[Asterisk-Dev] Codec not negotiating
Jerris, Michael MI
mjerris at ofllc.com
Mon Apr 4 11:23:22 MST 2005
http://bugs.digium.com/bug_view_page.php?bug_id=0003346
<http://bugs.digium.com/bug_view_page.php?bug_id=0003346> should
address this issue, but there is not yet a patch with the implementation
that was decided upon yet.
_____
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Clay Reiche
Sent: Monday, April 04, 2005 1:57 PM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Codec not negotiating
ok... I've trying to fix this for days... I got very little response
from the Users list. I have a sip device that registers with my *. The
sip device is ONLY set up to use ulaw. My asterisk server sends ALL PSTN
calls to a Sonus gateway/softswitch. When I place a PSTN call, the sip
device sends the INVITE with SDP and the ONLY codec option is ulaw.
Asterisk then turns around and sends an INVITE with SDP to the Sonus
gateway with ulaw as the first option and g729 as a second option. The
Sonus sees the TWO options and ALWAYS chooses g729. The codec
negotiation fails and the call never completes.
I understand that the TWO options are sent because I have no peer set up
for the Sonus in my sip.conf and it defaults to the [general] codec
settings which are ulaw and g729. However, MOST of my calls to the Sonus
ARE using g729, only a few need to use ulaw. (for faxing) So I can't
restrict the Sonus peer to only ulaw...
Here is my question:(finally...sorry:))
Can I force asterisk to send ONLY my prefered codec?(the first one in
the INVITE) or is this only fixed by pleading with the people who run
the Sonus sofswitch to stop ignoring my preferred codec? or is there
some other solution? Any suggestions would be very appreciated!
CONFIG FILES:
Sip.Conf:
[general]
context=default ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique
according to RFC 3261
; Set this to your host name or domain
name
port=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=no ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
;pedantic=yes ; Enable slow, pedantic checking for
Pingtel
; and multiline formatted headers for
strict
; SIP compatibility (defaults to "no")
;tos=184 ; Set IP QoS to either a keyword or
numeric val
;tos=lowdelay ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600 ; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI
NOTIFY
;videosupport=yes ; Turn on support for SIP video
disallow=all ; First disallow all codecs
allow=g729
allow=ulaw ; Allow codecs in order of preference
;allow=alaw
;allow=g723.1
;allow=ilbc ; Note: codec order is respected only in
[general]
;musicclass=default ; Sets the default music on hold class
for all SIP calls
; This may also be set for individual
users/peers
;language=en ; Default language setting for all
users/peers
; This may also be set for individual
users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP
activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no
RTP activity
; when we're on hold (must be >
rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;progressinband=no ; If we should generate in-band ringing
always
useragent=Abox SS1.0 ; Allows you to change the user agent
string
;nat=no ; NAT settings
; yes = Always ignore info and assume
NAT
; no = Use NAT mode only according to
RFC3581
; never = Never attempt NAT mode or
RFC3581 support
; route = Assume NAT, don't send rport
(work around more UNIDEN bugs)
;usereqphone=no
[8138644418]
type=friend
username=8138644418
secret=C34589Y
host=dynamic
nat=yes
context=from-sip
callerid=8138644418
canreinvite=yes
mailbox=8138644418
accountcode=accxx_group
disallow=all
allow=g729
allow=ulaw
######################################################################
extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
include => default
include => parkedcalls
include => iaxtel700
include => iaxprovider
include => from-sip
[default]
include => from-sip
[from-sip]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@216.229.127.60
<mailto:SIP/$%7bEXTEN%7d at 216.229.127.60> )
exten =>
18138644418,4,Dial(IAX2/poseidon:olympus at 72.21.12.4/8138644418 at from-sip)
exten => 18138644418,3,Wait(2)
exten => 18138644418,2,Dial(SIP/8138644418,20)
exten => 18138644418,1,SetCDRUserField(accxx_group)
###################################################################
Thank you!
Clay Reiche
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