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I'm trying to get sipp communicate with Asterisk in order to perform performance tests:

I've been through these steps:

1) In sip.conf

[sippuac]
type=friend
username=sippuac
host=127.0.0.1
port=5061
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes

[sippuas]
type=friend
username=sippuas
host=127.0.0.1
port=5062
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes

2) In extensions.conf

[test]
exten=>s,1,Dial(SIP/sippuas,20)

3) Running SIPp

sipp -sn uas -rsa 127.0.0.1:5060 -p 5062 -i 127.0.0.1 -mp 6001

sipp -sn uac 127.0.0.1:5060 -s s -p 5061 -i 127.0.0.1

Finally I get on Asterisk :

[Jun 14 07:36:56] WARNING[2600][C-00000120]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

How can I solve this and make the UAS receive the calls ?

Thanks for your help !

Striker
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2 Answers2

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I think in sip.conf should be type=peer for sippuas.

It is bad idea to run performance test from localhost. SIPP will impact performance of Asterisk. Additionally it make seance to run scenario with audio and I will recommend just answer a call on Asterisk and play some sound, it is not so important to send call out to second sipp.

os11k
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0

For receiving calls from asterisk, SIPp user(s) should be registered first to it. You can see if your user is registered or not by using the command:

sip show peers

in the asterisk CLI. If your uas is not registered and you are trying to tell asterisk to dial to a client and not giving the address of it. There are simple xml examples in this link for how to register and make calls to asterisk. Please follow the scenario you want.

Abhishek Kumar
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