01-24-2012 09:17 AM
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.
Solved! Go to Solution.
02-11-2012 02:05 AM
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.
02-29-2012 09:49 AM
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.
01-25-2012 01:19 AM
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=off211>
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.
01-25-2012 02:44 AM
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=off018877000>
From: "Sam " <018877000>;tag=D31DD558-289018877000>
To: <0862157774>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>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=D31DD558-289018877000>
To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b12370862157774>
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=D31DD558-289018877000>
To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b12370862157774>
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=D31DD558-289018877000>
To: <0862157774>;tag=7fad626f7fe8-100007f-13c4-55013-b1237-5716034e-b12370862157774>
Not sure where I can change that.
Regards,
01-25-2012 03:23 AM
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.
01-25-2012 03:27 AM
It could be also a problem of called number format.
Maybe, you must add a prefix or format the number in E.164.
01-25-2012 03:29 AM
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,
01-25-2012 03:48 AM
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.
01-25-2012 03:54 AM
Another idea.
Can you debug an incoming call?
In this way we can see the called number format and other sip parameters.
01-25-2012 06:29 AM
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.
01-25-2012 06:53 AM
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.
01-25-2012 08:02 AM
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=off18877000>
From: "Sam " <18877000>;tag=D4410748-1C9D18877000>
To: <0862157774>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>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=off18877000>
From: "Sam " <18877000>;tag=D4410748-1C9D18877000>
To: <0862157774>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>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=off18877000>
From: "Sam " <18877000>;tag=D4410748-1C9D18877000>
To: <0862157774>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>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=off18877000>
From: "Sam " <18877000>;tag=D4410748-1C9D18877000>
To: <0862157774>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>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.
01-25-2012 09:12 AM
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,
01-25-2012 10:01 AM
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?
01-26-2012 03:50 AM
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.
01-28-2012 12:37 AM
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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