Let me explain what i tried, I registered IP Phone to Asterisk IPPBX with
username & password and from SIPP I executed the UAC mode to reach the IP
Phone via Asterisk PBX. I am attaching the XML file I tried with my IP phone
registered to PBX.
On Fri, Aug 19, 2011 at 5:57 PM, kalyan chowdary <[email protected]>wrote:
> Polycom System Type: VSX 8000 #1.
> I am not poviding any username while calling just giving IP address.
>
> If i have to give any usename as u said can u plz tell me clearly where i
> have to give and what i have to give?
>
> Thanks,
> Kalyan.
>
>
>
> On Fri, Aug 19, 2011 at 5:39 PM, Gopal krishnan <
> [email protected]> wrote:
>
>> I think the username you are using "service" seems to be giving problem
>> for you. Change that to some extension number like 2001 which your polycom
>> phone's extension that has been registered to a PBX.
>>
>> On Fri, Aug 19, 2011 at 12:44 PM, kalyan chowdary
>> <[email protected]>wrote:
>>
>>> Hi,
>>>
>>> I am tying to call to polycom test machines with sip-tester(Sipp) using
>>> sipp -sn uac 140.242.250.200 -m 1 -tace_calldebug
>>>
>>> I am not getting any response from that machine but when i call from
>>> LifeSize unit(sip call),call established.
>>>
>>> Here is my debug information::::
>>> Call debugging information for call [email protected]:
>>> 2011-08-19 02:11:09:120 1313734269.120520 Starting call
>>> [email protected]
>>> 2011-08-19 02:11:09:120 1313734269.120596 Processing message 0 of
>>> type 2 for call [email protected] at 102.
>>> 2011-08-19 02:11:09:120 1313734269.120637 Sending UDP message for
>>> call [email protected] (index 0, hash 561789690):
>>> INVITE sip:[email protected]:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 127.0.1.1:5061;branch=z9hG4bK-14025-1-0
>>> From: sipp <sip:[email protected]:5061>;tag=14025SIPpTag001
>>> To: sut <sip:[email protected]:5060>
>>> Call-ID: [email protected]
>>> CSeq: 1 INVITE
>>> Contact: sip:[email protected]:5061
>>> Max-Forwards: 70
>>> Subject: Performance Test
>>> Content-Type: application/sdp
>>> Content-Length: 129
>>>
>>> v=0
>>> o=user1 53655765 2353687637 IN IP4 127.0.1.1
>>> s=-
>>> c=IN IP4 127.0.1.1
>>> t=0 0
>>> m=audio 6000 RTP/AVP 0
>>> a=rtpmap:0 PCMU/8000
>>>
>>>
>>> Can any one plz tell where i am doing wrong?
>>>
>>> Thanks
>>> kalyan
>>>
>>>
>>>
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>>
>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!--./sipp -i 192.168.0.83 -p 6060 -sf client1.xml -d 20000 192.168.0.58 -trace_err -->
<scenario name="UAClient">
<send retrans="500">
<![CDATA[
INVITE sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:2001@[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=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
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>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:2001@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:2001@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
<pause milliseconds="5000"/>
</scenario>------------------------------------------------------------------------------
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