE
Date: Tue, 21 Sep 2021 13:10:09 GMT
Call-ID: 3ddbd400-1491d9b1-2fdab-5ff020a@XX.XX.XX.5
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

*Sep 21 13:09:09.197: //10123/3DDBD4000000/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x7F84596086F8, Tag=0x0, Call Id=10123,
Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
*Sep 21 13:09:09.197: //10123/3DDBD4000000/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Sep 21 13:09:09.197: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*Sep 21 13:09:09.197: :cc_free_feature_vsa freeing 7F845C24D940
*Sep 21 13:09:09.197: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

And the dial-peer current status: 

Spoiler

dial-peer hunt 0
AD PRE PASS SESS-SER-GRP\ OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT KEEPALIVE VRF
100 voip up up 0 syst NA
200 voip up up 1T 0 syst SESS-SVR-GRP: 10000 active NA
300 pots up up 1024 0 down 0/1/0:15 NA
301 pots up up 1024 0 down 0/1/0:15 NA
400 pots up up 999 999.........$ 0 down 0/1/0:15 NA
401 pots up up 024 024.....$ 0 down 0/1/0:15 NA
For server-grp details please execute command:show voice class server-group <tag_id>
To see complete session target for ipv6 use 'sh running-config | section dial-peer <tag>

Am I missing something or is it just that the error is because my outbound PSTN dial-peers are down?

Thanks in advance !!

[Roger Kallberg · Accepted solution]

It is indeed because the outbound dial peer is in a down state. The call routing logic in the router checks the state of the dial peer before it can be selected as an outbound dial peer.

[Maren Mahoney · Accepted solution]

To extend what @Roger Kallberg said, a router will ignore dial-peers where the target is unavailable. This includes POTS dial-peers where the port is down or unavailable as well as VoIP dial-peers where the session target is not in the IP routing table.

Maren

[Roger Kallberg]

@Liv_Liv One additional thing for VoIP dial peers where SIP is used. These can also be in the down state when SIP option ping is used to check the destination availability. If it comes back as unavailable the dial peer is marked as down.

[Liv_Liv]

thank you both @Roger Kallberg and @Maren Mahoney  for confirming what I was thinking. So I ended up having a knowledge/concept issue rather than a real config issue. I thought that the call routing logic in a gateway checks potential matches before checking the actual state of the dial-peers, when in fact seems to be just the opposite way. I will now mark the post as solved. Again thanks for your valuable contribution !!!