SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field issue

[Mohammed Al-odhari]

dear all,

I`m trying to do test sip voip call using sip protocol and i faced sip error, don`t know the cause if it is nat or other.

Scenario is:

ip phone(111)-->VoipGW_1(192.168.2.240)-->internet modem_1(192.168.2.254)--------->internet modem_2(192.168.2.2)-->VoipGW_2(192.168.2.1)

internet mdoem_2 has public ip  82.114.181.235 and nated to VoipGW_2(192.168.2.1).

I want the call which initiated from extension 111 to be received by VoipGW_2 and then being send out of it as configured in the dial-peer 45 (sh run attached in debug files). Unfortunaltely the call is rejected by the VoipGW_2 and doesnt process the call and send it out. Error is shown as 

SIP/2.0 400 Bad Request - 'Malformed/Missing Contact field'
Via: SIP/2.0/UDP 10.101.103.220:16839:5060;

Attached is debug and show run gathered from VoipGW_2 and also show run and dial-peer for VoipGW_1.

please advice.

thanks.

[Chakshu Piplani]

Hi Mohammed,

After reviewing the logs, In the INVITE I see:

Contact: <sip:111@10.101.103.220:16839:5060>

Why do we have two port numbers in contact field?

Working contact field:

Contact: <sip:4153490879@10.1.254.6:5060>  

You can check this post:

https://supportforums.cisco.com/discussion/11689411/normalization-script#3810453

Regards,

Chakshu Piplani

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[Mohammed Al-odhari]

thanks for your replay, while i check the sip and the ip address and i`ll inform u back, i change the call to work with h323 protocol and it get disconnected as show in the debug in VoiceGW_2.(attached)

can you advice.

[Mohammed Al-odhari]

Hi again Chankshu,

i do check the ip address and now it is ok.

I did test calls by calling 967730452427,967730456693 and others using both sip and h323.
in the debug files, i started using sip as show also in the dial peer, it shows the below error.
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 104.251.178.102:5060;
From: <sip:atelecom@104.251.178.102>;

i removed dialpeer 45(sip) and i switch to h323 dialpeer 44 and i did test call with the error
DisconnectText=no route to destination (3) ..
please go thorugh the attached files and tell me where is the error and did the call get moved from VoiceGW_2 to the sim box(192.168.1.104) or no?????????
please advice. (sh call history voice is in the debug ccsip file)
thanks.

[Mohammed Al-odhari]

Hi again,

thanks for your time, can you go please again through the logs and replay to my last post.

regards