I am trying to setup one asterisk with voice over G729 and fax G711 using a local sip provider using direct rtp. My fax is connected in a ATA Linksys SPA 8000.
I have one issue regarding incoming fax calls, when my ATA is detecting the FAX tone:
SIP Provider invite(G729,G711)-> Asterisk invite (G729) ->ATA OK(G729) -> Asterisk
than the ATA detect the fax:
ATA Invite(G711) ->Asterisk ok(G711)->ATA
Asterisk is not sending the re-invite to G711 to the sip provider, and i do have now 2 leg one G729 and another one G711
ubuntu*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
xx.xxx.xxx.xxx fax 565730a028feef8 0x8 (alaw) No Rx: ACK fax
x.xxx.xx.xx DDI 1896289320_1272 0x100 (g729) No Rx: ACK SIPProvider
And it look like asterisk is trying to trancoding between codec:
channel.c:5064 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729)
My configurations are:
Can Asterisk send a re-invite to another codec the sip provider when receiving it from the ATA ? Alternatively, is it possible to have several incoming dialpeers in Asterisk from the same provider where I can set one for voice (allow=g729), and others for fax ddi’s (allow=alaw) ?