Can't make SIP/13-9d9a and SIP/12-0aaf compatible


I has test two xten’s softphones connect fine.
But I use two sip phones to connect using codec g.723 all not in NAT.
I can’t answer connect. log as below.
Do need the asterisk must support codec g.723 ?


;Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;regexten=12 ; When they register, create extension 1234
callerid=“AT22-13” <13>
nat=no ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
allow=all ; GSM consumes far less bandwidth than ulaw

asterisk log

Feb 21 08:52:34 NOTICE[3469]: channel.c:1736 ast_set_read_format: Unable to find a path from g723 to ulaw
Feb 21 08:52:34 NOTICE[3469]: channel.c:1703 ast_set_write_format: Unable to find a path from ulaw to g723
Feb 21 08:52:34 WARNING[3469]: chan_sip.c:1834 sip_write: Asked to transmit frame type 1, while native formats is 4 (read/write = 4/4)
Feb 21 08:52:34 WARNING[3469]: channel.c:2127 ast_channel_make_compatible: No path to translate from SIP/13-9d9a(1) to SIP/12-0aaf(4)
Feb 21 08:52:34 WARNING[3469]: channel.c:2653 ast_channel_bridge: Can’t make SIP/13-9d9a and SIP/12-0aaf compatible
Feb 21 08:52:34 WARNING[3469]: res_features.c:382 ast_bridge_call: Bridge failed on channels SIP/13-9d9a and SIP/12-0aaf
== Spawn extension (from-sip, 12, 1) exited non-zero on 'SIP/13-9d9a’


I just had a similar problem. In my case a sip phone was originating a call on g729 (someone had set this as a preference in [general] of sip.conf). When the X-Lite answered the call couldn’t be made compatibile. Once I made sure that [general] had ulaw, alaw, gsm in that order - and NO g729, it worked. (The issue was of course that my * box had no g729 license so couldn’t transcode).

Sounds like something related is going on here.