Displaying 7 results from an estimated 7 matches for "3dulaw".
2004 Jan 14
3
grandstream asterisk configuration
...D 5060 ; Port to bind to
bindaddr =3D 0.0.0.0 ; Address to bind to
;externip =3D 200.201.202.203 ; Address that we're going to put in =
SIP
messages if we're behind a NAT
tos=3Dlowdelay
disallow=3Dall ; Disallow all codecs
allow=3Dulaw ; Allow codecs in order of preference
dtmfmode=3Dinfo
[grandstream1]
type=3Dfriend
host=3Ddynamic
secret=3Dmysecret
context=3Doutgoing
nat=3Dyes
reinvite=3Dno
canreinvite=3Dno
qualify=3D2000
has anyone done this before?
chandra
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2009 Jul 20
0
No subject
...<br />faststart=3Dyes<br />h245tunnel=
ing=3Dyes<br />h323id=3DObjSysAsterisk<br />e164=3D100<br />callerid=3Daste=
risk<br />gatekeeper =3D DISABLE<br />context=3Ddefault<br />disallow=3Dall=
<br />allow=3Dulaw<br />dtmfmode=3Drfc2833<br />pr=
ogress_setup =3D 8<br />progress_alert =3D 8<br /><br />[denver]<br />type=
=3Dfriend<br />port=3D1720<br />ip=3Dip address of avaya<br />context=3Dfro=
m-internal<br />disallow=3Dall<br />allow=3Dula...
2007 Jul 12
0
No subject
[general]
port =3D 5060 ; Port to bind to
bindaddr =3D 0.0.0.0 ; Address to bind to A
realm=3D192.168.0.2
context =3D default ;Default for incoming calls [5549] disallow=3Dall =
allow=3Dulaw allow=3Dalaw allow=3Dgsm type=3Dfriend ;(inbound and =
outbound calls accepted) secret=3Dlocalphone ; obvious password for =
testing host=3Ddynamic callerid=3DJason White <5549> dtmfmode=3Dauto
mailbox=3D5549 ;(Asterisk VM-system's mailbox #)
The output from sip set debug is attached, as...
2007 Jul 12
0
No subject
...SIP/${NUMBER}@a.b.c.d
>
>
>
> How can you use authentication, along with SIP, along with specifying
> extension?
>
>
>
> My sip.conf has a friend defined:
>
>
>
> [priv]
>
> host=3Ddynamic
>
> secret=3Dpriv
>
> disallow=3Dall
>
> allow=3Dulaw
>
> canreinvite=3Dno
>
> nat=3Dno
>
> context=3Dfrom-internal\
>
> type=3Dfriend
>
>
>
> I need to specify the sip channel to use the priv peer, priv secret, and
> pass the extension. I've tried defining my mapping as:
>
>
>
> Priv =3D> dun...
2003 Nov 11
2
sip: 401 unauthorized with xlite
Hi there,
I have tried very hard to setup the x-lite with asterisk, but until now i didn't get sucess. When i start the asterisk in debug mode, i see the message: sip/2.0 401 unauthorized. I know that this problem with authentication. I put in my sip.conf as below.
[2203]
type=friend
username=2203
auth=md5
secret=1234
reinvite=no
canreinvite=no
dissallow=all
allow=gsm
2009 Jul 20
0
No subject
...of sip.conf works great and I can make ca=
lls<br>> between the phones using g722. However Asterisk is negotiating =
g722<br>
> for calls going out my voip provider and transcoding these to ulaw. In=
<br>> sip.conf for the provider I have deny=3Dall and allow=3Dulaw. This=
can<br>> cause potential audio degrading and wastes cpu cycles. If Aste=
risk<br>
> knows the trunk only supports ulaw why would it offer g722 to the<br>&=
gt; phone.<br>><br>> Ryan<br><br>Because the codec is already...
2004 Apr 28
9
chan_sip.c max number of retries?
Still getting the same error.
Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request)
please advise anyone!!!!!someone!!!
jai