[ravi28UK]

I am running asterisk@home 2.5 image(preconfigured with latest updates) with vmware player, also tried on standalone PC..

I can't dial any numbers Keep getting all circuits are busy please try again later?? I can't receive any calls on my
voipstunt number either, keeps going to voicemail.. in the DID section I put 00441216604522
I can dial between internal extensions and chat ok.


My network is 192.168.1.0/24 Asterisk@home is static on 192.168.1.222 and all firewall ports on my router are open for
sip to work. I can access amp ok and change all the settings ok.


I also tried setting for my voipfone account - same problem all circuits are busy please try again later??

I have not added anything to extensions.conf maybe this is the problem.

Sorry I am new to Asterisk@home and trying to learn - please help..

My settings..

Outgoing Dial Rules
Dial Rules:

0044+1.
0044+2.
0044+5.
0+0.

Outbound Routing
Dial Patterns
90|.
NXXXXXX

I press 9 then number I want to dial.

In sip.conf: (this is in a trunk called VOIPSTUNT)

voipstunt
type=friend ; (or "peer" if we don't need incoming calls, or if there is a separate section with "type=user")
host=sip.voipstunt.com
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
username={{YOURUSERNAME}}
fromuser={{YOURUSERNAME}}
secret={{YOURPASSWORD}}
qualify=1000 ; optional
canreinvite=no ; new SIP servers don't like reINVITEs

Dial(SIP/00{EXTEN}@voipstunt) >>> Not sure how configure or put this into extensions.conf so missed it out<<<

register => YOURUSERNAME:YOURPASSWORD@sip.voipstunt.com (added this incoming section of trunk)

[jmattwood]

Below is what I use in the outgoing settings of the trunk:

authuser=username
canreinvite=no
context=from-pstn
dtmfmode=inband
fromdomain=voipstunt.com
fromuser=username
host=sip.voipstunt.com
insecure=very
secret=password
type=peer
username=username

And here is the register line I use, again entered on the trunk settings page.

username:password@sip.voipstunt.com/username

I do not use voipstunt for incoming, so I leave the incoming section blank.

[ravi28UK]

Hi jmattwood, thanks for your help.

I have tried changing the trunk setting to what you suggested but still getting "all circuits are busy please try again later" I wonder if the problem is with my dialing rules or dial pattern??? Do I need to make manual changes to any of the conf files? ie. sip.conf , extensions.conf etc? they are all standard at the moment.

Heres a copy of my Asterisk info

- Note Only extension 200 is logged in at the moment and under sip providers sip.voipstunt is coming up as unregistered ?? maybe this is the problem??? I know port 5060 is open for 192.168.1.222 - as the soft phones can logon ok.

Asterisk Info: asterisk1.local (192.168.1.222)

Version
Asterisk 1.2.4 built by root @ asterisk1.local on a i686 running Linux on 2006-02-03 17:06:27 UTC
Verbosity is at least 3




Uptime
System uptime: 11 minutes, 54 seconds
Verbosity is at least 3




Active Channel(s)
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
192.168.1.100 (None) F5DB86AB5E3 00101/63609 unkn No Rx: REGISTER
1 active SIP channel
Verbosity is at least 3




Sip Registry
Name/username Host Dyn Nat ACL Port Status
400/400 (Unspecified) D 0 Unmonitored
200/200 192.168.1.100 D 5060 Unmonitored
2 sip peers [2 online , 0 offline]
Verbosity is at least 3




Sip Peers
Host Username Refresh State
sip.voipstunt.com:5060 [myusername] 120 Unregistered
sip.voipstunt.com:5060 [myusername] 120 Unregistered
Verbosity is at least 3




IAX2 Sip Registry
Host Username Perceived Refresh State
Verbosity is at least 3




IAX2 Peers
Name/Username Host Mask Port Status
0 iax2 peers [0 online, 0 offline, 0 unmonitored]
Verbosity is at least 3




Subscribe/Notify
-= Registered Asterisk Dial Plan Hints =-
400 : SIP/400 State:Unavailable Watchers 0
200 : SIP/200 State:Idle Watchers 0
----------------
- 2 hints registered
Verbosity is at least 3




Zaptel driver info
Chan Extension Context Language MusicOnHold
pseudo from-pstn en
Verbosity is at least 3




Conference Info
No active MeetMe conferences.
Verbosity is at least 3




Voicemail users
Context Mbox User Zone NewMsg
default 200 Ravi MCE2005 0
default 400 Ravi LAPTOP 2
Verbosity is at least 3




NTP peers
remote refid st t when poll reach delay offset jitter
================================================== ============================
*LOCAL(0) LOCAL(0) 10 l 13 64 377 0.000 0.000 0.004

[ravi28UK]

Hi again,

Just checked the log files, for some wierd reason, my username can't be registered with voipstunt???

Registration for '[username]@sip.voipstunt.com' timed out, trying again (Attempt #12)
Feb 18 13:22:24 WARNING[2541] chan_sip.c: No such host: sip.voipstunt.com
Feb 18 13:22:24 WARNING[2541] chan_sip.c: Probably a DNS error for registration to [username]@sip.voipstunt.com, trying REGISTER again (after 20 seconds)

I have tried forwarding port tcp/udp 53 for dns to 192.168.1.222 - same problem!

Ps- username and password are correct , can user voipstunt software to make / receive calls ok.

Any other ideas?? can I provide any more info / logs?

[jmattwood]

Dial rules for voipstunt should send the number in the full international format: 00.Countrycode.Number.

Therefore, a call to a UK number that would normally be dialled 01274666555 would translate to 00441274666555.

A simple test would be to have an outbound route of 00. (include the full stop) then dial the number in full international format, without your current 9

Or...

If you're in the UK as your dial rules & name suggest, the best outgoing route that you could employ might be 0|. select voipstunt as the trunk. Then in your voipstunt trunk, under Outgoing Dial Rules, add 00441274+XXXXXX (where 1274 is your local std) if you have a London STD use 004420+XXXXXXXX (check the length of your local number length & have enough Xs to allow for them all) . In addition, you need to include a rule for full UK numbers. 0044+XXXXXXXXXX

Another couple of things you could look at - what shows up in the Call Detail Reports as the number dialled?

And, from a linux command, log in as root, then type asterisk -vvvvvr and press return. Then make a call & watch the output of the CLI. You should be able to tell exactly what number your system is passing to voipstunt.

[ravi28UK]

Hi again,

Changed outgoing rules to :

0044121+XXXXXXX (as i am in Birmingham)
0044+XXXXXXXXXX

Outbound routing

Dial pattern

0|. (pointing to SIP/Voipstunt)

I dial the number as 044121xxxxxxx - same problem "all lines are busy - please try again later"

I thinks its a problem with my configuration somewhere? as Asterisk info suggests i can't register with voipstunt :

Host Username Refresh State
sip.voipstunt.com:5060 [USERNAME] 120 Unregistered
sip.voipstunt.com:5060 [USERNAME] 120 Unregistered
Verbosity is at least 3

All required firewall ports are open to 192.168.1.222 including udp/tcp 53 for DNS..

Any other ideas.. Thanks..

[ravi28UK]

Hi,

I tried asterisk -vvvvvr..

The number is been dialed correctly ie. 00441211231234 but then ...
Dial failed due to CHANUNAVAIL
Executing Macro("Sip/200-7ce", "outisbusy") in new stack
Playing 'all-circuits-busy-now' (language 'en')
Executing Macro("Sip/200-7ce", "pls-try-call-later") in new stack
Playing 'pls-try-call-later' (language 'en')
etc..

Looks like problem logging onto voipstunt sip ?? How can I resolve this issue??

[jmattwood]

Make sure all relevant ports are forwarded in your router to your asterisk sever.
5060
10000-20000
8000-8012
3478-3483
5004-5036
4560-4570
2727

It's a while since I needed to use port forwarding, so may have missed some (or added too many). I'm also sure it's UDP you need to forward, however, best double check & forward both UDP & TCP.

I had no end of trouble with voipstunt - my account would stop working,. Is yours a new account, or an old one? Have you put any credit on it? Re-run the voipstunt setup executable & create a new account with new credentials - that got mine working again.

You can check if you're registered by looking in the maintenance section of AMP.....Asterisk Info on mine (older version). Or from the CLI (logged in as root, type asterisk -r) type sip show registry & press return.

Incidentally, I cannot dial out using my first voipstunt account, for similar reasons, however, it shows registered. The newer, second account works fine.

[ravi28UK]

Hi again,

Tried forwarding to all those ports with udp/tcp but my router will not allow 5004-5036
4560-4570.. same problem. Voipstunt account is new.. only 5 days old got £10 credit on..
I'll try registering a new account.. with no credit.. see if this works?

Any other ideas.. you sure I don't need edit anything in sip.conf or extensions.conf files?

[ravi28UK]

Same problem with new account .. SIP is unregistered!!..

Is there some sort of test account that I obtain to test my configuration??

jmattwood, I have pm you my new voipstunt user / and password - can you please test on your Asterisk@home server. Thanks for your help.