Jorge Verastegui G
2004-Dec-06 17:11 UTC
[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@redcetus.com> RedCetus S.R.L.
Julio Tejera
2004-Dec-06 17:33 UTC
[Asterisk-Users] Asterisk ---> Cisco AS5XXX sip one way audio
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@redcetus.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@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@redcetus.com> > RedCetus S.R.L. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Jorge Verastegui G
2004-Dec-07 12:06 UTC
[Asterisk-Users] Asterisk ---> Cisco AS5XXX sip one way audio
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@redcetus.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@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@redcetus.com> > > RedCetus S.R.L. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users