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
> 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?That's because the codec is a bitmask. bkw
Excuse me, but can you explain a little more, what is the problem? so i can search for the solutions, regards -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brian West Sent: Tuesday, October 12, 2004 5:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] G729 to G711 bridge> 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?That's because the codec is a bitmask. bkw _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users