Asterisk and sipp UAS

nan128 · June 14, 2016, 11:03am

Hi everyone,

I’m trying to get sipp communicate with Asterisk in order to perform performance tests:
I’ve been through these steps:

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

In extensions.conf

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

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 !

Regards,
Hssan

jcolp · June 14, 2016, 12:10pm

What is the complete console output?

nan128 · June 14, 2016, 12:29pm

[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)

[Jun 14 07:36:56] WARNING[2601][C-00000121]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:56] WARNING[2602][C-00000122]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:56] WARNING[2603][C-00000123]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2604][C-00000124]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2605][C-00000125]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2606][C-00000126]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2607][C-00000127]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2608][C-00000128]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2609][C-00000129]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2610][C-0000012a]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2611][C-0000012b]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2612][C-0000012c]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:57] WARNING[2613][C-0000012d]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2614][C-0000012e]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2615][C-0000012f]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2616][C-00000130]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2617][C-00000131]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2618][C-00000132]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2619][C-00000133]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2620][C-00000134]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2621][C-00000135]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2622][C-00000136]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:58] WARNING[2623][C-00000137]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:59] WARNING[2624][C-00000138]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:59] WARNING[2625][C-00000139]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:59] WARNING[2626][C-0000013a]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:59] WARNING[2627][C-0000013b]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

[Jun 14 07:36:59] WARNING[2628][C-0000013c]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

jcolp · June 14, 2016, 12:33pm

Do “core set verbose 3” and “sip set debug on” and provide the new information with a new attempt.

nan128 · June 14, 2016, 12:41pm

<------------>

    -- Executing [s@test:1] Dial("SIP/sippuac-00011c24",

"SIP/sippuas,20") in new stack

Really destroying SIP dialog

'691de58c404c97734a024cab20b5cd8a@127.0.1.1:5060' Method: INVITE

[Jun 14 09:38:48] WARNING[11843][C-00011c24]: app_dial.c:2437

dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Subscriber absent)

  == Everyone is busy/congested at this time (1:0/0/1)

    -- Auto fallthrough, channel 'SIP/sippuac-00011c24' status is 'CHANUNAVAIL'



<--- Reliably Transmitting (NAT) to 127.0.0.1:5061 --->

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP

127.0.0.1:5061;branch=z9hG4bK-11683-159-0;received=127.0.0.1;rport=5061

From: sipp <sip:sipp@127.0.0.1:5061>;tag=11683SIPpTag00159

To: s <sip:s@127.0.0.1:5060>;tag=as5be8d4b8

Call-ID: 159-11683@127.0.0.1

CSeq: 1 INVITE

Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,

INFO, PUBLISH

Supported: replaces, timer

X-Asterisk-HangupCause: Subscriber absent

X-Asterisk-HangupCauseCode: 20

Content-Length: 0





<------------>



<--- SIP read from UDP:127.0.0.1:5061 --->

ACK sip:s@127.0.0.1:5060 SIP/2.0

Via: SIP/2.0/UDP

127.0.0.1:5061;branch=z9hG4bK-11683-159-0;received=127.0.0.1;rport=5061

From: sipp <sip:sipp@127.0.0.1:5061>;tag=11683SIPpTag00159

To: s <sip:s@127.0.0.1:5060>;tag=as5be8d4b8

Call-ID: 159-11683@127.0.0.1

CSeq: 1 ACK

Contact: <sip:sipp@127.0.0.1:5061;transport=UDP>

Max-Forwards: 70

Subject: Performance Test

Content-Length: 0



<------------->

--- (10 headers 0 lines) ---

Really destroying SIP dialog '159-11683@127.0.0.1' Method: ACK



<--- SIP read from UDP:127.0.0.1:5061 --->

INVITE sip:s@127.0.0.1:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-11683-160-0

From: sipp <sip:sipp@127.0.0.1:5061>;tag=11683SIPpTag00160

To: s <sip:s@127.0.0.1:5060>

Call-ID: 160-11683@127.0.0.1

CSeq: 1 INVITE

Contact: sip:sipp@127.0.0.1:5061

Max-Forwards: 70

Subject: Performance Test

Content-Type: application/sdp

Content-Length: 129



v=0

o=user1 53655765 2353687637 IN IP4 127.0.0.1

s=-

c=IN IP4 127.0.0.1

t=0 0

m=audio 6000 RTP/AVP 0

a=rtpmap:0 PCMU/8000

<------------->

--- (11 headers 7 lines) ---

Sending to 127.0.0.1:5061 (no NAT)

Sending to 127.0.0.1:5061 (no NAT)

Using INVITE request as basis request - 160-11683@127.0.0.1

Found peer 'sippuac' for 'sipp' from 127.0.0.1:5061

  == Using SIP RTP CoS mark 5

Found RTP audio format 0

Found audio description format PCMU for ID 0

Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -

audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0

(nothing), combined - 0x0 (nothing)

Peer audio RTP is at port 127.0.0.1:6000

Looking for s in test (domain 127.0.0.1)

list_route: hop: <sip:sipp@127.0.0.1:5061>



<--- Transmitting (NAT) to 127.0.0.1:5061 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP

127.0.0.1:5061;branch=z9hG4bK-11683-160-0;received=127.0.0.1;rport=5061

From: sipp <sip:sipp@127.0.0.1:5061>;tag=11683SIPpTag00160

To: s <sip:s@127.0.0.1:5060>

Call-ID: 160-11683@127.0.0.1

CSeq: 1 INVITE

Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,

INFO, PUBLISH

Supported: replaces, timer

Contact: <sip:s@127.0.0.1:5060>

Content-Length: 0

<------------>

david551 · June 14, 2016, 9:17pm

Whilst I can’t see anything that would cause it to treat sippuas as down, could you run “sip show peers”.

Also:

insecure=very
canreinvite=no
nat=yes

are all deprecated and I think insecure=verify may now be considered invalid. insecure=invite has no useful purpose if there is no secret, and insecure=port may even cause mis-operation , given that you have more than one port on the same IP address.

Moreover type=friend is not considered best practice. In most cases it causes security breaches and mis-operation. There are very few cases when it is better than type=peer, and you are not using one of those cases.

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