Displaying 5 results from an estimated 5 matches for "dumphistory".
2007 May 03
2
SIP peer / Maximum retries exceeded on transmission
...allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=proxy.myhostname
disallow=all
allow=alaw
sipdebug = yes
recordhistory=yes
dumphistory=yes
register => <authstuff>@sip.externalpeer.com
externhost=proxy.myhostname
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0
nat=never
canreinvite=no
[authent...
2010 Nov 03
1
inbound call issue...
...s: 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
promiscredir = no
realm = asterisk
reco...
2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
...060
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 PBX
usereqphone = no
videosupport = no
disallow = all
allow = alaw
type = friend
host=dynamic
context...
2010 Oct 12
0
rtpip patch
...DSCP EF (Expedited Forwarding), but the default is
0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO 0 /*!< Video packets
should be marked as DSCP AF41, but the default is 0 to be compatible
with previous versions. */
@@ -1106,10 +1103,6 @@
static int dumphistory; /*!< Dump history to verbose before
destroying SIP dialog */
static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for
auto-extensions */
static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for
the SIP channel */
-
-static char global_rtpip[AST_MAX_EXTENSION];
-
-...
2010 Nov 05
2
How to append custom option to Contact: header on outgoing SIP INVITE msgs?
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here>
where SITE-ID=us.here is set in a config file that * parses on
startup. Or in a Dial() command option? Or I don't care exactly
how. :-)
It is possible to