Im Lost - cant setup outgoing calls!

Rayz · November 17, 2006, 11:44am

Hello guys, It’s been 4 days and I still cant setup my outgoing calls! I posted once, read some docs and still cant figure it out, so you are my only hope…

I do not have my phone line (I use cell) so I got my number online from sipnumber.com/ -they offer free phone and free incoming calls so it’s a great piece to test all the stuff. I setup my incoming dial plan and it works just fine, but I cant call using my VOIP line (I added money and I can call STRAIGHT from softphone) but when I dial from asterisk, I’m getting welcome message sipnumber.com and nothing happens…
Can u please tell me what is my mistake?

PS and one more thing - when I type type=friend OR peer for this connection, it does not allow incoming calls and I can place that weird outgoing call I described above, when I change type to user, Asterisk gets the call and route the calls, but when I use exten => _01 I get No Such Host warning…

-----------
SIP.CONF
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
insecure=very
register => 31939*****:********@sipnumber.net


[sipnumber]
type=peer
fromuser=31939*****
secret=********
host=sipnumber.net
dtmfmode=rfc2833
username=31939*****

-----------
extensions.conf

exten => _01.,1,Dial(SIP/sipnumber/${EXTEN:2})

Please help. thank you.

vinod.vijayan · November 17, 2006, 12:09pm

turn on sip debugging by typing

“sip debug” on the asterisk CLI.

then post the SIP messages that are generated during registration.

baconbuttie · November 17, 2006, 12:10pm

either create 2 sections in sip.conf, 1 as a user (for incoming calls) and the other as a peer for the outgoing calls, or construct your dialstring so that username and password get passed as part of the dial.

Rayz · November 17, 2006, 12:44pm

vinod.vijayan
It’s here: mod.pc-zone.net/debug_sip.zip

baconbuttie
Thanks for answer, can u guide me in right direction (not door ) and explain what do u mean?

baconbuttie · November 17, 2006, 1:19pm

[code]; in sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
insecure=very
externip = 123.123.123.123 ; your public ip address
nat = yes
disallow = all
allow = gsm
allow = ulaw
allow = alaw
context = from-sip-unknown ; where unknown callers go to.
dtmfmode = auto
callerid = Unknown
checkmwi = 1
notifyringing = yes
register => 31939*****:********@sipnumber.net

[sipnumber]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
type=peer
nat=yes
insecure=very
qualify=yes
fromuser=31939*****
fromdomain=sipnumber.net
fromuser=31939*****
secret=********
host=sipnumber.net
username=31939*****

[31939*****]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
username=31939*****
type=user
secret=********
qualify=no
nat=yes
insecure=very
host=80.249.108.21
fromdomain=proxy.voip.co.uk
context=incoming-sipnumber

; extensions.conf
[from-sip-unknown]
exten => _X.,1,NoOp(Incoming SIP call from unknown caller)
exten => _X.,2,Congestion()
exten => _X.,3,Hangup()

[incoming-sipnumber]
exten => _X.,1,NoOp(Incoming SIP call from SIPNUMBER)
exten => _X.,2,Dial(SIP/200|30|tw)
exten => _X.,3,VoiceMail(200|ug(5))
exten => _X.,4,Hangup()

[outbound] ; this is a context your phones have access to
exten =>_01.,1,Dial(SIP/sipnumber/${EXTEN:2}|60|TW)
[/code]

that should work, i don’t have access to a sip.conf right now to check it again. you might want to check it against the wiki (search for sip.conf)

Rayz · November 21, 2006, 3:30am

Question:
what is it?
host=80.249.108.21
fromdomain=proxy.voip.co.uk

And another one: why asterisk does not register same SIP server for incoming and outgoing calls?

Is it me, or it’s more complicated than I think?

baconbuttie · November 21, 2006, 3:39am

well it’s obviously part of my config that i missed out removing !! surely you can work out from that what you need to do.

is your second question related to the first ?

Rayz · November 21, 2006, 5:14am

Thank you.

Your config works, but I still have same problems:

I can’t use same sip provider to call AND receive calls at the same time.
It works for incoming when I don’t register outbound setting (when I set type=user). When I set type=friend then I can call out, but cant receive calls.
When I call out I get “Thank you for using us” message from sipnumber.com instead of connection. (When I dial straight from X-Lite with sipnumber.com setting, I can call and receive calls)

baconbuttie · November 21, 2006, 5:31am

post your sip.conf here

Rayz · November 21, 2006, 5:39am

Confings pretty much empty…

[code]
SIP

[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
insecure=very
externip = 67.167..; your public ip address
nat = yes
disallow = all
allow = gsm
allow = ulaw
allow = alaw
context = from-sip-unknown ; where unknown callers go to.
dtmfmode = auto
callerid = Unknown
checkmwi = 1
notifyringing = yes
register => 319*******:********@sipnumber.net

[sipnumber]
type=friend
context=default
host=sipnumber.net
username=319*******
secret=qwerty
fromuser=********
disallow=all
allow=gsm
allow=ulaw
allow=alaw

[3000]
type = friend
context=default
username = 3000
host = dynamic
mailbox = 3000
dtmfmode=rfc2833

[3001]
type = friend
context=default
username = 3001
host = dynamic
mailbox = 3001
dtmfmode=rfc2833

extensions

[general]
static=yes
writeprotect=no

[globals]
[from-sip-unknown]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
include => internal

[sipnumber]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
include => internal

[outbound] ; this is a context your phones have access to
exten => _01.,1,Dial(SIP/sipnumber/${EXTEN:2}|60|TW)

[internal]
exten => 3000,1,Dial(SIP/3000,30,Ttm)
exten => 3000,2,Playback(invalid)
exten => 3000,3,Hangup
exten => 3000,102,Voicemail(u3000)
exten => 3000,103,Hangup
exten => 3001,1,Dial(SIP/3001,30,Ttm)
exten => 3001,2,Playback(invalid)
exten => 3001,3,Hangup
exten => 3001,102,Voicemail(u3001)
exten => 3001,103,Hangup

[default]
include => from-sip-unknown
include => sipnumber
include => outbound
include => internal
[/code][/b]

baconbuttie · November 21, 2006, 5:47am

as i suggested earlier in the thread, who not create 2 entries in sip.conf, 1 for outgoing, 1 for incoming ?

Rayz · November 21, 2006, 6:13am

sip:

[general] 
context=default 
port=5060 
bindaddr=0.0.0.0 
srvlookup=yes 
insecure=very 
externip = ************
nat = yes 
disallow = all 
allow = gsm 
allow = ulaw 
allow = alaw 
context = from-sip-unknown ; where unknown callers go to. 
dtmfmode = auto 
callerid = Unknown 
checkmwi = 1 
notifyringing = yes 
register => 31939*****:********@sipnumber.net 

[sipnumber]
type=peer 
context=default
host=sipnumber.net 
username=31939****
secret=********
fromuser=********
disallow=all 
allow=gsm 
allow=ulaw 
allow=alaw 

[31939*****] 
type=user 
host=sipnumber.net 
username=31939**** 
secret=******** 
host=sipnumber.net 
context=incoming-sipnumber 

[3000]   
type = friend
context=default
username = 3000
host = dynamic
mailbox = 3000
dtmfmode=rfc2833

[3001]   
type = friend
context=default
username = 3001
host = dynamic
mailbox = 3001
dtmfmode=rfc2833

ext:

[general]
static=yes
writeprotect=no

[globals]

[from-sip-unknown] 
exten => s,1,Wait,1			; Wait a second, just for fun
exten => s,2,Answer			; Answer the line
exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)	; Play some instructions
include => internal

[incoming-sipnumber] 
exten => s,1,Wait,1			; Wait a second, just for fun
exten => s,2,Answer			; Answer the line
exten => s,3,DigitTimeout,5		; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10		; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)	; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)	; Play some instructions
include => internal

[outbound] ; this is a context your phones have access to 
exten => _01.,1,Dial(SIP/sipnumber/${EXTEN:2}|60|TW) 

[internal]
exten => 3000,1,Dial(SIP/3000,30,Ttm)
exten => 3000,2,Playback(invalid)
exten => 3000,3,Hangup
exten => 3000,102,Voicemail(u3000)
exten => 3000,103,Hangup
exten => 3001,1,Dial(SIP/3001,30,Ttm)
exten => 3001,2,Playback(invalid)
exten => 3001,3,Hangup
exten => 3001,102,Voicemail(u3001)
exten => 3001,103,Hangup

[default]

include => from-sip-unknown
include => incoming-sipnumber
include => sipnumber
include => outbound
include => internal

With this confing same story

baconbuttie · November 21, 2006, 6:39am

post a log file fragment for a failed call.

Rayz · November 21, 2006, 6:54am

Situation: can’t call in, can partially call out (call out type=peer)
When I call from my cell to voip number I get message “contact not available”. When I call trough Asterisk, it connect me right away to server (it will not connect me with wrong user) and I’m getting “Welcome Message” instead of connection with recipient…

Situation: CAN call in, can’t call out (call out type=user )
I can dial number from my cell age get to Asterisk.
and get:
create_addr: No such host: sipnumber // because type is user

I posted dedug fragment before mod.pc-zone.net/debug_sip.zip

Could it be sipnumber.com error? If I give you my acc# there, can u check it?

vinod.vijayan · November 21, 2006, 8:31am

The link has file with 0 bytes.

Upload the file again or just paste the messages in this forum.

The link has file with 0 bytes.

Upload the file again or just paste the messages in this forum.

Rayz · November 21, 2006, 9:30am

OMG!
I digged out sipnumber.net and figured out that they are freedigits.com sub devision!

Used freedigits.net as host and everything worked!

THX guy for your time and PATIENCE.

especially baconbuttie

Topic		Replies	Views
SIP Dialing Issues Asterisk Support	3	215	June 24, 2008
Problem setting SIP incoming/outgoing Asterisk Support	2	394	October 11, 2005
Can? Asterisk Support	3	204	March 27, 2009
PROBLEM IN MAKING OUTGOING CALLS using SIp Asterisk Support	27	1162	May 3, 2007
480 Temporarily Unavailable Asterisk Support	0	1205	July 21, 2008

Im Lost - cant setup outgoing calls!

[code] SIP

extensions

Related topics

[code]
SIP