Daniel - Asterisk
2011-Jul-04 19:29 UTC
[asterisk-users] Testing Asterisk with media - sipp
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
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488 means no mutually acceptable codecs were negotiated between the endpoints. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk <earohuanca at gmail.com> wrote:> 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 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110704/cdce007b/attachment.htm>