Solution to SIP/2.0 484 Address Incomplete message

[bernsplagata]

Hi,,

I am having a problem making outbound calls to my SIP trunk.  Always having the Address incomplete message in my debugs.  Below are my SIP comfiguration. (some commands i think are already redundant)

**voice service voip**

!

voice service voip

qsig decode

dtmf-interworking standard

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service h450.2

no supplementary-service h450.3

supplementary-service h450.12

no supplementary-service sip moved-temporarily

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  bind control source-interface GigabitEthernet0/2

  bind media source-interface GigabitEthernet0/2

  rel1xx disable

  registrar server expires max 6300 min 3600

  transport switch udp tcp

  options-ping 60

  no update-callerid

!

**dial peer** (i have narrowed it down to one number for testing purposes)

dial-peer voice 10 voip

description **SIP TRUNK PEER**

translation-profile outgoing SIP_2

preference 1

destination-pattern 80535665576

session protocol sipv2

session target ipv4:10.200.7.157

session transport udp

voice-class codec 1

voice-class sip rel1xx disable

voice-class sip dtmf-relay force rtp-nte

voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3

voice-class sip bind control source-interface GigabitEthernet0/2

voice-class sip bind media source-interface GigabitEthernet0/2

dtmf-relay rtp-nte

no vad

THanks.

BR,

Bernard

[Hermanus Janse van Vuuren]

show us the debug ccsip as well please.

This is a working config:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip refer

supplementary-service ringback h225-info

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

!        

dial-peer voice 9 voip

destination-pattern 0T

session protocol sipv2

session target ipv4:61.99.235.244

session transport udp

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

!

[bernsplagata]

I started with the config you gave and started adding some more as it did not work.  i need to bind the sip media and control to the interface of the SIP prpvider.  I am also doing ping-options as the sip providers requires it for the hearbeat checking.  Hope the debugs help a bit

See blow debugs:

Jan  6 11:02:29.605: //110134/D7851DEFB4C0/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0535665576@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 172.29.36.46:5060;branch=z9hG4bK161926E0

Remote-Party-ID: "Amr Kassem" <114818300>;party=calling;screen=no;privacy=off

From: "Amr Kassem" <114818300>;tag=18579368-C01

To: <0535665576>

Date: Mon, 06 Jan 2014 11:02:29 GMT

Call-ID: DE784C1C-75F811E3-B4C7A4AC-E44AC759@172.29.36.46

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3615825391-1979191779-3032523948-3830105945

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: 1389006149

Contact: <114818300>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 247

v=0

o=CiscoSystemsSIP-GW-UserAgent 5240 9955 IN IP4 172.29.36.46

s=SIP Call

c=IN IP4 172.29.36.46

t=0 0

m=audio 23076 RTP/AVP 8 101

c=IN IP4 172.29.36.46

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Jan  6 11:02:29.625: //110134/D7851DEFB4C0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.29.36.46:5060;branch=z9hG4bK161926E0

Call-ID: DE784C1C-75F811E3-B4C7A4AC-E44AC759@172.29.36.46

From: "Amr Kassem"<114818300>;tag=18579368-C01

To: <0535665576>

CSeq: 101 INVITE

Content-Length: 0

Jan  6 11:02:29.693: //110134/D7851DEFB4C0/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 172.29.36.46:5060;branch=z9hG4bK161926E0

Record-Route: <10.200.7.157:5060>

Call-ID: DE784C1C-75F811E3-B4C7A4AC-E44AC759@172.29.36.46

From: "Amr Kassem"<114818300>;tag=18579368-C01

To: <0535665576>;tag=sbc0804pst42tcs

CSeq: 101 INVITE

Reason: Q.850;cause=28;text="address incomplete"

Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"

Content-Length: 0

Jan  6 11:02:29.697: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:0535665576@10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 172.29.36.46:5060;branch=z9hG4bK161926E0

From: "Amr Kassem" <114818300>;tag=18579368-C01

To: <0535665576>;tag=sbc0804pst42tcs

Date: Mon, 06 Jan 2014 11:02:29 GMT

Call-ID: DE784C1C-75F811E3-B4C7A4AC-E44AC759@172.29.36.46

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Jan  6 11:02:34.141: //110135/000000000000/SIP/Msg/ccsipDisplayMsg:

Sent:

OPTIONS sip:10.200.7.157:5060 SIP/2.0

Via: SIP/2.0/UDP 172.29.36.46:5060;branch=z9hG4bK16193266D

From: <172.29.36.46>;tag=1857A524-1D35

To: <10.200.7.157>

Date: Mon, 06 Jan 2014 11:02:34 GMT

Call-ID: E12D0C87-75F811E3-B4C8A4AC-E44AC759@172.29.36.46

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

Max-Forwards: 70

CSeq: 101 OPTIONS

Contact: <172.29.36.46:5060>

Content-Length: 0

[Hermanus Janse van Vuuren]

What does the provider expect from you how many digits should you send them and tcp or udp?  I see you have a switch udp tcp command but your dialpeer you sent via udp, what does the ISP expect from you?

[bernsplagata]

In the configuration guide they sent it should be 7 to 10 digits.  Also session transport is UDP.

[Hermanus Janse van Vuuren · Accepted solution]

here is a thread with someone who had more or less the same issue as you have.

Calling number and called number that ISP is expecting is very important.

https://supportforums.cisco.com/thread/2241718

Cause No. 28 - invalid number format (address incomplete)  [Q.850]

This cause indicates that the called party cannot be reached because  the called party number is not in a valid format or is not complete.

[Manish Prasad]

This generally happens when you send incomplete "URI" or "TO" in the INVITE message.

Please also post your translation rules for SIP_2.

Thanks

Manish

[bernsplagata]

Hi Manish,

See below trnaslation profile:

!

voice translation-profile SIP_2

translate calling 3

translate called 12

!

!

voice translation-rule 12

rule 1 /^8/ //

!

voice translation-rule 3

rule 1 /^.../ /114818300/

!

Only strip 8 and change internal ext number to the DID range provided by the SIP provider.

[Suresh Subramanian · Accepted solution]

Hi Bernard,

could you please confirm whether the ITSP is expecting 9 digits calling number (114818300) and 10 digit called number (0535665576)?

From: "Amr Kassem" <114818300>;tag=18579368-C01

To: <0535665576>

[bernsplagata]

i will check this with the ITSP and will came back once i get the answers.

[bernsplagata]

the thread nailed. It.  Thanks Hermanush and Manish and Suresh!!!

[Suresh Subramanian]

Hi Bernard, good to hear the issue is fixed. what did you change to make it working?

[bernsplagata]

Hi Suresh,

https://supportforums.cisco.com/thread/2241718 thi forum helped.  So basically i changed the translation-profile for the

!

voice translation-rule 3

rule 1 /^.../ /114818300/

!

i just removed the 11 as the ITSP is only accepting 7 digits calling number.  but i think this will dpepend on every ITSP.

So now the ITSP is seeing 4818300 as the calling number and they are the ones adding the area code which is 011

Cheers

[Suresh Subramanian]

Great, Thanks for the update Bernand. Cheers