Thank you I have also tried those settings. The main thing is coming from my voip provider all I am doing is bridging the calls to two other devices (1 trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with an IVR and when I call in I can not hear the IVR. However if I call directly to a SIP client the person answering the SIP phone can hear me but I can not hear them at all. Its definately not a NAT issue which is what makes it even more confusing. When the call is in place a sip show channels shows me both lefs of the call and they are both using either alaw or ulaw so it should not be a codec translation issue either.
On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel <[email protected]> wrote: > What about your sip clients? Are they on public network? > > Try on sip.conf > > Nat=no/yes > > conreinvite=yes/no > > -- > Sent from my iPhone > > On Mar 9, 2011, at 6:11 PM, Tim King <[email protected]> wrote: > > IPTBALES is off and I have all firewalls disabled. All network elements > currently involved have public IP's assigned to them. My main asterisk box > has a public IP. I have multiple trunks to voip peers for inbound and > outbound calls which are also all public IP's. My two clients are trunked > via IAX and one is a Trixbox and the other is a digium AA50 which both also > have public IP's assigned to them. > > On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime < <[email protected]> > [email protected]> wrote: > >> How is your network is organized? Is your server behind a firewal, about >> iptables ? >> >> >> >> >> On Wed, Mar 9, 2011 at 5:40 PM, Tim King < <[email protected]> >> [email protected]> wrote: >> >>> I am having trouble with no return audio on inbound calls. I have been >>> working on this for 18 hours and even built a fresh server and moved >>> everything over and I am getting the same results. I need someone that can >>> help get this resolved tonight. I can provide access to all machines >>> involved. >>> >>> Please email me at <[email protected]>[email protected] >>> you can help. >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by <http://www.api-digital.com> >>> http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> <http://www.asterisk.org/hello> >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> <http://lists.digium.com/mailman/listinfo/asterisk-users> >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> -- >> *Adolphe CHER-AIME >> Network / VoIP Engineer >> CCNA, CCNA VOICE, Global VSAT Forum Certified >> (509) 3449-4280* >> > > -- > _____________________________________________________________________ > -- 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> > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > <http://lists.digium.com/mailman/listinfo/asterisk-users> > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
