Trying to get SIP realtime working here...
I'm connected to the database...
*CLI> realtime mysql status
Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin
for 6 seconds.
I can get information for the extension in question...
*CLI> realtime load sipusers name 2944093
Column Name Column Value
-------------------- --------------------
id 1
name 2944093
accountcode 2944093
callgroup 1
canreinvite no
context From_OneEighty
dtmfmode auto
nat rfc35
pickupgroup 1
qualify no
type friend
username 2944093
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
regseconds 0
cancallforward yes
subscribecontext sub_oneeighty
First of all, why doesn't Asterisk show _ALL_ the fields in the table?
There's way more than this.
Second, when my phone comes up, asterisk displays this on the console:
*CLI> Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854
handle_request_register: Registration from
'<sip:2944093@ipt.oneeighty.com>' failed for
'216.xxx.142.205' - Username/auth name mismatch
I'm trying to do this in insecure mode, so Asterisk shouldn't even be
asking the phone for a password. What's the deal? When I run an ngrep on the
database, I can see that Asterisk isn't even TRYING to query the extension.
Huh??? My sip.conf just has a [global] section, no users are provisioned in it.
Doug.
Paul A Brown
2006-Mar-17 17:19 UTC
[Asterisk-Users] Newbie Questions - Any help appreciated
Sorry for the long email but I am having all sorts of
probs................................
I basically have a number od sip phones in the house....
I have 3 incoming numbers (sipgate) and one outbound service (sipdiscount)
I want all extensions to be able to call out using the outbound lines (one
at a time obviousley) and I want various extensions to ring depending on
which inbound number is called.
Problems............
1) When I boot Asterisk it no longer connects to sipgate to register the
inbound lines, it did earlier on today but isn't anymore, does it look like
I did something with my config?
2) When I select the extension and try and dial out, I immediately get the
engaged tone on the phone. It hasn't had time to dial out so I know its at
the asterisk end.
3) When I dial from ext to ext the voicemail doesn't work.....
Ho hum...............
Here are my sip and extensions conf. Any help appreciated
______________________________________________________________________________________
extensions.conf
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The "General" category is for certain variables.
;
[general]
static=yes
writeprotect=no
[globals]
PHONES1=SIP/220
PHONES1VM=220
PHONES2=SIP/221
PHONES2VM=221
PHONES3=SIP/222
PHONES3VM=222
PHONES4=SIP/223
PHONES4VM=223
PHONES5=SIP/224
PHONES5VM=224
PHONES5=SIP/225
PHONES5VM=225
[sipdiscount-outbound]
exten => <220>,1,Dial(${EXTEN}@sipdiscount)
exten => <221>,1,Dial(${EXTEN}@sipdiscount)
exten => <222>,1,Dial(${EXTEN}@sipdiscount)
exten => <223>,1,Dial(${EXTEN}@sipdiscount)
exten => <224>,1,Dial(${EXTEN}@sipdiscount)
exten => <225>,1,Dial(${EXTEN}@sipdiscount)
[sipgate-inbound]
exten => 3858313,1,Dial(SIP/220&SIP/221&SIP/223)
exten => 3858294,1,Dial(SIP/220)
exten => 3858817,1,Dial(SIP/221&SIP/220))
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
; include => iaxtel700
include => trunktollfree
include => iaxprovider
include => sipdiscount-outbound
;This will create a macro we will use in the dialling plan
[macro-vmessage]
exten => s,1,VoiceMail2(u${ARG1})
exten => s,2,Playback(groovy)
exten => s,3,Playback(goodbye)
exten => s,4,Hangup
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
VoicemailMain
; ----------------------------------------------
; DEFINE EXTENSIONS
; ----------------------------------------------
[home]
exten => 220,1,Dial(${PHONES1},20,Ttm)
exten => 220,2,Macro(vmessage,${PHONES1VM})
exten => 220,3,Hangup
; Line 2
exten => 221,1,Dial(${PHONES2},20,Ttm)
exten => 221,2,Macro(vmessage,${PHONES2VM})
exten => 221,3,Hangup
; Line 3
exten => 222,1,Dial(${PHONES3},20,Ttm)
exten => 222,2,Macro(vmessage,${PHONES3VM})
exten => 222,3,Hangup
; Line 4
exten => 223,1,Dial(${PHONES4},20,Ttm)
exten => 223,2,Macro(vmessage,${PHONES4VM})
exten => 223,3,Hangup
; Line 5
exten => 224,1,Dial(${PHONES5},20,Ttm)
exten => 224,2,Macro(vmessage,${PHONES5VM})
exten => 224,3,Hangup
; Line 6
exten => 225,1,Dial(${PHONES6},20,Ttm)include => sipdiscount-outbound
exten => 225,2,Macro(vmessage,${PHONES6VM})
exten => 225,3,Hangup
; ----------------------------------------------
; END DEFINE EXTENSIONS
; ----------------------------------------------
___________________________________________________________________________________________
sip.conf
;
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
; reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
defaultexpiry=3600 ; Default length of incoming/outoing registration
videosupport=yes ; Turn on support for SIP video
recordhistory=yes ; Record SIP history by default
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
register =>3858294:password@sipgate.co.uk/3858294
register =>3858817:password@sipgate.co.uk/3858817
register =>3858313:password@sipgate.co.uk/3858313
externip = 84.1.1.115 ; Address that we're going to put in outbound SIP
messages
; if we're behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
externhost=whatever.co.uk ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
localnet=192.192.192.0/255.255.255.0; All RFC 1918 addresses are local
networks
allow=ulaw
allow=alaw
[220]
type=friend
context=home
callerid=Paul<220>
nat=yes
host=dynamic
defaultip=192.192.192.220
username=220
secret=password
mailbox=220
dtmfmode=rfc2833
[221]
type=friend
context=home
callerid=Ellie<221>
nat=yes
host=dynamic
defaultip=192.192.192.221
username=221
secret=password
mailbox=221
dtmfmode=rfc2833
[222]
type=friend
context=home
callerid=Ellie<222>
nat=yes
host=dynamic
defaultip=192.192.192.222
username=222
secret=password
mailbox=222
dtmfmode=rfc2833
[223]
type=friend
context=home
callerid=Garage<223>
nat=yes
host=dynamic
defaultip=192.192.192.223
username=223
secret=PASSWORD
mailbox=223
dtmfmode=rfc2833
[sipdiscount]
type=peer
host=sip1.sipdiscount.com
fromdomain=sip1.sipdiscount.com
progressinband=yes
dtmfmode=inband
disallow=all
allow=alaw
allow=ulaw ; only alaw works with sip1...
;allow=g729 ; but no way to have DMTF with G.729 !
nat=yes
canreinvite=no
qualify=yes
insecure=very
context=incoming
authuser=user1
username=user1
fromuser=user1
secret=password
[sipgate]
type=friend
host=sipgate.co.uk
insecure=very
context=sipgate-inbound
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;username=goran ; Username to use when calling this device before
registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this
device
Douglas Garstang wrote:> Trying to get SIP realtime working here... > > I'm connected to the database... > > *CLI> realtime mysql status > Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. > > I can get information for the extension in question... > > *CLI> realtime load sipusers name 2944093 > Column Name Column Value > -------------------- -------------------- > id 1 > name 2944093 > accountcode 2944093 > callgroup 1 > canreinvite no > context> dtmfmode auto > nat rfc35 > pickupgroup 1 > qualify no > type friend > username 2944093 > disallow all > allow g729 > allow ilbc > allow gsm > allow ulaw > allow alaw > regseconds 0 > cancallforward yes > subscribecontext sub_oneeighty > > First of all, why doesn't Asterisk show _ALL_ the fields in the table? There's way more than this. > > Second, when my phone comes up, asterisk displays this on the console: > > *CLI> Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854 handle_request_register: Registration from '<sip:2944093@ipt.oneeighty.com>' failed for '216.xxx.142.205' - Username/auth name mismatch > > I'm trying to do this in insecure mode, so Asterisk shouldn't even be asking the phone for a password. What's the deal? When I run an ngrep on the database, I can see that Asterisk isn't even TRYING to query the extension. Huh??? My sip.conf just has a [global] section, no users are provisioned in it. > > Doug. > >Hi, do you have in sip.conf [From_OneEighty] switch => Realtime/sipusers@extensions
Yusuf, No I don't have the switch statement in extensions.conf. I'm not trying to do realtime extensions. I'm trying to do realtime SIP. They're different. Doug. -----Original Message----- From: yusuf [mailto:yusuf@ecntelecoms.com] Sent: Sat 3/18/2006 6:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] SIP Realtime Users Douglas Garstang wrote: > Trying to get SIP realtime working here... > > I'm connected to the database... > > *CLI> realtime mysql status > Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. > > I can get information for the extension in question... > > *CLI> realtime load sipusers name 2944093 > Column Name Column Value > -------------------- -------------------- > id 1 > name 2944093 > accountcode 2944093 > callgroup 1 > canreinvite no > context > dtmfmode auto > nat rfc35 > pickupgroup 1 > qualify no > type friend > username 2944093 > disallow all > allow g729 > allow ilbc > allow gsm > allow ulaw > allow alaw > regseconds 0 > cancallforward yes > subscribecontext sub_oneeighty > > First of all, why doesn't Asterisk show _ALL_ the fields in the table? There's way more than this. > > Second, when my phone comes up, asterisk displays this on the console: > > *CLI> Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854 handle_request_register: Registration from '<sip:2944093@ipt.oneeighty.com>' failed for '216.xxx.142.205' - Username/auth name mismatch > > I'm trying to do this in insecure mode, so Asterisk shouldn't even be asking the phone for a password. What's the deal? When I run an ngrep on the database, I can see that Asterisk isn't even TRYING to query the extension. Huh??? My sip.conf just has a [global] section, no users are provisioned in it. > > Doug. > > Hi, do you have in sip.conf [From_OneEighty] switch => Realtime/sipusers@extensions _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users