Thanks for all your comments, the destination device what I am using is a
Asterisk based IPPBX a small hardware PBX.
Hi Garron, As per your comments I understand that I have to create one IP
extension in the hardware box. Please correct me if i m wrong.
On Thu, Jan 6, 2011 at 12:26 AM, Nhat Phan <[email protected]> wrote:
> The destination phone is a real IP phone?
>
>
>
> I think that server could not find the IP of the destination phone because
> it haven’t registered yet. Then you will receive 404 not found.
>
>
>
> The destination phone should be registered to your SIP server.
>
>
>
> Regards,
>
> Nhat Phan
>
> *From:* Gopalakrishnan A.N [mailto:[email protected]]
> *Sent:* Wednesday, January 05, 2011 12:50 PM
> *To:* [email protected]
> *Subject:* [Sipp-users] SIP error 404
>
>
>
> I am trying to build the basic setup, I did but when I try to run my first
> script I am facing error like
>
>
>
> Aborting call on unexpected message for Call-Id '[email protected]':
> while expecting '100' (index 1), received 'SIP/2.0 404 Not Found
>
>
>
> *My xml file as*
>
> <scenario name="Basic Sipstone UAC">
>
>
>
> <send>
>
> <![CDATA[
>
>
>
> INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port]
>
> From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
>
> To: sut <sip:[servi...@[remote_ip]:[remote_port]>
>
> Call-ID: ABCDEFGHIJ///[call_id]
>
> Cseq: 1 INVITE
>
> Contact: sip:s...@[local_ip]:[local_port]
>
> Max-Forwards: 70
>
> Subject: Performance Test
>
> Content-Type: application/sdp
>
> Content-Length: [len]
>
>
>
> v=0
>
> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>
> s=-
>
> t=0 0
>
> c=IN IP[media_ip_type] [media_ip]
>
> m=audio [media_port] RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> ]]>
>
> </send>
>
> <recv response="100" optional="true">
>
> </recv>
>
>
>
> <recv response="180" optional="true">
>
> </recv>
>
>
>
> <recv response="200">
>
> </recv>
>
>
>
> <send>
>
> <![CDATA[
>
>
>
> ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port]
>
> From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
>
> To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param]
>
> Call-ID: ABCDEFGHIJ///[call_id]
>
> Cseq: 1 ACK
>
> Contact: sip:s...@[local_ip]:[local_port]
>
> Max-Forwards: 70
>
> Subject: Performance Test
>
> Content-Length: 0
>
>
>
> ]]>
>
> </send>
>
>
>
> <send retrans="500">
>
> <![CDATA[
>
>
>
> BYE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port]
>
> From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
>
> To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param]
>
> Call-ID: ABCDEFGHIJ///[call_id]
>
> Cseq: 2 BYE
>
> Contact: sip:s...@[local_ip]:[local_port]
>
> Max-Forwards: 70
>
> Subject: Performance Test
>
> Content-Length: 0
>
>
>
> ]]>
>
> </send>
>
>
>
> <recv response="200">
>
> </recv>
>
> </scenario>
>
>
>
> Any assistance would be appreciated.
>
>
> --
> Thank you with regards,
> Gopalakrishnan A.N.
>
> VoIP call - sip:[email protected] <sip%[email protected]>
>
>
>
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:[email protected] <sip%[email protected]>
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