first I would check if the network parameters(IP and RTP ports) of your flow
are matching the ones negotiated in SIP SDP....
this also remembers me that the first time I used the default pcap payload I
noticed it is very low, I barely could hear it and I couldnt even understand
what the person was saying.... so be sure it's not just a matter of volume
here....
On Wed, Jul 6, 2011 at 7:48 AM, Daniel - Asterisk <earohua...@gmail.com>wrote:
> I get this fixed when codec matching was done at both peers.
>
> But I still can't hear anything despite I can see rtp flowing, any advice?
>
> Elder
>
> 2011/7/5, Patrick Wakano <pwak...@gmail.com>:
> > strange....
> > maybe your asterisk is configured without support for alaw, which seems
> to
> > be the codec used by the default pcap script...
> > check the allow/disallow fields of your sip.conf
> >
> >
> >
> >
> > On Mon, Jul 4, 2011 at 4:32 PM, Daniel - Asterisk
> > <earohua...@gmail.com>wrote:
> >
> >> Hello everyone, I need some help to get this working.
> >>
> >> I'm trying to get working SIPp with media but something is wrong (it's
> >> working well without media)
> >>
> >> 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@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@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@127.0.0.1:5061>;tag=12768SIPpTag091
> >> To: sut <sip:2005@192.168.1.18:5060>;tag=as3614adc3
> >> Call-ID: 1-12768@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
> >>
> >>
> >>
> ------------------------------------------------------------------------------
> >> All of the data generated in your IT infrastructure is seriously
> valuable.
> >> Why? It contains a definitive record of application performance,
> security
> >> threats, fraudulent activity, and more. Splunk takes this data and makes
> >> sense of it. IT sense. And common sense.
> >> http://p.sf.net/sfu/splunk-d2d-c2
> >> _______________________________________________
> >> Sipp-users mailing list
> >> Sipp-users@lists.sourceforge.net
> >> https://lists.sourceforge.net/lists/listinfo/sipp-users
> >>
> >>
> >
>
> --
> Enviado desde mi dispositivo móvil
>
------------------------------------------------------------------------------
All of the data generated in your IT infrastructure is seriously valuable.
Why? It contains a definitive record of application performance, security
threats, fraudulent activity, and more. Splunk takes this data and makes
sense of it. IT sense. And common sense.
http://p.sf.net/sfu/splunk-d2d-c2
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
