Problem connecting SIP calls from asterisk server to another

hi
i have two asterisk servers
Asterisk 1.2.13 on my Debian machine (named obelisk)
Asterisk 1.2.14 on my wrt54gl router running openwrt as os (named asterisk)

now i want to connect calls made from users connected to the router to the debian machine.
the users “ma”,“am” and “da” call “conf” and they should be connected to a conferenceroom on the debian machine.
the sip.conf from the router

[general]
language=de
bindaddr=0.0.0.0
port=5060
disallow=all
allow=gsm
tos=lowdelay

[obelisk]
type=user
secret=passwort
auth=md5
host=192.168.137.111
context=vpnguests

[ma]
type=friend
context=vpnguests
secret=pw1
host=dynamic

[am]
type=friend
context=vpnguests
secret=pw2
host=dynamic

[da]
type=friend
context=vpnguests
secret=pw3
host=dynamic

the extensions.conf on the router:
i know one dial command should do it if it s done right, but i tried several at the same time to compare the output.

[general]
static=yes
writeprotect=no

[vpnguests]
exten => da,1,Dial(SIP/da)

exten => am,1,Dial(SIP/am)

exten => ma,1,Dial(SIP/ma)

exten => conf,1,Dial(SIP/asterisk:passwort@192.168.137.111 /router)
exten => conf,n,Dial(SIP/"asterisk:passwort@192.168.137.111")
exten => conf,n,Dial(SIP/asterisk:passwort@obelisk)
exten => conf,n,Dial(SIP/asterisk:passwort@192.168.137.111/${EXTEN})
exten => conf,n,Congestion()

the debians sip.conf :

[general] 
language=de 
bindaddr=0.0.0.0 
port=5060
disallow=all 
allow=gsm 
tos=lowdelay 

[asterisk] 
type=user 
secret=passwort 
auth=md5 
host=192.168.137.245 
context=router

the debians extensions.conf :

[general] 
static=yes 
writeprotect=no 

[router]
exten => conf,1,Answer()
exten => conf,n,Wait(1)
exten => conf,n,MeetMe(1234)
exten => conf,n,Hangup()
exten => s,1,Answer()
exten => s,n,Wait(1)
exten => s,n,MeetMe(1234)
exten => s,n,Hangup()
exten => i,1,Answer()
exten => i,n,Wait(1)
exten => i,n,MeetMe(1234)
exten => i,n,Hangup()

[default]

the background why i try this, is that we are 3 people that want to communicate via voip through a vpn. meetme does not work on my router because it lacks the needed timer device and i cant get kernel 2.6 running without loosing support for wifi.
if im not online the other 2 guys can use a normal(non-conference) connection to talk and if im online my debian is, too providing an asterisk server with meetme.

the output of the router running asterisk -c -vvv when i try to connect with user “ma” to conf:

    -- Executing Dial("SIP/ma-10046a68", "SIP/asterisk:passwort@192.168.137.111 /router") in new stack
    -- Called asterisk:passwort@192.168.137.111 /router
    -- SIP/192.168.137.111 /router-1004bfb0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Dial("SIP/ma-10046a68", "SIP/"asterisk:passwort@192.168.137.111"") in new stack
Jun  8 21:27:23 WARNING[1179]: chan_sip.c:1994 create_addr: No such host: 192.168.137.111"
Jun  8 21:27:23 NOTICE[1179]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Dial("SIP/ma-10046a68", "SIP/asterisk:passwort@obelisk") in new stack
Jun  8 21:27:23 WARNING[1179]: chan_sip.c:1994 create_addr: No such host: obelisk
Jun  8 21:27:23 NOTICE[1179]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Dial("SIP/ma-10046a68", "SIP/asterisk:passwort@192.168.137.111/conf") in new stack
Jun  8 21:27:23 WARNING[1179]: chan_sip.c:1994 create_addr: No such host: 192.168.137.111/conf
Jun  8 21:27:23 NOTICE[1179]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Congestion("SIP/ma-10046a68", "") in new stack

the same action doesnt create any output at all on the debian running asterisk -c -vvv

i spent the last 4 hours trying to figure out how to make it work (tried register command in various ways, tried the examples from voip-info.org/wiki/view/Aste … al+servers )
i hope someone here can show me the (hopefully simple) mistake i made.

thank you

Hi,

I think everything looks fine except your Dial on the router should be like this:

exten => conf,1,Dial(SIP/obelisk/${EXTEN})

i just added

exten => conf,n,Dial(SIP/obelisk/${EXTEN})

to the extension.conf on the router and tried again.

the message on the router when trying to dial “conf”:

    -- Executing Dial("SIP/ma-100439c0", "SIP/obelisk/conf") in new stack
Jun 10 09:52:57 WARNING[766]: chan_sip.c:1994 create_addr: No such host: obelisk
Jun 10 09:52:57 NOTICE[766]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Congestion("SIP/ma-100439c0", "") in new stack
  == Spawn extension (vpnguests, conf, 6) exited non-zero on 'SIP/ma-100439c0'

so i tried adding

exten => conf,n,Dial(SIP/192.168.137.111/${EXTEN})

which leads to

  -- Executing Dial("SIP/ma-10047858", "SIP/192.168.137.111/conf") in new stack
    -- Called 192.168.137.111/conf
Jun 10 09:55:55 NOTICE[778]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '"ich" <sip:ma@192.168.137.245>;tag=as7330bea0'
    -- SIP/192.168.137.111-1004cda0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing Congestion("SIP/ma-10047858", "") in new stack
  == Spawn extension (vpnguests, conf, 7) exited non-zero on 'SIP/ma-10047858'

:frowning:

Today i tested the same configuration with asterisk 1.4.4 on both the router and the debian machine.

Now i get

   -- Executing [conf@vpnguests:1] Dial("SIP/ma-1005df28", "SIP/asterisk:passwort@192.168.137.111") in new stack
    -- Called asterisk:passwort@192.168.137.111
[Jun 11 19:34:05] NOTICE[1700]: chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"ich" <sip:ma@192.168.137.245>;tag=as2e99e6a5'
    -- SIP/192.168.137.111-10063400 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [conf@vpnguests:2] Dial("SIP/ma-1005df28", "SIP/"asterisk:passwort@192.168.137.111"") in new stack
    -- Called asterisk:passwort@192.168.137.111
[Jun 11 19:34:05] NOTICE[1700]: chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"ich" <sip:ma@192.168.137.245>;tag=as07d9b834'
    -- SIP/192.168.137.111-10067ce0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [conf@vpnguests:3] Dial("SIP/ma-1005df28", "SIP/asterisk:passwort@obelisk") in new stack
[Jun 11 19:34:05] WARNING[1709]: chan_sip.c:2738 create_addr: No such host: obelisk
[Jun 11 19:34:05] WARNING[1709]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [conf@vpnguests:4] Dial("SIP/ma-1005df28", "SIP/asterisk:passwort@192.168.137.111/conf") in new stack
[Jun 11 19:34:06] WARNING[1709]: chan_sip.c:2738 create_addr: No such host: 192.168.137.111/conf
[Jun 11 19:34:06] WARNING[1709]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [conf@vpnguests:5] Dial("SIP/ma-1005df28", "SIP/obelisk/conf") in new stack
[Jun 11 19:34:06] WARNING[1709]: chan_sip.c:2738 create_addr: No such host: obelisk
[Jun 11 19:34:06] WARNING[1709]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [conf@vpnguests:6] Dial("SIP/ma-1005df28", "SIP/192.168.137.111/conf") in new stack
    -- Called 192.168.137.111/conf
[Jun 11 19:34:06] NOTICE[1700]: chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"ich" <sip:ma@192.168.137.245>;tag=as19c257b2'
    -- SIP/192.168.137.111-10063400 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [conf@vpnguests:7] Congestion("SIP/ma-1005df28", "") in new stack
  == Spawn extension (vpnguests, conf, 7) exited non-zero on 'SIP/ma-1005df28'

Cant anyone tell me whats wrong?

i have to have the user calling from asterisk to obelisk
in sip.conf on obelisk too, and without a password. If i set a PW It doesnt seem possible to connect.