Displaying 6 results from an estimated 6 matches for "g726nonstandard".
2006 Dec 06
1
FW: G.726 on Asterisk 1.4.0
...wever, there are existing devices that
improperly request one order and then use another; Sipura and Grandstream
ATAs are known to do this, and there may be others. To be able to continue
to use these devices with this version of Asterisk and the G726-32 codec, a
configuration parameter called 'g726nonstandard' has been added to sip.conf,
so that Asterisk can use the packing order expected by the device (even
though it requested a different order). In addition, the internal format
number for G726-32 has been changed, and the old number is now assigned to
AAL2-G726-32. The result of this is that this...
2010 Aug 02
6
Codec negotiation : expecting G726, getting G711a (alaw)
Hello list,
Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle.
Grandstream allows for 8 different codec specifications. I have defined
them as 4 x G726 & 4 x alaw.
Snom allow for 7 different codec specifications. I have defined them as
3 x G726 & 4 x G729.
The SIP peers are both defined as :
disallow=all
allow=g726
allow=alaw
allow=g729
allow=gsm
This is the
2010 Nov 03
1
inbound call issue...
...the configs:
subscribecontext = device-hints
allowexternaldomains = yes
allowguest = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = yes
checkmwi = 10
compactheaders = no
defaultexpiry = 120
dumphistory = no
externip = 216.26.109.22
g726nonstandard = no
jbenable = yes
jbforce = no
jblog = no
localnet = internal subnet
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
mohinterpret = default
nat = yes
notifyringing = yes
pedantic = no
progressinband = never
promiscredir = no
realm = asterisk
recordhistory = no
registerattempts = 0
registerti...
2014 Jul 26
1
Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
...trunkname: NULL
cid_number: NULL
callingpres: NULL
mohinterpret: NULL
mohsuggest: NULL
parkinglot: NULL
hasvoicemail: NULL
subscribemwi: NULL
vmexten: NULL
autoframing: NULL
rtpkeepalive: NULL
call-limit: NULL
g726nonstandard: NULL
ignoresdpversion: NULL
allowtransfer: NULL
dynamic: NULL
path: NULL
supportpath: NULL
sippasswd: my-md5-pwd
rpid: NULL
domain: testers.com
sippasswd2: NULL
I'd greatly appreciate help on this!
cheers,
Oll...
2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
...ldomains = no
allowoverlap = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = no
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain=sop-korniychuk
domain=172.30.8.13
domain=172.30.8.13:5060
dumphistory = no
externrefresh = 10
g726nonstandard = no
notifyringing = yes
srvlookup = yes
t1min = 100
t38pt_udptl = no
;tos_audio = none
;tos_sip = none
;tos_video = none
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no
disallow = all
allow = alaw
type = friend
host=dynamic
context = noop-context
dtmfmode=rfc2833
nat=no...
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...nable=yes
tlsbindaddr=0.0.0.0
tlscipher=ALL
tlsclientmethod=tlsv1
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
callevents=no
jbenable=no
videosupport=yes
allowguest=no
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes
[1000]
deny=0.0.0.0/0.0.0.0
secret=6ff108122cce3b0b45e0abf374c14ef4
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no...