Billy Huddleston
2004-Mar-12 12:02 UTC
[Asterisk-Users] Codec negotation with re-invites..
I'm about over this.. okay,, here is what I got..
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = inbound ; Default for incoming calls
tos=lowdelay
tos=184
disallow=all ; Disallow all codecs
allow=ulaw
[gateway]
type=friend
host=1.1.6.9
canreinvite=yes
qualify=yes
dtmfmode=rfc2833
context=default
disallow=all
allow=ulaw
allow=g729
[sipphoneg729]
type=friend
secret=password
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=longdistance-g729
dtmfmode=rfc2833
mailbox=2199
disallow=all
allow=g729
[sipphoneulaw]
type=friend
secret=password
nat=yes
host=dynamic
canreinvite=yes
qualify=200
context=longdistance
dtmfmode=rfc2833
mailbox=2199
disallow=all
allow=ulaw
okay, when I place a call from sipphoneulaw to the outside world via
gateway, everything works fine..
If I place a call from sipphoneg729, it doesn't work.. One leg to the
gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way
audio.. The sip phone can hear anything from the gateway, but, the gateway
can't hear the phone.
I've even went as far as to setup a seperate context for the g729 phone and
do this..
,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a
ulaw call.. Guys, this is a real problem... We're going be doing mixed
configs.. and if a gateway says it can do both, and phone says it can only
do one... then we should be using the compatable codec... PLEASE help..
This is going to cause problems in our rollout.
Thanks, Billy
+--------------------------------------------------+
| Billy Huddleston Senior Systems Administrator |
| Net-Express http://www.nxs.net |
| 114 Sherway Rd. Voice: 865-691-2011 |
| Knoxville, TN 37922 Fax: 865-691-9894 |
| billy@nxs.net |
+--------------------------------------------------+
I think it's because in de [general] section you only allow=ulaw and you shold allow=g729 to. I'm a newbie, hope I can help Best Regards Erick ----- Original Message ----- From: "Billy Huddleston" <billy@nxs.net> To: <asterisk-users@lists.digium.com> Sent: Friday, March 12, 2004 1:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites..> I'm about over this.. okay,, here is what I got.. > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = inbound ; Default for incoming calls > tos=lowdelay > tos=184 > disallow=all ; Disallow all codecs > allow=ulaw > > [gateway] > type=friend > host=1.1.6.9 > canreinvite=yes > qualify=yes > dtmfmode=rfc2833 > context=default > disallow=all > allow=ulaw > allow=g729 > > [sipphoneg729] > type=friend > secret=password > nat=yes > host=dynamic > canreinvite=yes > qualify=200 > context=longdistance-g729 > dtmfmode=rfc2833 > mailbox=2199 > disallow=all > allow=g729 > > [sipphoneulaw] > type=friend > secret=password > nat=yes > host=dynamic > canreinvite=yes > qualify=200 > context=longdistance > dtmfmode=rfc2833 > mailbox=2199 > disallow=all > allow=ulaw > > > okay, when I place a call from sipphoneulaw to the outside world via > gateway, everything works fine.. > If I place a call from sipphoneg729, it doesn't work.. One leg to the > gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way > audio.. The sip phone can hear anything from the gateway, but, the gateway > can't hear the phone. > > I've even went as far as to setup a seperate context for the g729 phoneand> do this.. > ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a > ulaw call.. Guys, this is a real problem... We're going be doing mixed > configs.. and if a gateway says it can do both, and phone says it can only > do one... then we should be using the compatable codec... PLEASE help.. > This is going to cause problems in our rollout. > > Thanks, Billy > > > +--------------------------------------------------+ > | Billy Huddleston Senior Systems Administrator | > | Net-Express http://www.nxs.net | > | 114 Sherway Rd. Voice: 865-691-2011 | > | Knoxville, TN 37922 Fax: 865-691-9894 | > | billy@nxs.net | > +--------------------------------------------------+ > > _______________________________________________ > 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 >
Sounds to me that your asterisk first negotiates g729 with your phone, then negotiates ulaw with the gateway (since it *is* the preferred codec in your config), and on a re-invite the logic breaks up either in the phone or in the gateway (or perhaps in the asterisk itself, I am not absolutely clear on the details of re-invites). Try changing the order of codec preference for the gateway and see if that fixes your g729 phone and breaks the ulaw phone at the same time. Alex. ----- Original Message ----- From: "Billy Huddleston" <billy@nxs.net> To: <asterisk-users@lists.digium.com> Sent: Friday, March 12, 2004 2:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites..> I'm about over this.. okay,, here is what I got.. > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = inbound ; Default for incoming calls > tos=lowdelay > tos=184 > disallow=all ; Disallow all codecs > allow=ulaw > > [gateway] > type=friend > host=1.1.6.9 > canreinvite=yes > qualify=yes > dtmfmode=rfc2833 > context=default > disallow=all > allow=ulaw > allow=g729 > > [sipphoneg729] > type=friend > secret=password > nat=yes > host=dynamic > canreinvite=yes > qualify=200 > context=longdistance-g729 > dtmfmode=rfc2833 > mailbox=2199 > disallow=all > allow=g729 > > [sipphoneulaw] > type=friend > secret=password > nat=yes > host=dynamic > canreinvite=yes > qualify=200 > context=longdistance > dtmfmode=rfc2833 > mailbox=2199 > disallow=all > allow=ulaw > > > okay, when I place a call from sipphoneulaw to the outside world via > gateway, everything works fine.. > If I place a call from sipphoneg729, it doesn't work.. One leg to the > gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way > audio.. The sip phone can hear anything from the gateway, but, the gateway > can't hear the phone. > > I've even went as far as to setup a seperate context for the g729 phoneand> do this.. > ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a > ulaw call.. Guys, this is a real problem... We're going be doing mixed > configs.. and if a gateway says it can do both, and phone says it can only > do one... then we should be using the compatable codec... PLEASE help.. > This is going to cause problems in our rollout. > > Thanks, Billy > > > +--------------------------------------------------+ > | Billy Huddleston Senior Systems Administrator | > | Net-Express http://www.nxs.net | > | 114 Sherway Rd. Voice: 865-691-2011 | > | Knoxville, TN 37922 Fax: 865-691-9894 | > | billy@nxs.net | > +--------------------------------------------------+ > >