Connect to sipgate

Asterisk Asterisk Support
1

Hi I’ve installed aah 1.4.24 on a spare box

got 2 softfones on laptops talking to each other.

trying to connect to sipgate.co.uk

I’ve now found I need to use the password supplied, but I’ve only tried it with my sipgate id, not the last 7 digits of the phone number as suggested elsewhere.

I’ve set my router to dmz so there shouldn’t be any port issues, or do I have to port forward?

I’ve put details into freepbx, inbound trunk and register string.

I think I have correct nameserver set up, I can nslookup sites.

Not sure where I need to put my ip, externip or externhost though. I on residential adsl so it is dynamic, but I’m using my ip at present to keep it simple before adding my dyndns details(another day/headache!)

My question is from a fresh install what is the minimum to put in to change to status on sipgate.co.uk from offline to online? (i’ve tried logging out/in)

Can all the details be entered through freepbx of do I still need to edit sip_custom_general.conf?

I’ve been on this for a week now and it may end up out the window,

Thanx.

ps. my linux experience is limited!!

2

From the SipGate Homepage:

Enclosed the necessary settings to configure to Asterisk for sipgate. (Very important to have the SIP ID at the end of the register command)

sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = error
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
register => SIPID:PASSWD@sipgate.co.uk/SIPID

[sipgate-out]
type=friend
insecure=invite ; otherwise I get authentication errors
nat=yes
username=SIPID
fromuser=SIPID
fromdomain=sipgate.co.uk
secret=SIPPW
host=sipgate.co.uk
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context = from-sipgate

extension.conf
[sipout]
exten => _X.,1,Set(CALLERID(num)=SIPID)
exten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trg)
exten => _X.,3,Hangup

Alternate versions below:

register => sip-id:sipgatepassword@sipgate.co.uk/sip-id

In the sip.conf:

[sipgate]
type=friend (or peer)
username=(seven digit SIP-ID)
secret=(SIP-Password, found under My Data at sipgate.co.uk)
host=sipgate.co.uk
fromuser=(seven digit SIP-ID)
fromdomain=sipgate.co.uk
nat=yes
authuser=SIP ID
dtmfmode=info (or inband)
context=inbound (or incomingsipgate)
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw

In the extensions.conf :

[incomingsipgate]
exten => h,1,Hangup
exten => seven digit SIP-ID,1,Dial(SIP/internestelefon,20,tr)

[sipgate]
exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
exten => _9.,2,Playback(invalid)
exten => _9.,3,Hangup

An alternate extension model:
[sipgate]

exten => _7.,1,SetCallerID(complete phone number excl. country code i.e. 02076644200)
exten => _7.,2,Dial(SIP/${EXTEN:1}@sipgate,20,tr)
exten => _7.,3,Congestion
exten => _7.,4,Busy
exten => _7.,5,Hangup

[inbound] (Where it says SIP ID, insert your SIP ID number)
exten => SIP ID,1,Answer
exten => SIP ID,2,Dial(Local/1000)

Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources.

We don’t currently support the IAX protocol (but Asterisk supports SIP).

You may also find help and information at the following website:

voip-info.org/wiki-Asterisk