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VG204 SIP error Address Incomplete mapping/binding

jensho
Level 1
Level 1

Hello,
LAB testing with a VG204 and getting no ring out of the fxs port.

In debug mode: call is there but throwing: SIP/2.0 484 Address Incomplete
homebox router provides SIP account with internal number 624, this account is working on laptop using a softphone client. no issues with user&password.
Login User for extension 624 is vg204001

I assume that the homebox is sending the username as caller id instead the real extension and therefore dial-peer will never match.
Tested many combinations but no luck.

Could somebody provide me a hint what's going wrong?

Config snipps:

voice service voip
gcid
clid substitute name
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
signaling forward unconditional
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
midcall-signaling passthru
no call service stop
!
!
voice class uri 200 sip
host ipv4:192.168.11.1
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g722-64
!
voice-port 0/0
cptone DE
timeouts interdigit 2
station-id number 624
caller-id enable
!
dial-peer voice 1 voip
description **Extension on VG204 port - 624**
destination-pattern .T
session protocol sipv2
session target ipv4:192.168.11.1:5060
session transport tcp
incoming called-number 624
voice-class codec 1
dtmf-relay rtp-nte
no vad
authentication username vg204001 password 7 XXXX realm 624.192.168.11.1
!
dial-peer voice 624 pots
description **Extension on VG204 port - 624**
destination-pattern 624
authentication username vg204001 password 7 XXXX realm 624.192.168.11.1
port 0/0
!
!
gateway
timer receive-rtp 1200
!
sip-ua
credentials username vg204001 password 7 XXXX realm 624.192.168.11.1
authentication username vg204001 password 7 XXXX
nat symmetric check-media-src
registrar ipv4:192.168.11.1 expires 180
sip-server ipv4:192.168.11.1
!


vg204#debug ccsip messages
SIP Call messages tracing is enabled
vg204#
Mar 16 15:47:45.209: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:vg204001@192.168.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK2DED35769F5335FE
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 88 INVITE
Contact: <sip:6D7CC38E708FFCC74B7AF4099E069@192.168.11.1>
Max-Forwards: 70
P-Called-Party-ID: <sip:**9@fritz.box>
Expires: 120
Session-Expires: 600;refresher=uac
Min-SE: 90
User-Agent: AVM FRITZ!Box 7490 113.07.29 (Oct 26 2021)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 359

v=0
o=user 11974073 11974073 IN IP4 192.168.11.1
s=call
c=IN IP4 192.168.11.1
t=0 0
m=aud
vg204#io 7090 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7091

Mar 16 15:47:45.213: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK2DED35769F5335FE
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>;tag=177F208-4E4
Date: Thu, 16 Mar 2023 15:47:45 GMT
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 88 INVITE
Allow-Events: telephone-event
Min-SE: 1800
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar 16 15:47:45.261: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:vg204001@192.168.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK2DED35769F5335FE
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>;tag=177F208-4E4
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 88 ACK
User-Agent: AVM FRITZ!Box 7490 113.07.29 (Oct 26 2021)
Content-Length: 0


Mar 16 15:47:45.281: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:vg204001@192.168.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK7917E88A18B901A4
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 89 INVITE
Contact: <sip:6D7CC38E708FFCC74B7AF4099E069@192.168.11.1>
Max-Forwards: 70
P-Called-Party-ID: <sip:**9@fritz.box>
Expires: 120
Session-Expires: 1800;refresher=uac
Min-SE: 1800
User-Agent: AVM FRITZ!Box 7490 113.07.29 (Oct 26 2021)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 359

v=0
o=user 11974073 11974073 IN IP4 192.168.11.1
s=call
c=IN IP4 192.168.11.1
t=0 0
m=audio 7090 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7091

Mar 16 15:47:45.297: //197/BA86E6E88167/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK7917E88A18B901A4
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>
Date: Thu, 16 Mar 2023 15:47:45 GMT
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 89 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Mar 16 15:47:45.301: //197/BA86E6E88167/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK7917E88A18B901A4
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>;tag=177F25C-12D6
Date: Thu, 16 Mar 2023 15:47:45 GMT
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 89 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=28
Content-Length: 0


Mar 16 15:47:45.349: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:vg204001@192.168.10.51:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.1:5060;branch=z9hG4bK7917E88A18B901A4
From: "HomeOffice" <sip:**622@fritz.box>;tag=FE04049844DD5779
To: <sip:vg204001@192.168.10.51:5060>;tag=177F25C-12D6
Call-ID: 112BC1E584DB420D@192.168.11.1
CSeq: 89 ACK
User-Agent: AVM FRITZ!Box 7490 113.07.29 (Oct 26 2021)
Content-Length: 0

1 Reply 1

b.winter
VIP
VIP

I don't know, if this works on your VG204 IOS version, but in newer versions for routers, you can also match a dial-peer based on URIs (and not only numbers). Maybe you need to upgrade to the latest version.

If that would be possible, you could match the incoming dial-peer based on the username and with the help of a sip-profile, you manipulate the TO-header, so that you replace the username with the number.

This doc should include everyting you need:
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-inbnd-dp-match-uri.html (advice: bookmark this, you probably will need it again)