Displaying 11 results from an estimated 11 matches for "compactheaders".
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2010 Oct 12
0
rtpip patch
...le I produced for chan_sip.c in asterisk 1.6.2.11
--- chan_sip.c 2010-10-12 13:47:49.000000000 +0200
+++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200
@@ -987,9 +987,6 @@
#define DEFAULT_CALLCOUNTER FALSE
#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact
(one-character) SIP headers. Default off */
-
-#define DEFAULT_RTPIP "auto"
-
#define DEFAULT_TOS_SIP 0 /*!< Call signalling
packets should be marked as DSCP CS3, but the default is 0 to be
compatible with previous versions. */
#defi...
2010 Nov 03
1
inbound call issue...
...241-538634340-1288768121281-
Max-Forwards: 70
Content-Length: 0
Here's 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
pr...
2007 Sep 18
1
stanaphone issues. can someone verify my config?
...s
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=no
allow=all
allow=g729
rtptimeout=4 (tried this on the default of 30 and it just makes it take
longer to give the error, and I like it low incase the internet dies I don't
end up talking to nothing for a long time without realizing it.)
compactheaders = yes
externip = 60.xxxxxx (our static IP is here)
localnet=192.168.0.0/255.255.0.0;
nat=yes
canreinvite=no
; richards stanaphone incoming to ext 8800
register => 089xyz:xxxxxxxx at sip.stanaphone.com/8800
; richards italk to ext 8800
register => 64997xxxxx:xxxxx at akl.italk.co.nz/8800...
2007 Oct 23
1
Short format of SIP INVITE - how to change
My Asterisk box send INVITEs in the short form, i.e.,
"f:" instead of "from", "v:" instead of "via" and so
on.
Is there a way to force asterisk to use full format?
thanks
Vitaly
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2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
...esult.
What I did not configured?
My sip.conf
[general]
context = default
allowguest = no
bindport = 5060
bindaddr = 0.0.0.0
allowexternaldomains = 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...
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
...allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
maxjitterbuffer=1500
;allow=ilbc
;musicclass=default
;language=en
;relaxdtmf=yes
rtptimeout=60
;rtpholdtimeout=300
;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no
;usereqphone = no
dtmfmode = rfc2833
;compactheaders = yes
;sipdebug = yes
;subscribecontext = default
;notifyringing = yes
And these are the extensions:
[xxxx]
type=friend
host=dynamic
dtmfmode=rfc2833
username=xxxx
secret=xxxx
[xxxx2]
type=friend
host=dynamic
dtmfmode=rfc2833
username=xxxx
secret=xxxx
As you can see I put the jitterbuf...
2011 Jul 19
3
CentOS 6
I finally switched workstations, and am running into a lot of truly
annoying details with CentOS 6, and the software with it. For one, I'm
doing this by webmail, and this version of Firefox *insists* on putting
what I'm typing in gray, rather than black. For another, I despise the new
version of thunderbird, since it now shows the full subject, if I have
what they used to call the preview
2009 Aug 04
0
SIP server behind NAT
...agent string
> promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
> ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
> dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
> ;compactheaders = yes ; send compact sip headers.
> ;sipdebug = yes ; Turn on SIP debugging by default, from
> ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
> ;notifyringing = yes ; Notify subscriptions on RINGING state
> ;alwaysaut...
2006 Dec 18
0
pap2/wrt54gs/asterisk
...ult: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64
kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband
otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this
configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE
requests...
2011 Feb 10
2
Unable to make outgoing calls with Internode
...one number>
;fromdomain = internode.on.net
;secret = <secret>
;qualify = yes
canreinvite = no
;auth = <phone number>:<secret>@BroadWorks
;nat = never
;pedantic = yes
;insecure = invite,port
;ignoresdpversion = yes
;compactheaders = yes
As you can see I've tried lots of settings. It registers and peers with
the provider, but no outgoing. The provider can call me though.
In extensions.conf:
[internode-outgoing]
exten => _X.,1,Dial(SIP/${EXTEN}@sip-out)
exten => _X.,n,Answer(2)
exten...
2005 Jan 05
5
Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and
are using the CVS version.
Goto the wiki and read the instructions for RealTime.
-Matthew
----- Original Message -----
From: "Muhammad Rizwan Khan" <rizwan@advcomm.net>
To: <Asterisk-Dev@lists.digium.com>
Sent: Wednesday, January 05, 2005 12:42 PM
Subject: [Asterisk-Dev] Asterisk with MySQL
>