[Asterisk-Dev] Codec not negotiating
Jerris, Michael MI
mjerris at ofllc.com
Mon Apr 4 12:04:17 MST 2005
That bug does create an override codec (it is named poorly in the bug description). Multiple peers would certainly work.
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Michael Giagnocavo
Sent: Monday, April 04, 2005 2:56 PM
To: 'Asterisk Developers Mailing List'
Subject: RE: [Asterisk-Dev] Codec not negotiating
I'm not even sure if there will ever be a way to restrict the codecs sent.
Some people seem to think just sending a preferred codec is a good solution when in reality to force a preferred codec, you must only send that codec.
The easy solution here is to create two peer entries, one for ULAW, and one for G729 and then dial one or another.
-Michael
________________________________________
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jerris, Michael MI
Sent: Monday, April 04, 2005 12:23 PM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Codec not negotiating
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)
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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