[asterisk-users] Testing Asterisk with media - sipp

Daniel - Asterisk earohuanca at gmail.com
Mon Jul 4 14:29:11 CDT 2011


I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:

This is the command I send at SIPp server:
      ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err

This is the result I see:
      Last Error: Aborting call on unexpected message for Call-Id
'19-12768 at 12...

What I see at sipp's logs:

2011-06-28      14:32:57:624    1309289577.624809: Aborting call on
unexpected message for Call-Id '1-12768 at 127.0.0.1': while expecting '100'
(index 1), received 'SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-12768-1-0;received=192.168.1.253
From: sipp <sip:sipp at 127.0.0.1:5061>;tag=12768SIPpTag091
To: sut <sip:2005 at 192.168.1.18:5060>;tag=as3614adc3
Call-ID: 1-12768 at 127.0.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

This is my asterisk 1.8's configuration:

*sip.conf*
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=ulaw
*
*
*extensions.conf:*
[sipp]
exten => 2005,1,Answer
same=>n,Dial(SIP/intern,30)
same=>n,Hangup()

exten => 2006,1,Answer()
same=> n,WaitMusicOnHold(4)
same=> n,Hangup()


I'm using sipp.3.1.src.tar.gz and I have installed it this way:
..sip.svn# make pcapplay

Thanks in advance.

Elder
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110704/fad35e47/attachment.htm>


More information about the asterisk-users mailing list