You can try following steps. it worked for me.
1)Register SIPP as UAS to asterisk. can take reference of following scenarios.
command: sipp -i 172.17.1.82 172.17.1.82 -sn uas -sf register_without_Auth.xml
-inf register_without_Auth.inf -m 1 Scenario for
register:register_without_Auth.xml<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Basic Sipstone UAC">
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
Max-Forwards: 70
To: [field0] <sip:[field0]@[field1]>
From: [field0] <sip:[field0]@[field1]>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Contact: <sip:[field0]@[local_ip]:[local_port]>;expires=300
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: SIPp [sipp_version]
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
Max-Forwards: 70
To: [field0] <sip:[field0]@[field1]>
From: [field0] <sip:[field0]@[field1]>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: SIPp [sipp_version]
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>INF file can have following contentSEQUENTIAL
sipp;172.17.1.82;
sip.conf at asterisk:[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
extention.conf at asterisk:
[default]
exten => sipp,1,Dial(SIP/sipp)
With above u will able register sipp to asterisk.
2) Now run SIPP as UAS scenario, which will receve invite and then play .pacp
file attached in last mail.
Command:sipp -i 172.17.1.82 172.17.1.82 -sn uas -sf test.xml -m 1
Scenario:test.xml<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true" rrs="true">
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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 [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="g711a.pcap"/>
</action>
</nop>
<pause milliseconds="90000"/>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- 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"/>
</scenario>
This will wait for invite from other end and play .pcap during call.
My Setup:
Microsip(UAC)---------->Asterisk-------->Sipp(UAS)
Microsip dials Sipp(extension registered), SIPP(UAS) recive the invite from
microsip then it plays .pcap in call.
Best Regards,Sakharam Thorat.
Date: Mon, 28 Apr 2014 12:24:54 +0200
Subject: Re: [Sipp-users] Problem connecting UAC and UAS via Asterisk
From: marija.srbl...@gmail.com
To: sakharam.tho...@outlook.com
I think that this will not work for me.
My architecture looks like this:
UAC (192.168.0.159) ---------- asterisk (192.158.0.161) ----------- UAS
(192.168.0.158)
Now I would like to set up the UAS to listen for calls from UAC.
Something like this:
on UAS: ./sipp -sn uas
on UAC: ./sipp -sn uac -d 10000 -s 2005 192.168.0.158
But how do I tell UAC to send calls on UAS via Asterisk?
Best regards,
Marija
On Fri, Apr 25, 2014 at 3:07 PM, Sakharam Thorat <sakharam.tho...@outlook.com>
wrote:
I tried with following it is working for me.
sipp -i 172.17.1.82 172.17.1.82 -sn uac -s 1000 -sf sipp_uac_pcap_g711a.xml -m
1 -l 1 -trace_msg -trace_err
-i 172.17.1.82 - local sipp ip
172.17.1.82 - asterisk server ip
-s 1000 -extension where sip soft phone is registered .
-sf sipp scenario file which used to invite 1000 softphone
-m max no of call.
-i max no of simultaneous call
sip.conf
[1000]
secret=2000
type=friend ; Can make inbound and outbound calls
host=dynamic ; This device needs to register
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
extensions.conf
[sipp]
exten => 1000,1,Dial(SIP/1000)
sipp playing .pcap in scenario, please find attachment for sipp scenario and
.pcap
set up -
sipp(sipp@172.17.1.4)
-------------->asterisk(172.17.1.82)----------->microsip(1000@172.17.1.4)(app
installed on windows pc)
Best Regards,
Sakharam Thorat.
Date: Fri, 25 Apr 2014 13:02:07 +0200
From: marija.srbl...@gmail.com
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] Problem connecting UAC and UAS via Asterisk
Hello everyone,
I am trying to connect UAC and UAS via Asterisk server.
I have the Asterisk server on one computer, and two computers for UAC and UAS.
I made an extension 2005 on Asterisk, made changes to the sip.conf files:
[sipp]
type=friend
context=sipp
host=dynamic
port=5060
user=sipp
canreinvite=no
disallow=all
allow=all
made changes to the extensions.conf:
[sipp]
exten => 2005,1,Answer
exten => 2005,2,SetMusicOnHold(default)
exten => 2005,3,WaitMusicOnHold(20)
exten => 2005,4,Hangup
and everything works via Asterisk when I use just UAC.
For example this will work:
./sipp -sn uac -d 10000 -s 2005 ip_address_asterisk
Can anyone explain how to connect UAC and UAS via Asterisk?
I don't know what to type in the command line, and how to tell SIPp to use
Asterisk server when connecting UAC and UAS.
Do I have to make additional changes on extensions.conf and sip.conf files?
Regards,
Marija
------------------------------------------------------------------------------
Start Your Social Network Today - Download eXo Platform
Build your Enterprise Intranet with eXo Platform Software
Java Based Open Source Intranet - Social, Extensible, Cloud Ready
Get Started Now And Turn Your Intranet Into A Collaboration Platform
http://p.sf.net/sfu/ExoPlatform
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
------------------------------------------------------------------------------
"Accelerate Dev Cycles with Automated Cross-Browser Testing - For FREE
Instantly run your Selenium tests across 300+ browser/OS combos. Get
unparalleled scalability from the best Selenium testing platform available.
Simple to use. Nothing to install. Get started now for free."
http://p.sf.net/sfu/SauceLabs
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
