Hi, I already have a G729 license and the voice is OK also I tried to use G729 codecs on SIP and AS5400 side, without any g729 license and works fine with a pass-trought configuration, so you only need to check out voip-info.org to solve that issue. on the stun and NAT part, I have not reached to that part yet
On Wed, 2005-06-01 at 21:06 -0500, Obaid Siddiqui wrote: > Hi, > I am using the exact scenario you are using (As5400-asterisk-ata's). But I > have not reached to fax > issue yet. > I tried codec g729 for voice call, it is not working. I think you have to > buy g729 from digium. > > Since both my Asterisk and SIP clients are in different NATS , I have to do > the painful part of port forwarding for every client. Specially if I have > more then one client on same NAT, what needs to be done. > > Voip-wiki suggest using SER or STUN. Can SER and MyStun can be installed on > same Server where Asterisk is residing. > > Will somebody suggest anything. > > Regards, > OMS > Prizm Communications > > > > ----- Original Message ----- > From: "Ren� Mayorga" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Wednesday, June 01, 2005 12:05 PM > Subject: [Asterisk-Users] Fax and codecs preferences to PSTN > > > Hi, > I have an asterisk running with a passtrought conf with G729, > when I try to send a fax from SIP to SIP the ATAs make a good codec > negociation and the fax transmicion is OK, > > But when I try to send the fax to PSTN fax machine > (SIP --> AS5400 --> PSTN) > The ATA Device try to send the RTP with G711ulaw and the Cisco keep > answereing with G729 > a snip some part of my confs. > > < sip.conf > > [general] > port=5060 > bindaddr=0.0.0.0 > context=default > allow=g729 > . > . > . > [22194007] > type=friend > host=dynamic > secret=22194007 > canreinvite=yes > callerid=ATA Sipura FAX <22194007> > . > . > . > > [as5400] > type=friend > host=XXX.XXX.XXX.XXX > canreinvite=yes > insecure=yes > insecure=very > qualify=yes > > </sip.conf> > > > <as5400> > > dial-peer voice 999001 pots > description PRUEBAS SIP > max-conn 3 > destination-pattern 65732.% > progress_ind alert enable 8 > port 7/5:D > ! > dial-peer voice 999000 voip > description PRUEBAS SIP > destination-pattern 2219400. > session protocol sipv2 > session target sip-server > dtmf-relay h245-alphanumeric > fax-relay ecm disable > fax rate 9600 > fax nsf 000000 > fax protocol pass-through g711alaw > no vad > > </as5400> > > Thanks in advance -- Ren� Mayorga Internet & Data El Salvador Telecom S.A. de S.V. Tel:(503) 2247-7246 (503) 2247-7156 Cel:(503) 7962-8205 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
