08-31-2022 08:00 PM
Hi,
i have a cube which will connect to ITSP with Public IP . Do I have to configure SIP-UA first?
because I debug ccsip message when making incoming call but debug doesn't enter CUBE (Term Monitor already)Topology :
CUCM-> SIPTrunk -> CUBE -> ITSP
I tried configuring SIP-UA with the username, password and realm provided by ITSP. But when I check show sip-ua register status , the result in the register column is "No"
is there something wrong with my config? how do I know if the cube sends a request to ITSP?
Solved! Go to Solution.
09-07-2022 02:42 AM - edited 09-07-2022 07:55 AM
Please create separate dial peers that are used for calls from your CM and from your service provider and on these use information in the VIA header to match them. Something along the line with this should work.
voice class e164-pattern-map 1
description E164 Pattern Map for called number to CUCM
e164 4...
!
!
voice class e164-pattern-map 2000
description E164 Pattern Map for called number to ITSP
e164 0T
!
voice class uri CUCM sip
host ipv4:IP ADD CUCM 1
host ipv4:IP ADD CUCM 2
! add as many line as you need, one for each CM that handles call processing
!
voice class uri PSTN sip
host ipv4:IP ADD ITSP
! add as many line as you need
!
voice class server-group 1
ipv4 IP ADD CUCM 1 preference 1
ipv4 IP ADD CUCM 2 preference 2
! add as many line as you need, one for each CM that is in the CMG use by the trunk for the SBC
description Inbound calls from ITSP to CUCM
huntstop 1 resp-code 404 to 404
!
voice class sip-options-keepalive 1
description Used for Server Group SIP OPTIONS PING
!
dial-peer voice 101 voip
description *** Inbound Dial-Peer From ITSP ***
no incoming called-number +6221303022xx
incoming uri via PSTN
!
dial-peer voice 10 voip
description *** Inbound Dial-Peer from CUCM ***
session protocol sipv2
incoming uri via CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 12 voip
description *** Outbound Dial-Peer for inbound calls from ITSP to CUCM ***
no destination-pattern 4...
no session target ipv4:IP ADD CUCM
session server-group 1
destination e164-pattern-map 1
voice-class sip options-keepalive profile 1
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 102 voip
description *** Outbound Dial-Peer to ITSP ***
no destination-pattern 0.
destination e164-pattern-map 2000
voice-class sip options-keepalive
08-31-2022 10:30 PM
Hi,
nobody can't tell you if you are right with your config or not, if you don't post any info like the config, debugs, ...
About SIP-UA: The current "standard" for SIP trunk registration is to use tenant configuration. Check out the related docs or search the forum, there have been a lot of discussions about SIP trunk registration.
Only a few global config commands are made in the SIP-UA.
09-01-2022 04:38 AM
Hi Winter,
How i can know , if the CUBE sent request to register SIP-UA ?
I just show sip service , show sip-ua connection udp , show sip-ua register status
I don't know if CUBE has sent a request to ITSP, how can I know?
Respectfully,
SH
09-01-2022 04:49 AM
debug ccsip non-call
With that, you should see REGISTER messages being sent by CUBE.
Which version are you using?
What do you see when using the command "show sip-ua register status"? (please make screenshot)
Do you have any documentation from the provider? Without it, it will be a guessing game.
09-04-2022 05:57 PM
Theres no log appear , when i run comman "debug ccsip non-call"
the term mon already running,
Version CUBE ? or Router ?
I have document from ITSP but only information number , username , password dan realm
09-04-2022 10:05 PM
The router it self has no version, it has a model. What version of IOS do you run on the router? Can you please share the configuration of your router and any information that you have got from your service provider? Please redact any sensitive information, like user name and password.
09-05-2022 07:15 AM
Hi Roger,
Thanks for reply, i hope you can help me
This my config VG
Version IOS -> 17.06.02
voice service voip
ip address trusted list
ipv4 IP Add CUCM
ipv4 IP Add VG
ipv4 IP Add ITSP
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
registrar server
early-offer forced
midcall-signaling passthru
call-route url
sip-profiles 1
!
voice class sip-profiles 1
request ANY sip-header SIP-Req-URI modify "ip address ITSP" "abc.def.com"
!
voice translation-rule 1
rule 1 /\(.*\)/ /+622130524000/
!
voice translation-rule 2
rule 1 /^\+\(.*\)/ /4556/
!
voice translation-rule 3
rule 1 /^7\(.*\)/ /\1/
!
!
voice translation-profile INCOMING
translate called 2
!
voice translation-profile OUTGOING
translate calling 1
!
dial-peer voice 101 voip
description *** Inbound Dialpeer From ITSP***
translation-profile incoming INCOMING
session protocol sipv2
incoming called-number +622130524....
voice-class codec 1
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 102 voip
description *** Outbound Dialpeer To ITSP ***
translation-profile outgoing OUTGOING
destination-pattern 7.
session protocol sipv2
session target ipv4:ip address ITSP
voice-class codec 1
voice-class sip profiles 1
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 12 voip
description description To CUCM Primary
destination-pattern 4...
session protocol sipv2
session target ipv4:ip add CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
!
dial-peer voice 13 voip
description description To CUCM 1st Backup
destination-pattern 4...
session protocol sipv2
session target ipv4:ip add cucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
!
dial-peer voice 14 voip
description description To CUCM 2nd Backup
destination-pattern 4...
session protocol sipv2
session target ipv4:ip add cucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
!
sip-ua
credentials username +6221305xxxxx@abc.def.com password 6 \eUA^Pc]H_[EbGCDU_FiBCJg]QKJNVPM^]QMUMSDAbAAB realm abc.def.com
credentials number +6221305xxxxx username +6221305xxxxx@abc.def.com password 6 fC]YbVUCeDG^DMONQcIiDcTD`NKGFR^da_PY`ZT_VFAAB realm abc.def.com:5060
credentials number +6221305xxxxxusername +6221305xxxxx@abc.def.com password 6 Y\_JH]FShOMNZ_P`XefKVa^YcEY^FiBMIKeb`J^fJhAAB realm abc.def.com:5060
authentication username +6221305xxxxx@abc.def.com password 6 RaIfOH_PdIbGQLfaFL_`AYPY^`dTCMCNU`GCGPJg\EAAB realm abc.def.com
retry invite 3
retry register 3
timers register 150
registrar ipv4:IP Address ITSP:5060 expires 120
connection-reuse
!
!
Information from ITSP :
Username ; +6221305xxxxx@abc.def.com
Password : XXXXXX
Realm : abc@def.com
09-05-2022 07:43 AM - edited 09-05-2022 07:48 AM
Try the following config. It uses tenant configuration, instead of SIP-UA.
I advise you check out the corresponding Cisco Docs, forum posts, ... to read and get used to that. SIP trunk registration with SIP-UA is "old" and using SIP tenant is the current "how-to" for SIP-trunk registration.
no sip-ua
!
voice class tenant 100
registrar ipv4:<IP of ITSP>:5060 expires 120
credentials number +6221305xxxxx username +6221305xxxxx@abc.def.com password <pwd> realm abc@def.com
no remote-party-id
timers buffer-invite 5000
timers dns registrar-cache ttl
sip-server ipv4:<IP of ITSP>
session refresh
no update-callerid
error-passthru
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
no pass-thru content custom-sdp
connection-reuse via-port
outbound-proxy ipv4:<IP of ITSP>
privacy-policy passthru
!
dial-peer voice 101 voip
voice-class sip tenant 100
no voice-class sip bind control source-interface GigabitEthernet0/0/0 --> already defined in the tenant
no voice-class sip bind media source-interface GigabitEthernet0/0/0 --> already defined in the tenant
!
dial-peer voice 102 voip
voice-class sip tenant 100
no voice-class sip bind control source-interface GigabitEthernet0/0/0 --> already defined in the tenant
no voice-class sip bind media source-interface GigabitEthernet0/0/0 --> already defined in the tenant
session target sip-server --> use sip-server defined in tenant
Also to clean up your config, assuming you are using this router only for the connection between CUCM and the ITSP:
voice service voip
no allow-connections h323 to h323 --> not needed
no allow-connections h323 to sip --> not needed
no allow-connections sip to h323 --> not needed
sip
no bind control source-interface GigabitEthernet0/0/0
no bind media source-interface GigabitEthernet0/0/0
no registrar server --> only needed, if you use SRST on the router
no call-route url
Before you apply the tenant config, turn on the debugs "debug ccsip non-call" and "debug ccsip messages" and collect the logs.
With "show sip-ua register status" you can check if the SIP trunk is registered or not, e.g.:
Tenant: 100
--------------------- Registrar-Index 1 ---------------------
Line peer expires(sec) reg survival P-Associ-URI
================================ ========= ============ === ======== ============
xxxxxxxxx -1 29 yes normal
09-06-2022 12:40 AM
Hi ,
This is result from debug CCSIP Message Incoming Call
id-tangg-vrtr-01#
*Sep 6 06:37:51.873: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:IP ADD VG:5060 SIP/2.0
Via: SIP/2.0/TCP IP ADD CUCM:5060;branch=z9hG4bK3e5bc2171d9d54
From: <sip:IP ADD CUCM>;tag=1058440017
To: <sip:IP ADD VG>
Date: Tue, 06 Sep 2022 06:42:26 GMT
Call-ID: 109c3a80-1ee14bb2-36bdc6-f3d965a1@IP ADD CUCM
User-Agent: Cisco-CUCM11.5
CSeq: 101 OPTIONS
Contact: <sip:IP ADD CUCM:5060;transport=tcp>
Max-Forwards: 0
Content-Length: 0
*Sep 6 06:37:51.874: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP IP ADD CUCM:5060;branch=z9hG4bK3e5bc2171d9d54
From: <sip:IP ADD CUCM>;tag=1058440017
To: <sip:IP ADD VG>;tag=85EC6076-2
Date: Tue, 06 Sep 2022 06:37:51 GMT
Call-ID: 109c3a80-1ee14bb2-36bdc6-f3d965a1@IP ADD CUCM
Server: Cisco-SIPGateway/IOS-17.6.2
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
id-tangg-vrtr-01#Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 385
v=0
o=CiscoSystemsSIP-GW-UserAgent 4801 2257 IN IP4 IP ADD VG
s=SIP Call
c=IN IP4 IP ADD VG
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 IP ADD VG
m=image 0 udptl t38
c=IN IP4 IP ADD VG
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
id-tangg-vrtr-01#u all
All possible debugging has been turned off
id-tangg-vrtr-01#
*Sep 6 06:37:53.505: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP IPADD VG:5060;branch=z9hG4bK27AB23DA
Call-ID: 4B361D9C-2CE311ED-FFFFFFFFB3579552-6543160B
From: <sip:anonymous@anonymous>
To: <sip:anonymous@anonymous>;tag=inittmld
CSeq: 8 REGISTER
Warning: 399 IP ITSP "SS280000F1048642L10295298[00000] From header absent or undecipherable",399 IP ITSP "SS280000F1048642L15013890[00000] To header absent or undecipherable"
Content-Length: 0
09-06-2022 02:11 AM
This is not a call.
This is one half of the registration process.
You are receiving "400 Bad Request" from the ITSP. But there is no Register-Message in that log, this would be the most interesting message right now.
09-06-2022 04:02 AM
The first part of your output is a SIP option ping, then the lower part is a 400 Bad Request that state that your request is invalid with this error.
"Warning: 399 IP ITSP "SS280000F1048642L10295298[00000] From header absent or undecipherable",399 IP ITSP "SS280000F1048642L15013890[00000] To header absent or undecipherable""
To help you with this you really need to share the output from the whole registration process/call attempt.
09-05-2022 09:37 AM
Apart from the advice given by @b.winter I would recommend you to upgrade your IOS version as the one you currently use is quite buggy. Use 17.6.3a or 17.6.4 instead.
09-06-2022 04:09 AM
Not directly related to your issue as such, but I suggest that you have a look at this document, In Depth Explanation of Cisco IOS and IOS-XE Call Routing, and modify your configuration so that you match inbound dial peer based on information in the VIA header and that you add a specific dial peer for the inbound direction from your CM, plus use one dial peer for your communication with all your CM nodes by using a server group. With this you can remove dial peer 13 and 14. This will greatly simplify any effort in troubleshooting and your future maintenance.
It would also be of help if you would share your interface configuration so that it could be verified.
09-07-2022 02:00 AM
Ok , now sip-ua registered . Thanks
But when i test incoming the result no match outgoing dial peer +6221303022xx
I do debug ccsip message , error appear 404 not found (no macthing outgoing dial peer)
let say , i do call from my mobile phone dialed number
see below my configuration :
voice translation-rule 1
rule 1 /\(.*\)/ /+6221303022xx/
!
voice translation-rule 2
rule 1 /^\+\(.*\)/ /4556/
!
voice translation-rule 3
rule 1 /^7\(.*\)/ /\1/
!
!
voice translation-profile INCOMING
translate called 2
!
voice translation-profile OUTGOING
translate calling 1
!
dial-peer voice 101 voip
description *** Inbound Dialpeer From ITSP***
translation-profile incoming INCOMING
session protocol sipv2
incoming called-number +6221303022xx
voice-class codec 1
voice-class sip tenant 100
dtmf-relay rtp-nte
no vad
!
dial-peer voice 102 voip
description *** Outbound Dialpeer To ITSP ***
translation-profile outgoing OUTGOING
destination-pattern 0.
session protocol sipv2
session target ipv4:IP ADD ITSP
voice-class codec 1
voice-class sip profiles 10
voice-class sip tenant 100
dtmf-relay rtp-nte
no vad
!
dial-peer voice 12 voip
description description To CUCM Primary
destination-pattern 4...
session protocol sipv2
session target ipv4:IP ADD CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
!
and when i test with outgoing call , outgoing work but only 1 way voice i hear
can someone give me advice ? please
Respectfully,
Solihul
09-07-2022 02:07 AM
Please: If you have problems, post debug outputs. We cannot give you advices without information from you.
Debug outputs and the current config are the basic infos that anyone needs to help you.
So, please get the the full output of the following debugs:
debug ccsip messages
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74
debug voice translation
Is the incoming call matching the correct inbound dial-peer?
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