Solved: UCM 10.5 - 2821 Inbound Sip Calls fail - Outbound work

Carl Fitzsimmons · ‎08-30-2022

All was good until a major power failure. No one was onsite and battery backups went down due to power outage length. When system came back up outbound from IP Phones worked theough SIP Trunk. However inbound calls all fail.

Flow is through ASA5506 (IP:Port to 2821) - 2821 - Voice Dial Peer - UCM10.5 - CTI RT PT - CUC10.5 - System Call Handler

When I call the Call Handler from an internal IP PHONE I hear the CUC recorded System Call Handler message

Checking 2821 DialPeer I see incoming dialpeer match however UCM is never contacted and therfore CUC System Call Handler does not start

Here is a piece of debug voip dialpeer inout when the number is called. This log goes on until the call fails and the Calling - Called Number eventually become blank

010694: Aug 31 02:58:56.424: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
010695: Aug 31 02:58:56.424: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
010696: Aug 31 02:58:56.424: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
010697: Aug 31 02:58:56.424: //-1/B035244E897A/DPM/dpAssociateIncomingPeerCore:
Calling Number=, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
010698: Aug 31 02:58:56.424: //-1/B035244E897A/DPM/dpAssociateIncomingPeerCore:
Result=NO_MATCH(-1) After All Match Rules Attempt
010699: Aug 31 02:58:56.424: //-1/B035244E897A/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
010700: Aug 31 02:58:58.160: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=5856725663, Called Number=5856725663, Peer Info Type=DIALPEER_INFO_SPEECH
010701: Aug 31 02:58:58.160: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=5856725663
010702: Aug 31 02:58:58.160: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
010703: Aug 31 02:58:58.160: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring=5856725663, saf_enabled=1, saf_dndb_lookup=1, dp_result=0
010704: Aug 31 02:58:58.160: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=92
2: Dial-peer Tag=10

Here are the dialpeers

dial-peer voice 92 voip

description From SIP provider to CUCM for BIZ

destination-pattern 5856725663

session target ipv4:10.8.32.20:5060

incoming called-number .

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

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

no vad

!
dial-peer voice 91 voip
description From SIP provider to CUCM for other
destination-pattern 5853472412
session target ipv4:10.8.32.20:5060
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte cisco-rtp h245-alphanumeric
no vad
!
dial-peer voice 10 voip
description Outgoing dial-peer to Vitelity
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad

SIP-UA info

sip-ua
no remote-party-id
retry invite 2
retry register 2
retry options 1
timers connect 100
registrar dns:sip34.vitelity.net expires 3600
sip-server dns:sip34.vitelity.net
host-registrar

Voice and SIP info

voice service voip
ip address trusted list
ipv4 66.241.99.181 255.255.255.255
ipv4 64.2.142.189 255.255.255.255
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
no supplementary-service sip handle-replaces
redirect ip2ip
fax protocol pass-through g711ulaw
no fax-relay sg3-to-g3
h323
h450 h450-2 timeout T1 1000
h450 h450-3 timeout T1 1000
h225 timeout ntf 500
h225 display-ie ccm-compatible
h225 connect-passthru
h245 passthru all
modem passthrough nse codec g711ulaw
sip
registrar server expires max 600 min 60
early-offer forced
midcall-signaling passthru
no call service stop

Any thoughts on how to proceed are welcomed.

TechLvr · ‎09-01-2022

Can you add the following SIP profile to the CUBE and test again?

configure terminal
voice service voip
sip
sip-profiles inbound

voice class sip-profiles 100
response ANY sip-header Contact modify "10.8.36.2" "24.213.128.10"
request ANY sip-header Contact modify "10.8.36.2" "24.213.128.10"
response ANY sdp-header Audio-Connection-Info modify "10.8.36.2" "24.213.128.10"
request ANY sdp-header Audio-Connection-Info modify "10.8.36.2" "24.213.128.10"
response ANY sdp-header Connection-Info modify "10.8.36.2" "24.213.128.10"
request ANY sdp-header Connection-Info modify "10.8.36.2" "24.213.128.10"
response ANY sdp-header Session-Owner modify "10.8.36.2" "24.213.128.10"
request ANY sdp-header Session-Owner modify "10.8.36.2" "24.213.128.10"

dial-peer voice 110 voip
voice-class sip profiles 100 inbound

View solution in original post

b.winter · ‎08-30-2022

Could you provide the following info for a better understanding:
sh run
output of debugs:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

Also: what should be your inbound dial-peer to match the call leg for an inbound call? (I know, I see the config, but I wouldn't configure it like that and it's not very good configured anyway)

RezaNazari3290 · ‎08-31-2022

Carl,
Make sure the sip trunk to your CUBE is up in the CUCM. I know you said outbound calls work but they could be using a failover route.

Also, can you add the IP address of your CUCM to the CUBE’s “ip address trusted list” under the “voice service voip” settings?

If the issue still persists, share the output of the following debug commands and also your CUBE version info.
debug voice ccapi inout
debug ccsip messages

Carl Fitzsimmons · ‎08-31-2022

I think that has been accomplished by the following from the 2821

voice service voip
ip address trusted list
ipv4 66.241.99.181 255.255.255.255
ipv4 64.2.142.189 255.255.255.255
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
no supplementary-service sip handle-replaces
redirect ip2ip
fax protocol pass-through g711ulaw
no fax-relay sg3-to-g3

Is that not correct?

Roger Kallberg · ‎08-31-2022

Is not the IP of one of your CM nodes 10.8.32.20? That would be what I would figure from your shared configuration snippets anyway. If you have more than one CM node, and I sort of assume that your do, I would recommend you to add all of them with these commands.

voice service voip
ip address trusted list
  ipv4 10.8.32.20 ;CPE Subscriber
  ipv4 10.8.32.?? ;CPE Subscriber
  ipv4 10.8.32.?? ;CPE Subscriber

And so on for all your CPE nodes

Carl Fitzsimmons · ‎08-31-2022

I performed separate debug for this request.

Being new to CUBE I recognize PING in TRUNK indicates the connection to 2821 - UCM is up

however incoming calls are not going through

Vladimir Adutskevich · ‎08-31-2022

I would also suggest to use a pair of dial-peers for CUCM - incoming/outgoing and same for Telco SIP trunk. This is more secure and easy for tuning - two dial-peers per system. Then you can see which dial-peers are matched for incoming call and correct your settings if dial-peer is down or wrong dial-peer is matched.

Carl Fitzsimmons · ‎08-31-2022

From what we currently have as dial peers what do you suggest? As mentioned I am normally on the device side not the system side. I understand Dial-Peers and thought what we had was Incoming and Outgoing but in the debugs I see both Dial Peers 92 and 10 matched.

So if we have 2 DIDs that we want separate Dial Peers for and point them to UCM to handle as needed one as CTI RT PT and another as a phones DN what changes do I need to make verses what we have?

Carl Fitzsimmons · ‎08-31-2022

output of debugs included:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74

I have been on the phone-device-MAC side of UCM. CUBE is something I have heard of but have not worked on. Digging into 2821 I saw 12.4. After checking and verifying I had more than enough flash and ram I upgraded to c2800nm-adventerprisek9-mz.151-4.M1.bin, reloaded and verified that was the current version

SHO VER

Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 15.1(4)M12a, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2016 by Cisco Systems, Inc.
Compiled Tue 04-Oct-16 03:37 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

SCG-2821 uptime is 24 minutes
System returned to ROM by reload at 10:18:51 EDT Wed Aug 31 2022
System restarted at 10:20:45 EDT Wed Aug 31 2022
System image file is "flash:/c2800nm-advipservicesk9-mz.151-4.M12a.bin"
Last reload type: Normal Reload

Calls outbound still work - calls inbound still fail

I checked the SIP TRUNK and PING was set on Device - Device Settings - SIP Profile and checking TRUNK I see the TRUNK is up. Short time up is due to restart.

Checking by dialing the CTI RT PT and I still hear the message and all caller inputs function. This tells me CTI RT PT and CUC are okay. So it appears the piece is between the 2821 UCM not communicating.

b.winter · ‎08-31-2022

Are you using SIP between CUCM and the router or H.323?
In CUCM it seems you have a SIP trunk configured, but the dial-peers in the router are in H.323 ("session protocol sipv2" is not there).
I would change the dial-peer config as follows (it's not perfect, but a starting point):

dial-peer voice 200 voip
description ### From CUCM ###
session protocol sipv2
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 210 voip
description ### To CUCM ###
session protocol sipv2
destination-pattern 5853472412
session target ipv4:10.8.32.20:5060
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 211 voip
description ### To CUCM ###
session protocol sipv2
destination-pattern 5856725663
session target ipv4:10.8.32.20:5060
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 voip
description ### To SIP Provider Vitelity ###
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 110 voip
description ### From SIP Provider Vitelity ###
incoming called-number 5856725663
session protocol sipv2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 111 voip
description ### From SIP Provider Vitelity ###
incoming called-number 5853472412
session protocol sipv2
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad

Carl Fitzsimmons · ‎08-31-2022

First thanks to everyone who have provided guidance.

@b.winter That looks different. This worked for many years prior to that major power failure. Now I am in the forest for sure.

Ok - I deleted Dial-Peers. Established inbound and out bound failed. Then inserted your dial peers - thanks. Both inbound and outbound work.

Inbound 5856725663 DTMF is not passed. DTMF does not matter on 5853472412 as call will go to VM which is normal. CUC will allow keys during message play. I checked internally and DTMF works and will cycle twice. When calling 5856725663 using my mobile that message plays and then drops and does not receive DTMF either.

I provide a debug. I see Dial-Peers accepting and handing off, then passing to UCM then passing to CUC however STATE_DEAD occurs while message is playing.

RezaNazari3290 · ‎08-31-2022

Can you try adding the command below to your dial peers (incoming dial peer from ISP and outgoing dial peer to CUCM?

voice-class sip dtmf-relay force rtp-nte

For some reason, my responses disappear from the forum.

Carl Fitzsimmons · ‎08-31-2022

@RezaNazari3290 Thanks - I added that to FROM SIP provider and TO UCM. Same problem. Just in case I missed something

Here are the new Dial-Peers

dial-peer voice 200 voip
description ### From CUCM ###
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 210 voip
description ### To CUCM ###
destination-pattern 5853472412
session protocol sipv2
session target ipv4:10.8.32.20:5060
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 211 voip
description ### To CUCM ###
destination-pattern 5856725663
session protocol sipv2
session target ipv4:10.8.32.20:5060
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 100 voip
description ### To SIP Provider Vitelity ###
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 110 voip
description ### From SIP Provider Vitelity ###
session protocol sipv2
session target sip-server
incoming called-number 5856725663
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 111 voip
description ### From SIP Provider Vitelity ###
session protocol sipv2
session target sip-server
incoming called-number 5853472412
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad

Roger Kallberg · ‎08-31-2022

Not related to your problem with DTMF, but I would advise you to remove this line from your inbound CM dial peer, session target sip-server, ie remove that on dial peer 200. That is used on outbound dial peers, so it is irrelevant on that dial peer.

Another thing, you still have destination-pattern .T on your outbound dial peer towards your ITSP. As I wrote before this is not a very good way to match calls on a dial peer as it has the potential to match anything, including any specifically defined numbers like the two you have for sending calls to CM if they for any reason fails to route the call. What will happen then is that the call would make a match on this dial peer and try to send the inbound call back out to your service provider, this is what is known as a call routing loop and you should avoid that. How are you calling outside calls, can you please give an example on how you enter that on the phones and also explain the setup for route pattern(s) and if used for this type of call also translation pattern(s), so that we can give advice on how you could alter your configuration to not have a potential routing loop?

Also I meant to ask if you Cube is using separate interfaces for the setup with your service provider, ie one interface towards CM and another interface towards your service provider? If you do it is advisable to add bind statements to your respective dial peers so that the correct interface is used for each call case.

Carl Fitzsimmons · ‎09-01-2022

Thanks. I have removed the 'T' from dial peer destination patter.

CUBE is a good question. Mentioned earlier I am from the device side specializing in BAT and database conversions for small to large phone and UCM-CUC version migration and new site development. I actually thought this would be an easy piece to pick up as documentation existed as did functionality. So much for that.

Diving into the CUBE questions it appears CUBE is not running on the 2821. I deduced that by trying the following

SCG-2821(config)#voice service voip
SCG-2821(conf-voi-serv)#mode border-element
^
% Invalid input detected at '^' marker.

I have confirmed the IOS upgrade is installed

Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 15.1(4)M12a, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2016 by Cisco Systems, Inc.
Compiled Tue 04-Oct-16 03:37 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

SCG-2821 uptime is 23 hours, 40 minutes
System returned to ROM by reload at 10:18:51 EDT Wed Aug 31 2022
System restarted at 10:20:45 EDT Wed Aug 31 2022
System image file is "flash:/c2800nm-advipservicesk9-mz.151-4.M12a.bin"
Last reload type: Normal Reload

Which is the last version for the 2821 and has CUBE capability. I checked and there is no license info for CUBE. Then I noticed a comment on a Cisco blog that CUBE is not needed when termination is with ISP. So at this time I am in the weeds. If CUBE is required then that may very well be what the problem is. If true then how to proceed on an EOL 2821 needs to be defined. If it is true that when SIP is terminated with the ITSP (Vitelity) then we are okay. Which is true I am not 100% certain but I think that SIP is terminated directly with ITSP or otherwise I have no idea how this worked for so many years. Your thoughts?