Help with ring group


#1

I just installed asterisk 1.2 and setup 2 softphones and broadvoice as my sip gateway. All is working except that I want incoming calls to ring both phones and if no one answers go to a voicemail menu asking for the users you’d like to leave a message for. I added both phones to the incoming call context and it tries to ring both but I get this from the console:

    -- Executing Dial("SIP/xxxxxxxxxx-2120", "SIP/2000&SIP/2001&SIP/user|15|tTr") in new stack
    -- Called 2000
    -- Called 2001
Jan  2 10:18:01 WARNING[2924]: chan_sip.c:1966 create_addr: No such host: user
Jan  2 10:18:01 NOTICE[2924]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
    -- SIP/2000-bc54 is ringing
    -- SIP/2001-2c8c is ringing
  == Spawn extension (default, xxxxxxxxxx, 1) exited non-zero on 'SIP/xxxxxxxxxx-2120'

From the outside line that I’m calling from I hear it ring twice and it goes to user 2000’s voicemail. I’m very new so any advice or suggestions are welcome.

[code]
extensions.conf:
[general]
static=yes
writeprotect=no

[default]
; Echo test - dial 600
exten => 600,1,Playback(demo-echotest)
exten => 600,2,Echo
exten => 600,3,Playback(demo-echodone)
exten => 600,4,Goto(s,6)

; Dial 2999 for voicemail
exten => 2999,1,Ringing
exten => 2999,2,VoicemailMain,s2000

; Dial out - dial 9 for external line
exten => _9.,1,Dial(SIP/${EXTEN:1}@sip.broadvoice.com)

; Incoming calls
;exten => xxxxxxxxxx,1,Dial(SIP/2000,15,tTr)
; Same as above but rings ext 2000 and 2001
exten => xxxxxxxxxx,1,Dial(SIP/2000&SIP/2001&SIP/user,15,tTr)
exten => xxxxxxxxxx,2,Voicemail,u2000
exten => xxxxxxxxxx,102,Voicemail,b2000
exten => xxxxxxxxxx,103,Hangup

sip.conf

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all ; Dis-allow all codecs
allow=ulaw
allow=alaw
maxexpirey=3600
defaultexpirey=120
contex=default

pedantic=no
register => xxxxxxxxxx@sip.broadvoice.com:TNyrcNiiiA:xxxxxxxxxx@sip.broadvoice.com/xxxxxxxxxx
;register => xxxxxxxxxx@sip.broadvoice.com:TNyrcNiiiA:xxxxxxxxxx@sip.broadvoice.com/2000

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=xxxxxxxxxx
secret=password
username=xxxxxxxxxx
insecure=very
context=default
authname=xxxxxxxxxx
dtmfmode=inband
dtmf=inband
canreinvite=no

[2000]
type=friend
secret=password
mailbox=2000
canreinvite=yes
context=default
insecure=very
host=dynamic
nat=no
username=user1

[2001]
type=friend
secret=password
mailbox=2001
canreinvite=yes
context=default
insecure=very
host=dynamic
nat=no
username=user2
[/code][/code]


#2

Well as soon as I posted I saw part of the problem. I removed the “user” from the extensions.conf:

- exten => xxxxxxxxxx,1,Dial(SIP/2000&SIP/2001&SIP/user,15,tTr) +exten => xxxxxxxxxx,1,Dial(SIP/2000&SIP/2001&SIP/,15,tTr)

So it now rings both phones until it goes to voicemail. It is still going to user 2000’s voicemail, though and I’m still getting errors on the console:

-- Executing Dial("SIP/xxxxxxxxxx-a27b", "SIP/2000&SIP/2001&SIP/|15|tTr") in new stack -- Called 2000 -- Called 2001 Jan 2 10:49:36 WARNING[2951]: chan_sip.c:1966 create_addr: No such host: Jan 2 10:49:36 NOTICE[2951]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) -- SIP/2000-200d is ringing -- SIP/2001-c79b is ringing -- Nobody picked up in 15000 ms -- Executing VoiceMail("SIP/xxxxxxxxxx-a27b", "u2000") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (default, xxxxxxxxxx, 2) exited non-zero on 'SIP/xxxxxxxxxx-a27b'

Thanks!