[Asterisk-Users] Asterisk ---> Cisco AS5XXX sip one way audio
Jorge Verastegui G
jorge at redcetus.com
Tue Dec 7 12:06:10 MST 2004
Hi,
I changed the bindaddr in sip.conf, still one way audio.
this did not work
This is the cisco config
voice service voip
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
sip
voice class codec 11
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 gsmfr
codec preference 4 g726r32
codec preference 6 g726r16
codec preference 7 g723r63
codec preference 8 g723r53
codec preference 9 g726r24
codec preference 10 g723ar63
codec preference 11 g723ar53
codec preference 12 g711ulaw
codec preference 13 g711alaw
codec preference 14 clear-channel
no voice hpi capture buffer
no voice hpi capture destination
fax interface-type fax-mail
mta receive maximum-recipients 0
controller E1 7/0
framing NO-CRC4
line-termination 75-ohm
ds0-group 0 timeslots 1-15,17-31 type r2-digital r2-compelled
cas-custom 0
country bolivia
no ip http server
voice-port 7/0:0
dial-peer voice 4444 voip
destination-pattern 44T
voice-class codec 11
session protocol sipv2
session target ipv4:AAAA.BBBB.CCCC.DDD
On Mon, 2004-12-06 at 20:33, Julio Tejera wrote:
> check "bindaddr" at [general] on sip.conf
> that hapen to me and I solved it putting
> a bindaddr instead of bindaddr=0.0.0.0
>
> HTH
>
> -------
> Ing. Julio Alvarez Tejera
> Unix Trends
> *BSD, Solaris & Linux
> ---------------
> "extremely stable systems"
> ----- Original Message -----
> From: "Jorge Verastegui G" <jorge at redcetus.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Monday, December 06, 2004 6:11 PM
> Subject: [Asterisk-Users] Asterisk ---> Cisco AS5XXX sip one way audio
>
>
> > Hi,
> >
> > I trying to use asterisk for
> >
> > PSTN(A)----> Cisco AS5xxx ----> ASterisk---->PSTN(B)
> >
> > (No Nat no Firewall)
> >
> > I hear (on the PSTN(A)) clearly what the other person is saying, but the
> > other person (on the PSTN(B) side) hears nothing from PSTN(A).
> >
> > I use tcpdump for debug de rtp trafic, and ouput contains only trafic
> > from Asterisk to Cisco AS5XXX
> >
> > The sip.conf configuration contains
> >
> > [cisco1]
> > type=friend
> > host=XXX.YYY.ZZZ.VV
> > dtmfmode=inband
> > insecure=yes
> > insecure=very
> > context=fromsip
> > reinvite=no
> > canreinvite=no
> > disallow=all
> > allow=g729
> > allow=ulaw
> > allow=alaw
> >
> > I tried to search the internet for the message, but I got no results
> >
> > Please help me
> >
> > --
> > Jorge Verastegui G <jorge at redcetus.com>
> > RedCetus S.R.L.
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
>
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