Cisco ip phones

[asfandyar779514494]

Hey guys

I have configured IP phones using CUCM 12.5. Now the issue I am facing is that in outgoing call (IP phone to mobile and vice versa) I do receive call but unfortunately it ends within a second.

I am sharing router's configuration, if anyone could tell me if there is issue with my configuration or there is some problem with  ISP's configuration.

service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
service call-home
platform qfp utilization monitor load 80
no platform punt-keepalive disable-kernel-core
platform hardware throughput level 100000
!
hostname Khi-Voice-Router
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!

!
aaa new-model
!
!
aaa authentication login default group tacacs+ local
aaa authentication enable default group tacacs+ enable
aaa authorization config-commands
aaa authorization exec default group tacacs+ local
aaa authorization commands 1 default group tacacs+ local
aaa authorization commands 15 default group tacacs+ local
aaa authorization network default local
aaa accounting exec default start-stop group tacacs+
aaa accounting commands 1 default start-stop group tacacs+
aaa accounting commands 15 default start-stop group tacacs+
aaa accounting connection default start-stop group tacacs+
!
!
!
!
!
!
aaa session-id common
call-home
! If contact email address in call-home is configured as sch-smart-licensing@ci sco.com
! the email address configured in Cisco Smart License Portal will be used as co ntact email address to send SCH notifications.
contact-email-addr sch-smart-licensing@cisco.com
profile "CiscoTAC-1"
active
destination transport-method http
no destination transport-method email
!
!
!
!
!
!
!
ip name-server 10.16.128.4 10.16.6.11
ip domain name telenorbank.pk
!
!
!
!
!
!
!
!
!
!
subscriber templating
!
!
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
crypto pki trustpoint TP-self-signed-3202423008
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3202423008
revocation-check none
rsakeypair TP-self-signed-3202423008
!
crypto pki trustpoint SLA-TrustPoint
enrollment terminal
revocation-check crl
!
!

!
!
!
!
voice service voip
no ip address trusted authenticate
mode border-element license capacity 10
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
header-passing
early-offer forced
midcall-signaling passthru
error-code-override total-calls failure 486
!
voice class codec 1
codec preference 1 g711ulaw
!
!
!
voice class e164-pattern-map 1
description **2 CUCM Cluster**
!
!
voice class server-group 1
ipv4 10.255.250.114 preference 1
description **CUCM Cluster**
!
voice class server-group 2
ipv4 10.2.42.114
description **to ITSP SIP Trunk**
!
!
!
!
voice translation-rule 1
rule 1 /^0218005800/ /2001/
!
!
voice translation-profile inbound
translate called 1
!
!
!
!
!
voice-card 0/1
no watchdog
!
voice-card 0/4
no watchdog
!
license feature hseck9
license udi pid ISR4321/K9 sn FDO233610BA
license accept end user agreement
license boot level appxk9
license boot level uck9
license boot level securityk9
license smart enable
license smart conversion automatic
diagnostic bootup level minimal
spanning-tree extend system-id
!
!
!

!
redundancy
mode none
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
no ip address
negotiation auto
!
interface GigabitEthernet0/0/0.136
description *connectd with lan **
encapsulation dot1Q 136
ip address 10.255.250.114 255.255.255.240
!
interface GigabitEthernet0/0/1
description *Connected with ISp*
ip address 10.2.42.114 255.255.255.252
negotiation auto
!
interface Service-Engine0/1/0
!
interface Service-Engine0/4/0
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
shutdown
negotiation auto
!
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip tftp source-interface GigabitEthernet0
ip route 0.0.0.0 0.0.0.0 10.255.250.113
ip route 10.16.17.44 255.255.255.255 10.2.42.113
ip route 10.16.17.61 255.255.255.255 10.2.42.113
ip route 10.112.1.132 255.255.255.255 10.2.42.113
ip tacacs source-interface GigabitEthernet0/0/0.136
!
ip ssh version 2
!
!
!

!
!
!
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/1/4
!
voice-port 0/1/5
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 2 voip
description **CUBE TO PSTN**
incoming called e164-pattern-map 1
!
dial-peer voice 1 voip
destination-pattern 0218005800
session protocol sipv2
session target ipv4:192.168.3.175
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0.136
voice-class sip bind media source-interface GigabitEthernet0/0/0.136
dtmf-relay rtp-nte sip-notify
!
dial-peer voice 3 voip
description * Outbound to ITSP***
destination-pattern .T
session protocol sipv2
session target ipv4:10.112.1.132
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
no vad
!
dial-peer voice 4 voip
description * Inbound calls from ITSP***
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
!
!
gateway
media-inactivity-criteria all
timer receive-rtp 1200
!
sip-ua
!
!
line con 0

!
wsma agent exec
!
wsma agent config
!
wsma agent filesys
!
wsma agent notify
!
!
end

[mubashirathar1999]

Hi,

Does the call drop within one second after picking up from the other end? or you get only 1 ring and the call gets disconnected? if that's the case then it could be signalling or Dial-peer issue. May be some Translation-pattern and Route-Pattern configs needed on the CUCM side. 

Please configure the following two debugs command and do a test incoming and outgoing call for better understanding. 

terminal monitor

Debug ccsip messages

debug voip ccapi inout

Please share the output here. May be the call is not even coming to the CUBE at all, in that case we will have to check the CUCM side first. 

[asfandyar779514494]

I get one ring and call disconnects, I will run these debugs and share output shortly

[asfandyar779514494]

Log is attached, please take a look

[mubashirathar1999]

Alright, so here are a few things I have noticed. First of all, the call is rejected from the SIP provider but the reason could be that you are not sending the call to them with required details. From your debugs I can see the following:

Received:
SIP/2.0 603 Declined -----> Message received from ITSP 
Via: SIP/2.0/UDP 10.2.42.114:5060;branch=z9hG4bK7D77
Call-ID: A801D650-26E911EA-83E5FAFE-FB88ACD@10.2.42.114
To: <sip:03160103001@10.112.1.132:5060>;tag=as736680c5
From: <sip:0218005800@10.2.42.114>;tag=2B6B29-D7E
CSeq: 101 INVITE
Contact: sip:0218005800@10.112.1.132:5060
Server: Asterisk PBX 14.6.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

603 Decline
The destination does not wish to participate in the call, or cannot do so, and additionally the destination knows there are no alternative destinations (such as a voicemail server) willing to accept the call

There could be multiple reasons for this response, before reaching out to the ITSP I would suggest to check the following : 

1) NAT configuration if you are behind a firewall

2) Check if the DID is activated from the ITSP (0218005800)

3) Check if there is any "sip-registrar" , "sip-authentication" or any other config required by the ITSP under the "sip-ua" field

4) Add the following configs:

voice service voip

ip address trusted list
ipv4 10.112.1.132

ipv4 10.2.42.113

ipv4 10.16.17.44

ipv4 10.16.17.61 

ipv4 10.255.250.114

ipv4 10.255.250.113

ipv4 192.168.3.175

5) I noticed the call flow as following, please correct me if I am wrong

Calling (Source): 0218005800 --->192.168.3.175 (CUCM)
Called (Destination) : 03160103001 ---- > 10.255.250.114 (GigabitEthernet0/0/0.136)

IP Phone 2001 --> CUCM (192.168.3.175) ---> SIP Trunk to CUBE (10.255.250.114)----> SIP to ITSP (10.2.42.113) ---ITSP RTP (10.112.1.132) ----> Destination Number 03160103001

In this case, the first Dial-peer it should hit on your CUBE should be the incoming Dial-peer coming from CUCM but I see that it hits following:

Incoming Dial Peer = 4 (LAN DP) ---> This is inbound Dial-peer from ITSP to CUBE for receiving calls from ITSP ...this should be replaced with Dial-peer for CUCM to CUBE

Outgoing Dial Peer = 3 (WAN DP)

I do not see any Inbound Dial-peer configured that would send the calls from CUCM to CUBE, Incoming Dial-peer should be that Inbound. I would recommend using uri for that purpose

conf t 

voice class uri 100 sip

host ipv4:192.168.3.175 (Your CUCM Pub/Sub IP address, you can add multiple lines here)

dial-peer voice 100 voip
description <LAN DP - Inbound calls from CUCM to CUBE>
session protocol sipv2
incoming uri via 100
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0.136
voice-class sip bind media source-interface GigabitEthernet0/0/0.136
dtmf-relay rtp-nte sip-notify sip-kpml
no vad

end

Also change the Dial-peer 4 as follows:

dial-peer voice 4 voip

incoming called-number .T

voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1

After configuring the above and confirming the other remaining, make another test call from the same Phone to the same destination and take debugs. The dial-peer it should chose be:

incoming dial-peer = 100 (LAN DP - Inbound)

Outgoing Dial-peer = 3 (WAN DP- Outbound)

Hope this helps and let us know how it goes.