[Asterisk-Users] G729 to G711 bridge

Miranda Gomez Miguel Angel mmiranda at americatel.com.sv
Tue Oct 12 16:19:19 MST 2004


Hi,
Im having some bridging problems between the grandstream phones using g729
and sjphones using g711, this is my setup:

 _ _ _ _ _ _ _         _ _ _       _ _ _ _ _ ____
| GS Phone    |     |  *     |     | SJ Phone       |
| G729 (2008) |- - -| PBX |- - -| G711 (2000)    |
|_ _ _  _ _ _ _ |     |_ _ _ |     |_ _ _ _ _ _ _ _ |
 

when i call from the g711 (ext 2000) to g729 (ext 2008) the call quality is
ok, but i cant hear nothing when i call in the other direction, these are
the relevant config and logs in debug mode:

sip.conf

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
context = bogon-calls ; Send SIP callers that we don't know about here
disallow=all
allow=alaw

[2000]

type=friend           ; This device takes and makes calls
username=2000         ; Username on device
secret=mmiranda ; Password for device
host=dynamic          ; This host is not on the same IP addr every time
context=from-sip      ; Inbound calls from this host go here
mailbox=2000           ; Activate the message waiting light if this
                      ; voicemailbox has messages in it
nat=yes

[2008]
type=friend           ; This device takes and makes calls
username=2008         ; Username on device
secret=2008 ; Password for device
host=dynamic          ; This host is not on the same IP addr every time
context=from-sip      ; Inbound calls from this host go here
mailbox=2008          ; Activate the message waiting light if this
                      ; voicemailbox has messages in it
nat=yes
disallow=all                   
allow=g729


logs in debug:

Called 2001
    -- SIP/2001-0a50 is ringing
    -- SIP/2001-0a50 answered SIP/2008-24fc
    -- Attempting native bridge of SIP/2008-24fc and SIP/2001-0a50
Oct 12 17:15:59 WARNING[76823]: rtp.c:1392 ast_rtp_bridge: codec0 = 277 is
not codec1 = 8, cannot native bridge.
  == Spawn extension (from-sip, 2001, 1) exited non-zero on 'SIP/2008-24fc'


the g729 codecs is  showed as number 256 and not 277 in show codecs, what
can be wrong?

thanks
Miguel




More information about the asterisk-users mailing list