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Please help with SIP configuration on 2801 router

firestormnet
Level 1
Level 1

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.

2 Accepted Solutions

Accepted Solutions

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.

View solution in original post

Ok, now is clear.

You must adjust the session-target of dial-peer 9002 to forward the incoming call to Call Manage:

dial-peer voice 9002 voip

description ** Outbound DialPeer from SIP Trunk ### handle incoming calls to CME **

preference 10

translate-outgoing called 20

voice-class codec 1

session protocol sipv2

session target ipv4: 10.1.1.101 "ip address of f0/0.1 of your 2801"

destination-pattern 018877000

dtmf-relay rtp-nte

no vad

It will work.

Regards.

View solution in original post

33 Replies 33

Now your provider responds to you with a 404 User Not Found message.

In my opinion the provider must receive a different Remote-Party-ID.

Sometimes ITSPs authorize a call on the base of whitelist/blacklist of CALLING NUMBER.

ITSP instructions contain "CLI Method:-  Remote-Party-ID".

In your Remote Party ID there is the private extension number:

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

You can try to use a translation rule to insert in the RPID one of public numbers assigned from your ITSP, e.g. 018877000

This can be an example:

translation-rule 1

Rule 0 any 018877000

dial-peer 9000

translate-outgoing calling 11

Let me know what happens.

Regards.

Hi Daniele.

Thank you for your reply. I did those changes, and debugging looks like that now:

109131: Jan 25 10:10:24.313: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

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

109132: Jan 25 10:10:24.313: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=90862157774T

109133: Jan 25 10:10:24.313: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

109134: Jan 25 10:10:24.313: //-1/9FBDA65D9466/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

109135: Jan 25 10:10:24.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

109136: Jan 25 10:10:24.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=90862157774

109137: Jan 25 10:10:24.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

109138: Jan 25 10:10:24.317: //-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

109139: Jan 25 10:10:24.317: //-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

109140: Jan 25 10:10:24.317: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

109141: Jan 25 10:10:24.317: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

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

109142: Jan 25 10:10:24.317: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=90862157774

109143: Jan 25 10:10:24.317: //-1/9FBDA65D9466/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

109144: Jan 25 10:10:24.317: //-1/9FBDA65D9466/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

109145: Jan 25 10:10:24.321: fb_get_reject_cause_code: ERROR cause_code NULL

109146: Jan 25 10:10:24.329: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK48ADAAD

Remote-Party-ID: "Sam " <018877000>;party=calling;screen=no;privacy=off

From: "Sam " <018877000>;tag=D31DD558-289

To: <0862157774>

Date: Wed, 25 Jan 2012 10:10:24 GMT

Call-ID: A18B03E6-467311E1-946B813F-74BAA80B@10.1.1.101

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

Min-SE:  1800

Cisco-Guid: 2680006237-1181946337-2489745727-1958389771

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: 1327486224

Contact: <018877000>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 3121 3032 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 19490 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

109147: Jan 25 10:10:24.389: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

From: "Sam "<018877000>;tag=D31DD558-289

To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b1237

Call-ID: A18B03E6-467311E1-946B813F-74BAA80B@10.1.1.101

CSeq: 101 INVITE

Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK48ADAAD

Content-Length: 0

109148: Jan 25 10:10:24.393: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK48ADAAD

From: "Sam " <018877000>;tag=D31DD558-289

To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b1237

Date: Wed, 25 Jan 2012 10:10:24 GMT

Call-ID: A18B03E6-467311E1-946B813F-74BAA80B@10.1.1.101

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

109149: Jan 25 10:10:24.413: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up

109150: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

109151: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=211

109152: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

109153: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20018

109154: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

109155: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=0862157774

109156: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

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

109157: Jan 25 10:10:28.448: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

   Result=NO_MATCH(-1)

109158: Jan 25 10:10:56.108: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down

As you can see nothing happened.

I was told that there is no registration or authentication required, only setup Dial-peer with session target to Invite. This Invite should work when it sees our public IP address 88.99.77.44.

Do you think that this line should have our public IP 88.99.77.44 instead of 10.1.1.101? And normal extension number?

From: "Sam "<018877000>>;tag=D31DD558-289

To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b1237

Not sure where I can change that.

Regards,

In my experience, the IP ACL of SIP provider checks the ip address in IP packet (layer 3) and not in SIP signalling. This because a sip user agent can be behind a router with NAT.

Probably the better way to fix your issue is ask to your provider the reason of the 404 message.

Can you do this?

With more informations we can fix the problem.

Regards.

It could be also a problem of called number format.

Maybe, you must add a prefix or format the number in E.164.

I asked them already yesterday. They said it's my configuration problem and they will come back to me to talk about that. It's just I can't find out where the problem is in my config.

P.S. I've tried many combinations of prefixes in rule 11, which I used to configure BRI or PRI, but none of them worked.

translation-rule 11

Rule 1 any 018877000

Such as 18877000; 7000.

And for called number as I think I need just remove 9 in from of number and send the rest to SIP server.

Thanks.

Cheeers,

What is your ITSP? Can you try another change?

Try to remove the name in your outgoing SIP messages.

Use "clid strip name" under dial-peer config.

Another idea.

Can you debug an incoming call?

In this way we can see the called number format and other sip parameters.

Hi Daniele.

Thanks for your comments.

I spoke to our SIP provider. He checked the connecton on his side and as I think he added our public IP address.

So now I can place the outgoing call and a receiving phone shows SIP provider phone number 018877000.

But there is no audio.

They told me that our router keep sending Invites, but can't receive replies. So I need to configure a few things:

1) Cisco Gateway, there is no 100 (code) for replies are coming through.

2) RTP for audio. Not sure how I can change that and to what.

3) "bind control source-interface" and " bind media source-interface" commands under "voice service voip" should show ip address of Cisco gateway. Not sure about that, but I was pointing to Fa0/0.1 and Fa0/0.2 only.

4) Configure ASA5505 for NAT translation to forward Proxy server to Router for port 5060. I think I did that but it didn't work, so maybe I was doing something wrong.

I haven't put Dial-peer for incoming calls yet, but will do that later.

Any thoughts what should be done?

P.S. Now it's very funny. When mobile phone rings using SIP provider, I end a call in the office, but mobile phone keeps ringing.

Thanks.

Cheers.

The RTP stream is negotiated by SIP/SDP parameters:

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

In this case your router says "contact me on IP 10.1.1.101, use port 19412 and use codec PCMA".

I think that your ASA uses "fix up or inspection" to change the private ip address with the public one.

Can you add an incoming dial-peer? If an incoming call doesn't match any dial-peer, the cisco router uses the hidden dial-peer 0 but sometimes the default config is not right for specific cases.

Can you add a new debug ccsip messages?

Can you add the ASA config of NAT and fix up or inspect?

Regards.

Hi Dan.

Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?

I use Cisco ASDM for ASA to make changes.

There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44)  for a few ports.

Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.

For NAT:

I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44

(AS TRANSLATED) UDP 5060

Because there is already translation for the Server.

Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.

116013: Jan 25 15:28:25.584: //-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

116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

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

116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:

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

116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=90862157774

116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/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

116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL

116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D

Remote-Party-ID: "Sam " <18877000>;party=calling;screen=no;privacy=off

From: "Sam " <18877000>;tag=D4410748-1C9D

To: <0862157774>

Date: Wed, 25 Jan 2012 15:28:25 GMT

Call-ID: EDD5040-46A011E1-970D813F-74BAA80B@10.1.1.101

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

Min-SE:  1800

Cisco-Guid: 219068878-1184895457-2533916991-1958389771

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: 1327505305

Contact: <18877000>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 18782 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

116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D

Remote-Party-ID: "Sam " <18877000>;party=calling;screen=no;privacy=off

From: "Sam " <18877000>;tag=D4410748-1C9D

To: <0862157774>

Date: Wed, 25 Jan 2012 15:28:26 GMT

Call-ID: EDD5040-46A011E1-970D813F-74BAA80B@10.1.1.101

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

Min-SE:  1800

Cisco-Guid: 219068878-1184895457-2533916991-1958389771

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: 1327505306

Contact: <18877000>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 18782 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

116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D

Remote-Party-ID: "Sam " <18877000>;party=calling;screen=no;privacy=off

From: "Sam " <18877000>;tag=D4410748-1C9D

To: <0862157774>

Date: Wed, 25 Jan 2012 15:28:27 GMT

Call-ID: EDD5040-46A011E1-970D813F-74BAA80B@10.1.1.101

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

Min-SE:  1800

Cisco-Guid: 219068878-1184895457-2533916991-1958389771

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: 1327505307

Contact: <18877000>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 18782 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

116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0862157774@99.234.56.78:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D

Remote-Party-ID: "Sam" <18877000>;party=calling;screen=no;privacy=off

From: "Sam " <18877000>;tag=D4410748-1C9D

To: <0862157774>

Date: Wed, 25 Jan 2012 15:28:57 GMT

Call-ID: EDD5040-46A011E1-970D813F-74BAA80B@10.1.1.101

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

Min-SE:  1800

Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771

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: 1327505337

Contact: <18877000>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101

s=SIP Call

c=IN IP4 10.1.1.101

t=0 0

m=audio 18782 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

I'll add Incoming dial-peer now.

Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.

Appretiate your help.

Thanks a mill.

I added incoming peer:

!

translation-rule 20

rule 1 ^18877000 211

!

voice translation-profile IN

translate called 20

!

!

dial-peer voice 9001 voip

description ** Incoming Call from SIP Trunk **

preference 1

translation-profile incoming IN

codec g711alaw

session protocol sipv2

session target ipv4:99.234.56.78

dtmf-relay rtp-nte

no vad

!

But no calls coming through. I can hear only beep twice then call ends.

Debugging doesn't show anything.

Is it Dial-peer configuration or ASA not letting calls in problem?

Cheers,

From debug we can see outgoing INVITE from your 2801 but no responses.

Now I think in a NAT problem. We must check your ASA.

Can you run the "debug sip" on the firewall?

Hi Dan.

Coming back again. I've tried some ASA monitoring but with no much luck.

I added:

#debug sip (level 1 and 255)

#term mon

then, #capture sip match udp 10.1.1.101 255.255.255.255 any

when I make in and out calls, nothing showing, its empty.

In ASDM, in Real-time log viewer I found only this, when I made an outside call. Nothing for incoming call.

6

Jan 26   2012

09:29:38

302016

SIP-Primary-Proxy(source)

  router(destin)

Teardown   UDP connection 20523150 for outside:SIP-Primary-Proxy/5060 to   LAN:router/51914 duration 0:02:31 bytes 8078








6

Jan 26   2012

09:30:04

302015

SIP-Primary-Proxy

router

Built   outbound UDP connection 20523435 for outside:SIP-Primary-Proxy/5060   (SIP-Primary-Proxy/5060) to LAN:router/51914 (88.99.77.44/36007)

6

Jan 26   2012

09:47:06

302016

SIP-Primary-Proxy

router

Teardown   UDP connection 20524230 for outside:SIP-Primary-Proxy/5060 to   LAN:router/63675 duration 0:03:50 bytes 16135

Also, some #SH NAT in ASA:

There are NAT for DMZ and LAN only.

LAN NAT has Exempts, Static and Dynamic translations. Some examples below:

#sh nat lan

  match ip LAN Voice-LAN 255.255.255.0 _internal_loopback remote1-LAN 255.255.255.0

    NAT exempt

    translate_hits = 0, untranslate_hits = 0

  match ip LAN Voice-LAN 255.255.255.0 _internal_loopback Remote2-LAN 255.255.255.0

    NAT exempt

    translate_hits = 0, untranslate_hits = 0

  match ip LAN Voice-LAN 255.255.255.0 _internal_loopback Voice-LAN 255.255.255.0

    NAT exempt

    translate_hits = 0, untranslate_hits = 0

  match ip LAN host LAN-Server _internal_loopback host NMC

    NAT exempt

    translate_hits = 0, untranslate_hits = 0

  match tcp LAN host LAN-Server eq 25 outside any

    static translation to 88.99.77.44/25

    translate_hits = 0, untranslate_hits = 50522

  match tcp LAN host LAN-Server eq 443 outside any

    static translation to 88.99.77.44/443

    translate_hits = 0, untranslate_hits = 26588

  match tcp LAN host LAN-Server eq 3389 outside any

    static translation to 88.99.77.44/3389

    translate_hits = 0, untranslate_hits = 1727035

  match tcp LAN host LAN-Server eq 8 outside any

    static translation to 88.99.77.44/8

    translate_hits = 0, untranslate_hits = 4

  match ip LAN 10.1.1.0 255.255.255.0 LAN any

    dynamic translation to pool 1 (No matching global)

    translate_hits = 0, untranslate_hits = 0

  match ip LAN 10.1.1.0 255.255.255.0 DMZ any

    dynamic translation to pool 1 (No matching global)

    translate_hits = 0, untranslate_hits = 0

  match ip LAN 10.1.1.0 255.255.255.0 outside any

    dynamic translation to pool 1 (88.99.77.44 [Interface PAT])

    translate_hits = 2527191, untranslate_hits = 133920

  match ip LAN 10.1.1.0 255.255.255.0 _internal_loopback any

    dynamic translation to pool 1 (No matching global)

    translate_hits = 0, untranslate_hits = 0

  match ip LAN any DMZ any

    no translation group, implicit deny

    policy_hits = 0

  match ip LAN any outside any

    no translation group, implicit deny

    policy_hits = 28368

firewall# sh nat outside

ERROR: No matching NAT policy found

ASA5505 NAT config:

nat-control

global (DMZ) 2 interface

global (outside) 1 interface

nat (LAN) 0 access-list remote-vpn

nat (LAN) 1 10.1.1.0 255.255.255.0

nat (DMZ) 1 NMC 255.255.255.255

static (LAN,outside) tcp interface smtp LAN-Server smtp netmask 255.255.255.255

static (LAN,outside) tcp interface https LAN-Server https netmask 255.255.255.255

static (LAN,outside) tcp interface 3389 LAN-Server 3389 netmask 255.255.255.255

static (DMZ,outside) tcp 88.99.77.45 5900 NMC 5900 netmask 255.255.255.255

static (LAN,outside) tcp interface 8 LAN-Server 8 netmask 255.255.255.255

I'm not sure if it gives you some clues.

Thank you for time.

Cheers.

If I've correctly understand, the call is established but without audio in both direction.

Right?

Your ASA real time log shows the call, so SIP signalling should be ok.

Do you have ACL that can block RTP traffic?

Regards.

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