media issue to my sip config ISR4321

[angembaki2002]

dear support can you help me on this eureur please 

call outgoing is no work

1208 : 9 5006360ms.1 (21:43:47.720 UTC Thu Jun 15 2023) +-1 +21190 pid:100 Originate 0043814448189
dur 00:00:00 tx:0/0 rx:1022/20440 0 () dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.145.127.99:48786 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off Transcoded No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00
120F : 11 5059800ms.2 (21:44:41.160 UTC Thu Jun 15 2023) +11660 +34710 pid:100 Originate 00439999212548
dur 00:00:23 tx:0/0 rx:1529/30580 0 () dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.145.127.98:49080 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off Transcoded No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00

[Roger Kallberg]

Please share the output from these debugs running simultaneously, debug ccsip message and debug voip ccapi inout, when you’re doing a call attempt. Also please share your running configuration and outline what the different dial peers purpose is so that we can better understand the intended use of them.

[angembaki2002]

hello sir this is my config now 

voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
!
voice class media 1
!
voice class codec 1
codec preference 1 g723r53
codec preference 2 g729br8
codec preference 3 g726r16
codec preference 4 g726r24
codec preference 5 g726r32
codec preference 6 g728
codec preference 8 g729r8
codec preference 11 g711alaw
codec preference 12 g711ulaw
codec preference 13 clear-channel
!
!
!
!
!
!
!
!
license udi pid ISR4321/K9 sn FDO22222NBY
license boot level uck9
license boot level securityk9
license smart enable
!
spanning-tree extend system-id
!
username Cisco password 0 Cisco
!
redundancy
mode none
!
!
!
!
!
vlan internal allocation policy ascending
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0/0
ip address 192.168.10.2 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/1
ip address 10.108.202.138 255.255.255.252
negotiation auto
!
interface GigabitEthernet0/1/0
!
interface GigabitEthernet0/1/1
!
interface GigabitEthernet0/1/2
!
interface GigabitEthernet0/1/3
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
negotiation auto
!
interface Vlan1
no ip address
!
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.108.202.137
ip route 10.108.202.137 255.255.255.255 10.145.127.XXX
!
!
control-plane
!
!
!
!
!
!
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 100 voip
description **** Outgoing Call to SIP-TRUNK *****
translation-profile outgoing OUT
preference 1
destination-pattern .%
session protocol sipv2
session target ipv4:10.145.127.XXX
voice-class codec 1
clid network-number 004815789658
no vad
!
dial-peer voice 101 voip
description **** Outgoing Call to SIP-TRUNK *****
translation-profile outgoing OUT
destination-pattern .%
session protocol sipv2
session target ipv4:10.145.127.XXX
no vad
!
dial-peer voice 102 voip
description **** Outgoing Call to SIP-TRUNK *****
translation-profile outgoing OUT
destination-pattern .%
session protocol sipv2
session target ipv4:10.145.127.XXX
no vad
!
dial-peer voice 103 voip
description **** Outgoing Call to SIP-TRUNK *****
translation-profile outgoing OUT
destination-pattern .%
session protocol sipv2
session target ipv4:10.145.127.XXX
no vad
!
dial-peer voice 104 voip
description **** Outgoing Call to SIP-TRUNK *****
translation-profile outgoing OUT
destination-pattern .%
session protocol sipv2
session target ipv4:10.145.XXX.XXX
no vad
!
dial-peer voice 140 voip
description OUTBOUND G711 Voice SIP calls to VzB
translation-profile outgoing Outgoing-Translate
destination-pattern .T
session protocol sipv2
session target ipv4:10.145.127.XXX:5060
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
!
sip-ua
retry invite 3
retry register 10
timers register 150
sip-server ipv4:10.145.127.XXX
!
!
line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
privilege level 15
password LINE
login local
transport input telnet ssh
!
ntp master
!
end

[Roger Kallberg · Accepted solution]

Your dial peers are very badly written. I would strongly recommend you to read this document to understand how call routing in IOS operates. In Depth Explanation of Cisco IOS and IOS-XE Call Routing - Cisco Once you have done that rework your dial peers from scratch. Some help on the way, you should have these dial peers at a minimum.

• Inbound from CM - Recommend you to use information in the VIA header to match traffic from CM
• Outbound to CM - Use server groups if you have more than one CM and use E.164 number maps if you have multiple DID ranges or discontinued numbers
• Inbound from SP - Recommend you to use information in the VIA header to match traffic from SP
• Outbound to SP - Use server groups if you have more than one destination to reach at the SP

On the outbound dial peers do not use wide destination patterns like .T or .% as that will have the potential to cause call routing loops between CM and the SP. That is something you should avoid.

I would as well recommend you to read up on how to setup a Cisco router as an SBC by using the Cube functionality as you have not activated that feature set in your configuration. This document would be appropriate for that. Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 

Then for anyone to be able to give you any help on this you will need to share the output from the debugs I listed before. Without that we have no way of knowing what issue you run into.

[angembaki2002]

tank you sir 

the topology of my network is to configure incoming calls and still not telephones or other services all I want to do is just my calls come from a paxb in 192.168.xxx and send his call to my cisco and my cisco ends his call call to the service provider and only outgoing calls with 20 lines so my configuration is simple as you see then the real problem when I make a first call they go through but when the call ends I relaunch a second call it no longer passes someone can have an idea of what is missing on the config of this model of router really

[Roger Kallberg]

Again read the shared documents and if you still cannot figure out how to get this working reach out to a reputable Cisco system integrator and request their help with this.

The community can help you if you provide the necessary information as been requested in various replies. If you select not to do so we cannot help you. If you want examples of configurations use the search function in the community.

[angembaki2002]

hello sir 

*Jun 22 16:50:28.707: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:50:29.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:50:30.307: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:50:32.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:50:36.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:50:44.259: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61480;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61480>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61478 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:51:00.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 102.223.130.51:61484;branch=z9hG4bK-d87543-380be508223f3d16-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:3314857448789@102.223.130.51:61484>
To: "0844433774"<sip:0844433774@197.231.255.90>
From: "TEST ANGE"<sip:3314857448789@197.231.255.90>;tag=c12eca03
Call-ID: NzgwZDUxMzlhZmM2OTZmY2I0YzUwZTQyNGY5YWNjOWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 329

v=0
o=- 4 2 IN IP4 102.223.130.51
s=CounterPath X-Lite 3.0
c=IN IP4 102.223.130.51
t=0 0
m=audio 61482 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : 2bDrdzPS 6AglutID 192.168.13.24 1360
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

*Jun 22 16:51:05.619: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:62710015594758748@197.231.255.90 SIP/2.0
Via: SIP/2.0/UDP 140.99.156.74:63849;branch=z9hG4bK1203916585
Max-Forwards: 70
From: <sip:voip01@197.231.255.90>;tag=2023236854
To: <sip:62710015594758748@197.231.255.90>
Call-ID: 2059397628-1434465430-1215051359
CSeq: 1 INVITE
Contact: <sip:voip01@140.99.156.74:63849>
Content-Type: application/sdp
Content-Length: 210
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH

v=0
o=voip01 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

[Roger Kallberg]

All I can see in what you shared is a bunch of inbound invites coming from the same source, but nothing more, the gateway isn’t even responding to the invite.

Received:
INVITE sip:0844433774@197.231.255.90 SIP/2.0

What is 197.231.255.90?

[angembaki2002]

hello sir this ip for my pabx on remote site 197.231.255.90?

normaly my sip server is 192.168.30.21

[Roger Kallberg]

I’m sorry, but none of what you’re saying makes any sense. For anyone to be able to support you on this you’re going to need to provide an outline of your topology. Can you please provide a graphical overview of your system landscape so that an outsider can understand your system topology? In the outline please provide IP addressing of all elements so that it’s possible for an outsider to understand your system landscape.

Apart from that please also provide a detailed outline of your call flow, broken down into each step, like call from phone A, connected to CM or whatever it might be connected to, call goes via SIP trunk to gateway, sent to service provider via SIP trunk and so on.

[angembaki2002]

to day i have this message 

SIP: Trying to parse unsupported attribute at media level
SIP: Trying to parse unsupported attribute at media levelsh call
Router#sh callhis
Router#sh call his
Router#sh call history voi
Router#sh call history voice br
Router#sh call history voice brief
<ID>: <CallID> <start>ms.<index> (<start>) +<connect> +<disc> pid:<peer_id> <direction> <addr>
dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>) dscp:<packets violation> media:<packets violation> audio tos:<audio tos value> video tos:<video tos value>
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
delay:<last>/<min>/<max>ms <codec> <textrelay> <transcoded>

media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
LostPacketRate:<%> OutOfOrderRate:<%>
MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
last <buf event time>s dur:<Min>/<Max>s
FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
<codec> (payload size)
Telephony <int> (callID) [channel_id] tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops> disc:<cause code>
speeds(bps): local <rx>/<tx> remote <rx>/<tx>
Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
bw: <req>/<act> codec: <audio>/<video>
tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>

Telephony call-legs: 0
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
Total call-legs: 1
11F0 : 1 216120ms.1 (*00:54:07.839 UTC Fri Jun 23 2023) +-1 +10 pid:100 Answer 814596887
dur 00:00:00 tx:0/0 rx:0/0 15 (call rejected (21)) dscp:0 media:0 audio tos:0x0 video tos:0x0
IP 192.168.50.11:28312 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long dur callduration :n/a timestamp:n/a

LostPacketRate:0.00 OutOfOrderRate:0.00

[Roger Kallberg]

The call is being rejected by the gateway, (call rejected (21)). Very likely because you don’t have any dial peer that has a destination for the IP address where the invite is coming from as then the built in security in IOS will reject the call.

Have you reworked your configuration with the suggestion you got in this post? If so can you please post the current running configuration so that we can verify it?

[angembaki2002]

hello sir this is my actuel config 

Router#sh run
Building configuration...

Current configuration : 3566 bytes
!
! Last configuration change at 06:27:46 UTC Fri Jun 23 2023
!
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
!
!
!
!
!
!
!
!
!
!
!
!
!
!
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
cts logging verbose
!
!
voice-card 0
dspfarm
!
!
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
rtp-media-loop count 21
allow-connections sip to sip
no supplementary-service sip handle-replaces
sip
no silent-discard untrusted
!
voice class codec 1
codec preference 1 g723r53
codec preference 2 g729br8
codec preference 3 g726r16
codec preference 4 g726r24
codec preference 5 g726r32
codec preference 6 g728
codec preference 8 g729r8
codec preference 11 g711alaw
codec preference 12 g711ulaw
codec preference 13 clear-channel
!
!
!
voice class dualtone-detect-params 1
freq-max-deviation 50
!
!
voice register global
mode srst
!
!
!
!
voice translation-profile 1
!
!
!
license udi pid CISCO2911/K9 sn FGL19501014
license accept end user agreement
license boot module c2900 technology-package uck9
!
!
username 1000 password 0 cisco
!
redundancy
!
!
!
!
!
!
class-map match-any SIP
match ip precedence 3
match ip dscp af31
match protocol sip
class-map match-any RTP
match ip precedence 5
match ip dscp ef
match protocol rtp
!
policy-map VoIP
class SIP
bandwidth percent 5
class RTP
priority percent 70
policy-map Outbound
class class-default
shape average 10000000
service-policy VoIP
!
!
!
!
!
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 197.231.255.90 255.255.255.252
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
!
interface GigabitEthernet0/1
ip address 192.168.30.10 255.255.255.0
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
!
interface GigabitEthernet0/2
ip address 192.168.50.10 255.255.255.0
duplex auto
speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip nat inside source static tcp 192.168.30.1 5060 197.231.255.90 80 extendable
ip nat outside source static 192.168.30.21 197.231.255.90
ip route 0.0.0.0 0.0.0.0 197.231.255.89
ip route 172.231.255.89 255.255.255.255 192.168.30.21
!
!
!
!
control-plane
!
!
!
!
!
!
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 101 voip
tone ringback alert-no-PI
description *** SIP Internal Server [[ CN+ ]] ***
destination-pattern 0[8,9]........
session protocol sipv2
session target ipv4:192.168.30.21
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
clid strip name
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
retry options 1
timers trying 1000
registrar ipv4:192.168.30.21 expires 3600
sip-server ipv4:192.168.30.21
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
password cisco
login
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0
privilege level 15
password cisco
login local
transport input telnet
transport output telnet
line vty 1 4
password cisco
login local
transport input ssh
transport output telnet
!
scheduler allocate 20000 1000
!
end

[Roger Kallberg]

Sorry, but it quite clear that you need help from a reputable company to get this sorted. I don’t think that the community can help you with this as based on your latest shared configuration you clearly do not follow the advice given. Based upon the configurations, present and former, nothing can work related to call services.

[M02@rt37]

Hello @angembaki2002,

Agree with @Roger Kallberg, you have several dial peers configured for outgoing calls but it's unclear which dial peer is being used for the specific outgoing calls that are not working.