Solved: Please help with SIP configuration on 2801 router - Page 2

firestormnet · ‎01-24-2012

Hi All.

Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.

The information from our SIP provider:

We have issued the following DDI range: 018877000 – 99

There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.

Configuration details are as follows:

Our Primary Proxy:- 99.234.56.78

Codec supported:- G711Alaw, G729 (G711Alaw is the preferred codec)

Fax Support:- T38 and G711Alaw

DTMF:- RFC2833 and INFO

CLI Method:- Remote-Party-ID

Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.

This is a SIP configuration on Cisco2801 router (I used outgoing calls only):

!

translation-rule 10

Rule 0 ^90 0

Rule 1 ^91 1

Rule 2 ^92 2

Rule 3 ^93 3

Rule 4 ^94 4

Rule 5 ^95 5

Rule 6 ^96 6

Rule 7 ^97 7

Rule 8 ^98 8

Rule 9 ^99 9

!

interface FastEthernet0/0.1

description ***DATA VLAN***

encapsulation dot1Q 1 native

ip address 10.1.1.101 255.255.255.0

!

interface FastEthernet0/0.2

description ***VOICE VLAN***

encapsulation dot1Q 2

ip address 192.168.22.1 255.255.255.0

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

h323

call start slow

sip

bind control source-interface FastEthernet0/0.2

bind media source-interface FastEthernet0/0.2

registrar server expires max 36000 min 600

!

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

codec preference 3 g711alaw

!

dial-peer voice 1 pots

description ### External Dialling via BRI ###

preference 7

destination-pattern 9T

translate-outgoing called 10

direct-inward-dial

port 0/0/0

forward-digits all

!

dial-peer voice 2 pots

description ### External Dialling via BRI ###

preference 2

destination-pattern 9T

translate-outgoing called 10

direct-inward-dial

port 0/0/1

forward-digits all

!

dial-peer voice 9000 voip

description ** Outgoing calls to SIP **

preference 1

destination-pattern 9T

voice-class sip dtmf-relay force rtp-nte

session protocol sipv2

session target ipv4:99.234.56.78:5060

dtmf-relay rtp-nte

codec g711alaw

no vad

!

sip-ua

timers connect 100

sip-server ipv4:99.234.56.78

!

I used debugging commands to troubleshoot the calls.

!

2801(config-dial-peer)#

094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,

Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH

094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018

094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH

094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=9

094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH

094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90

094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH

094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=908

094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH

094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=9086

094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH

094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862

094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH

094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=908621

094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH

094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=9086215

094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH

094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157

094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH

094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=908621577

094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH

094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=9086215777

094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH

094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774

094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Partial Matches(1) after DP_MATCH_DEST

094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=MORE_DIGITS_NEEDED(1)

094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH

094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774T

094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH

094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774

094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Calling Number=90862157774, Called Number=, Voice-Interface=0x0,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH

094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000

094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH

094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774

094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL

094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3

Remote-Party-ID: "Sam " <sip:211@192.168.22.1>;party=calling;screen=no;privacy=off

From: "Sam " <sip:211@192.168.22.1>;tag=CDCFB8AC-F98

To: <sip:90862157774@99.234.56.78>

Date: Tue, 24 Jan 2012 09:27:10 GMT

Call-ID: 6D3919CF-45A411E1-9059813F-74BAA80B@192.168.22.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1787264879-1168380385-2421457215-1958389771

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1327397230

Contact: <sip:211@192.168.22.1:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 244

v=0

o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1

s=SIP Call

c=IN IP4 192.168.22.1

t=0 0

m=audio 18258 RTP/AVP 8 101

c=IN IP4 192.168.22.1

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3

Remote-Party-ID: "Sam" <sip:211@192.168.22.1>;party=calling;screen=no;privacy=off

From: "Sam " <sip:211@192.168.22.1>;tag=CDCFB8AC-F98

To: <sip:90862157774@99.234.56.78>

Date: Tue, 24 Jan 2012 09:27:11 GMT

Call-ID: 6D3919CF-45A411E1-9059813F-74BAA80B@192.168.22.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1787264879-1168380385-2421457215-1958389771

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1327397231

Contact: <sip:211@192.168.22.1:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 244

v=0

o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1

s=SIP Call

c=IN IP4 192.168.22.1

t=0 0

m=audio 18258 RTP/AVP 8 101

c=IN IP4 192.168.22.1

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3

Remote-Party-ID: "Sam " <sip:211@192.168.22.1>;party=calling;screen=no;privacy=off

From: "Sam " <sip:211@192.168.22.1>;tag=CDCFB8AC-F98

To: <sip:90862157774@99.234.56.78>

Date: Tue, 24 Jan 2012 09:27:12 GMT

Call-ID: 6D3919CF-45A411E1-9059813F-74BAA80B@192.168.22.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1787264879-1168380385-2421457215-1958389771

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1327397232

Contact: <sip:211@192.168.22.1:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 244

v=0

o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1

s=SIP Call

c=IN IP4 192.168.22.1

t=0 0

m=audio 18258 RTP/AVP 8 101

c=IN IP4 192.168.22.1

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3

Remote-Party-ID: "Sam" <sip:211@192.168.22.1>;party=calling;screen=no;privacy=off

From: "Sam" <sip:211@192.168.22.1>;tag=CDCFB8AC-F98

To: <sip:90862157774@99.234.56.78>

Date: Tue, 24 Jan 2012 09:27:14 GMT

Call-ID: 6D3919CF-45A411E1-9059813F-74BAA80B@192.168.22.1

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 1787264879-1168380385-2421457215-1958389771

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1327397234

Contact: <sip:211@192.168.22.1:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 244

v=0

o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1

s=SIP Call

c=IN IP4 192.168.22.1

t=0 0

m=audio 18258 RTP/AVP 8 101

c=IN IP4 192.168.22.1

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

I made some changes in the router configuration.

I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).

The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found

Then it moves to ISDN line, and use this line to make a call.

102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH

102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774T

102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH

102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774

102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Calling Number=90862157774, Called Number=, Voice-Interface=0x0,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH

102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000

102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH

102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=90862157774

103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=9000

2: Dial-peer Tag=2

3: Dial-peer Tag=1

103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL

103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9

Remote-Party-ID: "Sam" <sip:211@10.1.1.101>;party=calling;screen=no;privacy=off

From: "Seam" <sip:211@10.1.1.101>;tag=CEF37490-172C

To: <sip:90862157774@99.234.56.78>

Date: Tue, 24 Jan 2012 14:45:47 GMT

Call-ID: EF977E91-45D011E1-9309813F-74BAA80B@10.1.1.101

Supported: 100rel,timer,resource-priority,replaces

Min-SE: 1800

Cisco-Guid: 3989446920-1171263969-2466545983-1958389771

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1327416347

Contact: <sip:211@10.1.1.101:5060>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 19412 RTP/AVP 8 101

c=IN IP4 10.1.1.101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

From: "Sam "<sip:211@10.1.1.101>;tag=CEF37490-172C

To: <sip:90862157774@99.234.56.78>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142

Call-ID: EF977E91-45D011E1-9309813F-74BAA80B@10.1.1.101

CSeq: 101 INVITE

Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9

Content-Length: 0

103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:90862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9

From: "Sam " <sip:211@10.1.1.101>;tag=CEF37490-172C

To: <sip:90862157774@99.234.56.78>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142

Date: Tue, 24 Jan 2012 14:45:47 GMT

Call-ID: EF977E91-45D011E1-9309813F-74BAA80B@10.1.1.101

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up

103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH

103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=211

103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=20018

103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH

103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=0862157774

103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

Result=NO_MATCH(-1)

103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down

2801(config-dial-peer)#

Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.

But it didn’t affect anything.

Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.

Really stuck and don't know where to look at.

Any help will be highly appreciated.

Thanks.

firestormnet · ‎01-30-2012

Hi Dan.

I can make only Outgoing call without Audio.

Incoming call I can't make at all, and ASA debug sip doesn't show anything.

Also I added Security policy access rules for LAN: Any to SIP(UDP and TCP: 5060), and Outside: SIP(UDP and TCP: 5060) to any.

Should I open more ports? As I can see in ASA log (LAN:route/51914 and /63675).

Regards,

Daniele Giordano · ‎01-30-2012

Please try these steps:

1) The default RTP port range is from 16384 to 32767 UDP port.

Can you try to add a rule in your ACL to permit this range?

2) Can you add the output of "debug ccsip messages" command from your 2801 again?

Ensure that SIP uses interface FastEthernet 0/0.2 of the router. This must be the port that connect to ASA (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2).

provider ===> ASA ===> f0/0.2 2801

3) What is the IOS version of ASA? Can you try to disable SIP inspect?

policy-map global_policy

class inspection_default

no inspect sip

Sometimes SIP ALG or SIP inspection can create issues with privider SBC.

4) As last option and temporary test, can you try to bypass ASA and connect the 2801 direct to internet?

Let me know what happens.

Regards.

firestormnet · ‎02-03-2012

Hi Dan.

Thank you for your advise, was busy on other projects, so will try that hopefully today.

Cheers

firestormnet · ‎02-09-2012

Ok, did a lot of testing. Some results are below:

On ASA and 2801 side:

1) The default RTP port range is from 16384 to 32767 UDP port. Can you try to add a rule in your ACL to permit this range?

I tried to implement many ACL rules and NAT translations but nothing worked. I used:

ACL:

LAN: any to SIP(IP) tcp/udp sip(5060)

Outside: SIP(IP) to any tcp/udp sip(5060)

NAT:

for LAN: Static source - LANrouter (10.1.1.101) udp 5060 to translated - SIP (IP) udp 5060

for outside: Static source - SIP (IP) udp 5060 to translated - LANrouter(10.1.1.101) udp 5060

I couldn't find how to allow RTP port range, it was using default port range 0-65553 only. ASDM has a Packet Tracer, so I did some testing (for UDP):

1) Int:LAN Source:10.1.1.101 Dest:88.99.77.44 Port:sip Status: ok

2) Int:LAN Source:10.1.1.101 Dest:99.234.56.78 Port:sip Status: ok

3) Int:LAN Source:192.168.22.1 Dest:88.99.77.44 Port:sip Status: drop NAT action - drop

4) Int:outside Source:99.234.56.78 Dest:10.1.1.101 Port:sip Status:drop NAT action - drop

5) Int:outside Source:88.99.77.44 Dest:10.1.1.101 Port:sip Status:drop ACL action - drop

6) Int:outside Source:99.234.56.78 Dest:192.168.22.1 Port:sip Status:drop NAT or ACL action - drop

7) Int:outside Source:99.234.56.78 Dest:88.99.77.44 Port:sip Status: ok

I tried #debug ip nat sip command, shows nothing.

Tried to use 192.168.22.1 insteed of 10.1.1.101 in NAT and ACL but with no results.

2) Can you add the output of "debug ccsip messages" command from your 2801 again?

Ensure that SIP uses interface FastEthernet 0/0.2 of the router. This must be the port that connect to ASA (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2).

provider ===> ASA ===> f0/0.2 2801

When I use Fa0/0.2 (voice subint) nothing works, when I remove it, outgoing call works without audio. Debug looks the same, shows only Send Invite, no Receive.

3) What is the IOS version of ASA? Can you try to disable SIP inspect?

ASA v.7.2(4). SIP inspect is disabled.

4) As last option and temporary test, can you try to bypass ASA and connect the 2801 direct to internet?

As nothing worked for me I decided to create a new connection. I connected UC540 to public ip 88.99.77.45 one from our ip range and asked SIP provider to change ip to .45 then configured UC with SIP details. So now all calls are working. There are NAT and ACL, but I didn't configure them at all and all calls work fine. It means SIP connection is working, now I need to move back to ASA and 2801 to make it work.

What I can see from working config and debug is that there is no Private IPs, but my old debugs shows them it means there is NAT problem in ASA, and maybe ACL problem as well.

Does anybody can provide what NAT and ACL commands should look like to allow SIP calls through ASA?

I want to configure spare ASA port with ip .45 and forward all sip calls through it. Is it possible to do that?

Thanks for your advices and time.

Regards

Daniele Giordano · ‎02-09-2012

From your last test we can assume that your issue is a NAT issue of ASA.

You have write that the SIP inspect feature of ASA is disabled.

Do you have already tried with SIP inspection enabled?

Regards.

firestormnet · ‎02-10-2012

Hi.

There is no commands in ASA such as:

class-map inspection_default

match default-inspection-traffic

!

policy-map global_policy

class inspection_default

inspect sip

!

service-policy global_policy global

I pressume there is no sip inspection configured.

Do I really need that, can we just use ACL and NAT for SIP signalling and RTP packets to flow through?

As SIP calls are working now on public ip 88.99.77.45, I need to configure a new interface Ethernet 0/5 and vlan with this ip then forward all sip traffic to that interface. Is there a way how to do this simplier?

Thanks a lot. Cheers.

Daniele Giordano · ‎02-11-2012

SIP + NAT = problem

Can you enable the inspection?

from command line use these commands:

conf t

class-map inspection_default

match default-inspection-traffic

policy-map global_policy

class inspection_default

inspect sip

service-policy global_policy global

wr mem

Regards.

firestormnet · ‎02-22-2012

Hi Daniele.

Finally something start working.

The outgoing calls are working fine now after I added policy-map and inspection rules, and NAT rule in ASA.

But incoming calls are not getting through. Debug shows:

Received:

INVITE sip:018877000@10.1.1.101 SIP/2.0

Sent:

SIP/2.0 100 Trying

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 99.234.56.78:5060;branch=z9hG4bK-303f99-bc78707e-29931e37

Received:

ACK sip:018877000@10.1.1.101 SIP/2.0

It looks like dial-peer problem, but I tried so many configurations with no results.

My dial-peer config looks:

!

translation-rule 20

Rule 1 ^018877000 211

!

dial-peer voice 9001 voip

description ** Incoming Call from SIP Trunk **

preference 10

translate-outgoing called 20

voice-class codec 1

session protocol sipv2

session target ipv4:99.234.56.78:5060

incoming called-number 018877000

dtmf-relay rtp-nte

no vad

!

Any thoughts what is missing there?

Best regards,

Daniele Giordano · ‎02-22-2012

I'm happy to hear that.

Can you add a full debug of 2801 incoming call:

debug ccsip all

and

debug voip dialpeer

Are you sure of dp matching?

Does the 211 extension exist?

Best regards.

firestormnet · ‎02-24-2012

Hi Dan.

Ok, you can see debugging from 2801 and ASA5505 in a file.

As I can see the firewall is letting packets through, so we don't need to configure anything there.

From 2801 debugging I can see that 404 error appears after it says Dial-peer no match or maybe I'm missing something.

Ext. 211 is my working phone (we use ISDN lines in office), so it should work and ring directly (direct number reach). Also I tried ext.461, spare phone, with the same result. I'll try to connect to our hunt group or AutoAttendant, I don't think it's gonna make difference.

Thanks a lot.

Regards,

Daniele Giordano · ‎02-24-2012

Your debug shows this:

SIP/2.0 100 Trying

....

007452: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING

007453: Feb 24 11:32:28.947: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

Calling Number=, Called Number=018877000, Peer Info Type=DIALPEER_INFO_SPEECH

007454: Feb 24 11:32:28.947: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=018877000

007455: Feb 24 11:32:28.947: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

007456: Feb 24 11:32:28.947: //-1/1137C22FA7A5/DPM/dpMatchPeersMoreArg:

Result=NO_MATCH(-1)

007457: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=018877000, Called Number=018877000, Peer Info Type=DIALPEER_INFO_SPEECH

007458: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=018877000

007459: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

007460: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result=NO_MATCH(-1)

007461: Feb 24 11:32:28.947: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Calling Number=018877000, Called Number=, Voice-Interface=0x0,

Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

Peer Info Type=DIALPEER_INFO_SPEECH

007462: Feb 24 11:32:28.951: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

Result=NO_MATCH(-1) After All Match Rules Attempt

007463: Feb 24 11:32:28.951: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

Calling Number=, Called Number=018877000, Peer Info Type=DIALPEER_INFO_SPEECH

007464: Feb 24 11:32:28.951: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=018877000

007465: Feb 24 11:32:28.951: //-1/1137C22FA7A5/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

007466: Feb 24 11:32:28.951: //-1/1137C22FA7A5/DPM/dpMatchPeersMoreArg:

Result=NO_MATCH(-1)

007467: Feb 24 11:32:28.951: //3598/1137C22FA7A5/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.

007468: Feb 24 11:32:28.951: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

007469: Feb 24 11:32:28.955: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3

007470: Feb 24 11:32:28.955: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7

007471: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Info/act_recdinvite_disconnect: Performing disconnect

007472: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.

007473: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x6528AC0C key=7f6ecc2ed340-837fad6643d7d6c-001b78e4b2e4@99.234.56.786C13AB08-550

007474: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Info/sipSPISendInviteResponse: Associated container=0x65C49190 to Invite Response 404

007475: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Transport/sipSPITransportSendMessage: msg=0x64FC93B4, addr=99.234.56.78, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x61108734

007476: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately

007477: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0

007478: Feb 24 11:32:28.955: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x64FC93B4, addr=99.234.56.78, port=5060, connId=0 for UDP

007479: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response

007480: Feb 24 11:32:28.955: //3598/1137C22FA7A5/SIP/State/sipSPIChangeState: 0x6528AC0C : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)

007481: Feb 24 11:32:28.959: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

The incoming call doesn't match any dial-peer.

Try this config:

dial-peer voice 9001 voip

description ** Inbound DialPeer from SIP Trunk **

voice-class codec 1

session protocol sipv2

incoming called-number 018877000

dtmf-relay rtp-nte

no vad

dial-peer voice 9002 voip

description ** Outbound DialPeer from SIP Trunk **

preference 10

translate-outgoing called 20

voice-class codec 1

session protocol sipv2

session target ipv4:99.234.56.78:5060

destination-pattern 018877000

dtmf-relay rtp-nte

no vad

The inbound dial-peers don't have a "destination". The gateway itself is the destination. These dial-peer are used to set the incoming "capability" of the gateway like codec, fax and modem gateway, vad, etc.

The outbound dial-peers are used to "forward" the call to a destination.

In order to match outbound dial peers, the router or gateway uses the dial peer destination-pattern called_number command.