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SIP incoming/outgoing call failed. Reason: Q.850;cause=28;text="address incomplete"

mohsin majeed
Level 2
Level 2

Hi Experts,

I am facing an issue where i am going to add a new sip range on the same physical trunk.

We already had two ranges 2835XXX, 2438XXX and there is one more new range need to be added.

Scenario is Phones(SIP/SCCP) ----> CUCM ---siptrunk---> UBE ---sip---> ITSP

Newly range is 4945XXX. Necessary configuration is done CM and UBE.

Outgoing call failed with "Call cannot be complete as dialed, this is the recording"

Incoming call failed with a message "not a valid number"

 

Calling number 4945001

Called number 0501029946

 

debug ccsip messages

debug voip ccapi inout

sh run 

are attached.

 

please help

 

 

Regards,

Mohsin Majeed

5 Replies 5

Nadeem Ahmed
Cisco Employee
Cisco Employee

Hi Mohsin,

 

By checking the logs looks like we are sending wrong/incomplete number to provider thats why they are sending 484.

 

Here you receive 484 from provider.

 

077838: Sep  1 19:42:50.691: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 10.66.7.126:5060;branch=z9hG4bK7AAC52192

Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>

Call-ID: 76BCDCD9-501811E5-8181A4F2-85CD1AE5@10.66.7.126

From: "Test11"<sip:4945001@10.66.7.126>;tag=42F41C24-873

To: <sip:0501029946@10.200.7.157>;tag=sbc0802auud4e7b

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

 

077839: Sep  1 19:42:50.695: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 484 Address Incomplete

Via: SIP/2.0/TCP 172.16.200.20:5060;branch=z9hG4bK24638b7f7e7c

From: "Test11" <sip:4945001@172.16.200.20>;tag=173064~7a4acf75-5f09-4773-8466-a6fba898a872-46739733

To: <sip:00501029946@10.1.0.4>;tag=42F41C6C-2037

Date: Tue, 01 Sep 2015 19:42:50 GMT

Call-ID: 9f901e80-5e51ffba-14df9-14c810ac@172.16.200.20

CSeq: 101 INVITE

Allow-Events: kpml, telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=28

Content-Length: 0

 

 

Subsequently CUBE is sending 484 to CUCM.This 484 message was sent to CUCM which caused the CUCM ANN prompt to play to the ip phone
user.

 

Calling party number:4945001

Called party number:00501029946

 

Is this the correct calling and called party if not you can translate on GW and then send what exactly provider is looking for. If everything is correct then probably you can check with your provider on this.

 

But looks its a issue with the number.

 

Br,

nadeem

 

 

Br, Nadeem Please rate all useful post.

Kunal Narula
Level 1
Level 1

Hi Mohsin,

 

As the cause value suggest 484 Address incomplete is seen when called number routing is not configured at ITSP/Telco end. Can you verify if called /calling number routing is configured for the digits we are sending to them?

Also verify what cause value does PSTN see at their end? Is it 47? if so then verify codec configured at their end?

Incase it is different then we need to configure Xcoder/mtp?

 

Snippet from logs:

=============

 

INVITE sip:00501029946@10.1.0.4:5060 SIP/2.0

Via: SIP/2.0/TCP 172.16.200.20:5060;branch=z9hG4bK24638b7f7e7c

From: "Test11" <sip:4945001@172.16.200.20>;tag=173064~7a4acf75-5f09-4773-8466-a6fba898a872-46739733

To: <sip:00501029946@10.1.0.4>

Date: Tue, 01 Sep 2015 19:42:50 GMT

Call-ID: 9f901e80-5e51ffba-14df9-14c810ac@172.16.200.20

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM9.1

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

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback,X-cisco-original-called

Call-Info: <sip:172.16.200.20:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Cisco-Guid: 2677022336-0000065536-0000085445-0348655788

Session-Expires:  1800

P-Asserted-Identity: "Test11" <sip:4945001@172.16.200.20>

Remote-Party-ID: "Test11" <sip:4945001@172.16.200.20>;party=calling;screen=yes;privacy=off

Contact: <sip:4945001@172.16.200.20:5060;transport=tcp>

Max-Forwards: 70

Content-Type: application/sdp

Content-Length: 423

 

v=0

o=CiscoSystemsCCM-SIP 173064 1 IN IP4 172.16.200.20

s=SIP Call

c=IN IP4 172.16.3.158

b=TIAS:64000

b=AS:64

t=0 0

m=audio 19490 RTP/AVP 8 9 18 0 116 101

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:9 G722/8000

a=ptime:20

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:116 iLBC/8000

a=ptime:20

a=maxptime:60

a=fmtp:116 mode=20

a=rtpmap:18 G729/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Outgoing dial-peer configuration:

=========================

dial-peer voice 802 voip
 description ** SIP TO STC **
 translation-profile outgoing OUT
 destination-pattern .T
 rtp payload-type cisco-codec-fax-ack 98
 rtp payload-type nte 97
 session protocol sipv2
 session target ipv4:10.200.7.157:5060
 session transport udp
 dtmf-relay sip-notify rtp-nte sip-kpml
 codec g711alaw

Incoming translation rule/dial-peer

==========================

dial-peer voice 202 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIPINCOMING283
 translation-profile outgoing REDIAL283
 preference 1
 destination-pattern 5...
 session protocol sipv2
 session target ipv4:172.16.200.13
 incoming called-number 2835...$
 voice-class codec 1
 dtmf-relay sip-notify rtp-nte sip-kpml
 no vad
!
dial-peer voice 301 voip
 description ** SIP INCOMING FROM STC **
 translation-profile incoming SIPINCOMING243
 translation-profile outgoing REDIAL243
 destination-pattern 8...
 session protocol sipv2
 session target ipv4:172.16.200.20
 incoming called-number 2438...$
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay sip-notify rtp-nte sip-kpml
 no vad


voice translation-profile SIPINCOMING243
 translate called 152
!
voice translation-rule 152
 rule 1 /^2438/ /8/
 rule 2 /^12438/ /8/
 rule 3 /^112438/ /8/
 rule 4 /^0112438/ /8/

voice translation-profile SIPINCOMING283
 translate called 52

voice translation-rule 52
 rule 1 /^2835/ /5/
 rule 2 /^12835/ /5/
 rule 3 /^112835/ /5/
 rule 4 /^0112835/ /5/

I also see a significant change between configuration of 2835XXX/ 2438XXX and 4945XXX. In case of later, you don't have a customized incoming dial-peer that translation rule reducing seven digit number to 4 digits. Is there a change in calling number requirement from telco?

Thanks!

Kunal

Please rate all helpful post

Hi,

 

Your ITSP is rejecting the call and not accepting the called number which you are sending.

 

1. Make sure that you are sending the called number in the expected format as needed by provider.

2. Make sure that the provider isn't having problem in routing calls from your SIP trunk

Wilson Samuel
Level 7
Level 7

Hi,

Just adding to others point. As per the observations from the debugs it appears that:

 

1. Calling Number is being sent as 494-5001

2. Called Number is 050-102-9946

May be the STC is expecting Calling Number  in the National Number, hence please try sending the Calling Number as 01X-494-5001 ?

HTH

 

mohsin majeed
Level 2
Level 2

Thank you all for your valuable information. I will visit the site and will test as suggest by you all. I will keep posting updates.

 

Thanks you so much

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