In face I have one IP extension configured in the Asterisk based hardware
box, still I am getting the error.
On Thu, Jan 6, 2011 at 3:05 PM, Gopalakrishnan A.N <[email protected]> wrote:
> 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]>
>
>
>
--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:[email protected] <sip%[email protected]>
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