01-06-2014 03:23 AM - edited 03-16-2019 09:06 PM
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
Solved! Go to Solution.
01-06-2014 03:57 AM
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-C01114818300>
To: <0535665576>0535665576>
01-06-2014 04:04 AM
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.
01-06-2014 03:32 AM
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
!
01-06-2014 03:40 AM
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=off114818300>
From: "Amr Kassem" <114818300>;tag=18579368-C01114818300>
To: <0535665576>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>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-C01114818300>
To: <0535665576>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>10.200.7.157:5060>
Call-ID: DE784C1C-75F811E3-B4C7A4AC-E44AC759@172.29.36.46
From: "Amr Kassem"<114818300>;tag=18579368-C01114818300>
To: <0535665576>;tag=sbc0804pst42tcs0535665576>
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-C01114818300>
To: <0535665576>;tag=sbc0804pst42tcs0535665576>
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=1857A524-1D35172.29.36.46>
To: <10.200.7.157>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>172.29.36.46:5060>
Content-Length: 0
01-06-2014 03:52 AM
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?
01-06-2014 03:55 AM
In the configuration guide they sent it should be 7 to 10 digits. Also session transport is UDP.
01-06-2014 04:04 AM
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.
01-06-2014 03:37 AM
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
01-06-2014 03:42 AM
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.
01-06-2014 03:57 AM
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-C01114818300>
To: <0535665576>0535665576>
01-06-2014 04:09 AM
i will check this with the ITSP and will came back once i get the answers.
01-06-2014 05:01 AM
the thread nailed. It. Thanks Hermanush and Manish and Suresh!!!
01-06-2014 05:31 AM
Hi Bernard, good to hear the issue is fixed. what did you change to make it working?
01-06-2014 05:41 AM
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
01-06-2014 05:46 AM
Great, Thanks for the update Bernand. Cheers
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