Problem with calling "normal" numbers via SIP

reindje · #1 (**permalink**) August 10th, 2005, 02:29 PM

When i let Asterisk call a Internet phone number (exten => 2,3,Dial(SIP/0307110319@budgetphone.nl,20,rT)) everything works fine, but when I let asterisk call my mobile number (exten => 4,1,Dial(SIP/06**506476@budgetphone.nl,40)), asterisk return a error:

---------------------------------------------

8 headers, 0 lines

linuxbak*CLI>

Sip read:

SIP/2.0 403 PSTN calls are forbidden

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK172f97f5;rport=1027 ;received=145.97.216.214

From: "06***06476" <sip:06***06476@192.168.0.1>;tag=as6aa21211

To: <sip:06***06476@budgetphone.nl>;tag=9b5971f23d1887 2ff678d4e9dae023f8.280e

Call-ID: 74bac71044fff5bf39f1292f514b0c75@192.168.0.1

CSeq: 102 INVITE

Server: Sip EXpress router (0.8.14-6 (i386/linux))

Content-Length: 0

8 headers, 0 lines

Transmitting:

ACK sip:06***06476@budgetphone.nl SIP/2.0

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK172f97f5

From: "06***06476" <sip:06***06476@192.168.0.1>;tag=as6aa21211

To: <sip:06***06476@budgetphone.nl>;tag=9b5971f23d1887 2ff678d4e9dae023f8.280e

Contact: <sip:06***06476@192.168.0.1>

Call-ID: 74bac71044fff5bf39f1292f514b0c75@192.168.0.1

CSeq: 102 ACK

User-Agent: Asterisk PBX

Content-Length: 0

(no NAT) to 81.23.228.150:5060

Aug 10 15:07:41 WARNING[5065]: chan_sip.c:6811 handle_response: Forbidden - wrong password on authentication for INVITE to '"06***06476" <sip:06***06476@192.168.0.1>;tag=as6aa21211'

-- SIP/budgetphone.nl-9045 is circuit-busy

Reliably Transmitting:

CANCEL sip:06***06476@budgetphone.nl SIP/2.0

Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK172f97f5

From: "0***06476" <sip:06***06476@192.168.0.1>;tag=as6aa21211

To: <sip:06***506476@budgetphone.nl>

Contact: <sip:06***06476@192.168.0.1>

Call-ID: **bac71044fff5bf39f1292f514b0c75@192.168.0.1

CSeq: 102 CANCEL

User-Agent: Asterisk PBX

Content-Length: 0

(no NAT) to 81.23.228.150:5060

Scheduling destruction of call '**bac71044fff5bf39f1292f514b0c75@192.168.0.1' in 15000 ms

== Everyone is busy/congested at this time

When I dial my mobivle via my handitone with the same account evrything works also fine. So there is a problem in asterisk calling other numbers via SIP than internet phone numbers via SIP.

Has anybody a suggestion for this one?

Thanks!

chandave · #2 (**permalink**) August 10th, 2005, 04:19 PM

It appears that bugetphone.nl allows anonymous calling to phone numbers associated with an VOIP client. But, when you want to dial a real phone number, you must provide valid credentials that specify who you are.

Do you have a peer entry in your sip.conf looks like:

Code:

[budgetphone.nl]
type=peer
host=budgetphone.nl
fromdomain=budgetphone.nl
username=account_from_budgetphone.nl
secret=password_from_budgetphone.nl

You will something like the above in order for Asterisk to send the necessary authentication information to budgetphone.nl's SIP Proxy.

See ya...

d.c.

reindje · #3 (**permalink**) August 10th, 2005, 07:28 PM

Thanks for helping me!

When I inserted the following in sip.conf:

[budgetphone.nl]
type=peer
secret=****
username=***307110317
fromdomain=budgetphone.nl
host=sip.budgetphone.nl

Asterisk won't connect any more. Then I can't even call asterisk. Is there something more that I must do?

Agian thanks!

chandave · #4 (**permalink**) August 11th, 2005, 11:20 AM

Quote:

Originally Posted by reindje

Thanks for helping me!

When I inserted the following in sip.conf:

[budgetphone.nl]
type=peer
secret=****
username=***307110317
fromdomain=budgetphone.nl
host=sip.budgetphone.nl

Asterisk won't connect any more. Then I can't even call asterisk. Is there something more that I must do?

Agian thanks!

I'm assuming your are using budgetphone.nl just for dialing outgoing calls, correct? If not, then for incoming calls routed through budgetphone.nl, are you already registering your presence via "register => ...." in sip.conf's [general] section?

What is the console logs when you try to dial through budgetphone.nl?

See ya...

d.c.

reindje · #5 (**permalink**) August 11th, 2005, 02:40 PM

Hi,

yes I use budgetphone for incomming en outgoing calls. But i have to accounts and when i enter the seccond account (this account works also fine with my sip phone) Is there the same problem.

When i entered any peer user from budgetphone like:

[budgetphone]
type=peer
secret=***
username=3***7
fromuser=3***7
host=sip.budgetphone.nl

I get the following message:

Aug 11 10:36:27 WARNING[7139]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 69a439c70b9dc46238d791a420ca513e [!at] 192.168.0.1 (replace the [!at] with a @) for seqno 102 (Critical Request)
Aug 11 10:36:41 NOTICE[7139]: chan_sip.c:4027 sip_reg_timeout: -- Registration for '31***317@budgetphone.nl' timed out, trying again

When i delete the [budgetphone] section, everythins works fine, i can call asterisk and asterisk can forward the call to another budgetphone number. But i cannot dial an other number (for exampel my mobile)...

mhillerbrand · #6 (**permalink**) August 11th, 2005, 03:52 PM

Should type=friend, since you are doing both in and outbound calling? Have you checked registration with sip show peers? Mike.

reindje · #7 (**permalink**) August 11th, 2005, 07:58 PM

Sorry but if I do:

[budgetphone]
type=friend
secret=***
username=3***7
fromuser=3***7
fromdomain=budgetphone.nl
host=sip.budgetphone.nl

Asterisk don't connect at all. the command "sip show peer" returns then:

budgetphone.nl/ 81.23.228.150 255.255.255.255 5060 Unmonitored

But the asterisk dont register on my sip acount. When i delelete the above section in sip.conf asterisk does connect but then there is no peer.....

chandave · #8 (**permalink**) August 12th, 2005, 05:56 AM

It would really help to see your sip.conf, extensions.conf, and console logs so we can see:

How you are registering with budgetphone.nl
What extension and context you are assigning to handle incoming calls from budgetphone.nl
What SIP messages you are getting back from budgetphone.nl
What SIP peer entity you are using for your Dial() application

See ya...

d.c.

reindje · #9 (**permalink**) August 13th, 2005, 02:03 PM

Ok, here are my files:

------------------------------
SIP.conf
--------------------------------

[general]
context=default ; Default context for realm=budgetphone.nl ; Realm for digest authentication
disallow=all
allow=ilbc
allow=speex
allow=ulaw
allow=alaw
allow=gsm
canreinvite=no
port=5060
bindaddr=0.0.0.0

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
maxexpirey=3600 ; Max length of incoming registration we allow
defaultexpirey=120
relaxdtmf=yes
rtptimeout=60
rtpholdtimeout=300
nat=yes

register => 3130***:***@budgetphone.nl:5060

externip = 192.168.0.1
localnet=192.168.0.0/255.255.255.0

[sip_proxy-out]
type=peer
secret=***
username=3***7
fromuser=3***7

;[budgetphone]
;type=peer
;secret=***
;username=3***7
;fromuser=3***7
;host=sip.budgetphone.nl

------------------------------
Extensions.conf
--------------------------------
exten => s,1,Ringing
exten => s,2,Wait,3 ; Wait a second, ju
exten => s,3,Answer ; Answer the line
exten => s,4,Playback(welkom_n)
exten => s,5,Background(een_n)
exten => s,6,Background(belgie)
exten => s,7,Background(twee_n)
exten => s,8,Background(vragenbestelling_n)
exten => s,9,Background(drie_n)
exten => s,10,Background(overigevragen_n)
exten => s,11,ResponseTimeout(20)

exten => t,1,Playback(timeout)
exten => t,2,Hangup

exten => 1,1,Goto(submenu,s,1)

exten => 2,1,Background(ordernr_n)
exten => 2,2,Background(doorverbinden_n)
exten => 2,3,Dial(SIP/0307***9@budgetphone.nl,20,rT)
exten => 2,4,Background(doorvmislukt_n)
exten => 2,5,Voicemail(1234)

exten => 3,1,Background(doorverbinden_n)
exten => 3,2,Dial(SIP/0307***19@budgetphone.nl,20,rT)
exten => 3,3,Background(doorvmislukt_n)
exten => 3,4,Voicemail(1234)

exten => *,1,VoicemailMain(1234)

;exten => 4,1,Dial(SIP/06*****@budgetphone.nl,20,rT)

exten => 5,1,Background(nietgeldig)
exten => 5,2,Goto(s,5)

exten => 6,1,Background(nietgeldig)
exten => 6,2,Goto(s,5)

exten => 7,1,Background(nietgeldig)
exten => 7,2,Goto(s,5)

exten => 8,1,Background(nietgeldig)
exten => 8,2,Goto(s,5)

exten => 9,1,Background(nietgeldig)
exten => 9,2,Goto(s,5)

exten => 0,1,Background(nietgeldig)
exten => 0,2,Goto(s,5)

exten => #,1,Background(nietgeldig)
exten => #,2,Goto(s,5)

[submenu]
exten => s,1,Playback(belgie0)
exten => s,2,Background(een_n)
exten => s,3,Background(belgie1)
exten => s,4,Background(twee_n)
exten => s,5,Background(belgie2)
exten => s,6,Background(drie_n)
exten => s,7,Background(overigevragen_n)

exten => s,8,ResponseTimeout,20

exten => 1,1,Background(belgie1a)
exten => 1,2,Goto(s,1)

exten => 2,1,Background(belgie2a)
exten => 2,2,Goto(s,1)

exten => 3,1,Background(doorverbinden_n)
exten => 3,2,Dial(SIP/0307110319@budgetphone.nl,20,rT)
exten => 3,3,Background(doorvmislukt_n)
exten => 3,4,Voicemail(1234)

exten => 4,1,Background(nietgeldig)
exten => 4,2,Goto(s,1)

exten => 5,1,Background(nietgeldig)
exten => 5,2,Goto(s,1)

exten => 6,1,Background(nietgeldig)
exten => 6,2,Goto(s,1)

exten => 7,1,Background(nietgeldig)
exten => 7,2,Goto(s,1)

exten => 8,1,Background(nietgeldig)
exten => 8,2,Goto(s,1)

exten => 9,1,Background(nietgeldig)
exten => 9,2,Goto(s,1)

exten => t,1,Playback(timeout)
exten => t,2,Hangup
-------------------------------------------------

When somebody dial 2 or 3 asterisk calls my other bugetphone number. Thats works fine. But when somebody push 4 asterisk should call mine mobile numberr and that goes wrong. Then asterisk returns the messeges wich are in the first topic.

Asterisk is running on my firewall with an internal and external network adapter. The "Externip=192.168.0.1" is a bit strange but when i changed it in my external ip asterisk could not regitser....

Is this the information you want?

mberlant · #10 (**permalink**) August 14th, 2005, 12:51 AM

Why doesn't selection number 4 look like selections 2 and 3 (except for the target phone number, of course)? First of all, exten 4,1 is commented out with a semicolon. Also, you don't play any message first, although that should not be a factor. Try taking out the semicolon and see if it starts working.