RESOLVED - 2 x Sip Accounts (Only 1 working for PSTN calls)

Asterisk Asterisk Support
1

Hi Experts,

[UPDATE -
Looking at the CLI output when both are active, I get 2 messages about every 30 seconds - “SRV mapped to host sipgate.co.uk, port 5060” - I don’t get messages about the other service providors]

Please can you help…

I have registered 2 SIPGATE accounts (1 for my Wife & General Home, 1 for myself (Private Line)), and set them up on *. Either works independantly if I comment the other out in the config below.

But, I am having a really strange problem (and starting to think that it is SIPGATE at fault!) - I can only dial in on the number for the last sipgate acct registered, the first one ALWAYS reports: “The service cannot be connected” on dialing.

The problem follows the order of registration, NOT a specific number…?

I am sure (well at least hope!) I’m not going mad… Please let me know if you can see any error in my config files

Regards

Dave

[SIP.CONF] - (Only bits related to SIPGATE)

[general]
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
fromdomain=[mydomain]
relaxdtmf=yes
rtpiholdtimeout=300
rtptimeout=60
useragent=Asterisk
nat=yes
dtmfmode=rfc2833
localnet=192.168.0.0/255.255.0.0
externip=[My external IP]

; Used for incomming & Outgoing
register => ACCT1:PASS@sipgate.co.uk/ACCT1
register => ACCT2:PASS@sipgate.co.uk/ACCT2

; Others…
; register => @sip.voipbuster.com
; register => @fwd.pulver.com
; register => @sip.sipdiscount.com
; register => @lvdx.com

[sipgate2]
type=peer
username=ACCT2
secret=PASS
host=sipgate.co.uk
fromuser=ACCT2
fromdomain=sipgate.co.uk
nat=yes
authuser=PASS
dtmfmode=info
context=SIPGATE2-in
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
; allow=gsm

[sipgate1]
type=peer
username=ACCT1
secret=PASS
host=sipgate.co.uk
fromuser=ACCT1
fromdomain=sipgate.co.uk
nat=yes
authuser=PASS
dtmfmode=info
context=SIPGATE1-in
insecure=very
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
; allow=gsm

[extenstions.conf] - Again only relevent stuff

[globals]
SIPGATE1 => SIP/SIPGATE1
SIPGATE2 => SIP/SIPGATE2
VOIPBUSTER => SIP/VOIPBUSTER
FWD => SIP/FWD
SIPDISCOUNT => SIP/SIPDISCOUNT
LVDX = SIP/LVDX

[SIPGATE1]
exten => _X.,1,SetCallerID([MYCALLERID])
exten => _X.,2,Dial(SIP/${EXTEN:1}@sipgate,20,tr)
exten => _X.,3,Congestion
exten => _X.,4,Busy
exten => _X.,5,Hangup

[SIPGATE1-in]
exten => h,1,Hangup
exten => ACCT1,1,Dial(SIP/805,20,tr)

[SIPGATE2]
exten => _X.,1,SetCallerID([MYCALLERID])
exten => _X.,2,Dial(SIP/${EXTEN:1}@sipgate,20,tr)
exten => _X.,3,Congestion
exten => _X.,4,Busy
exten => _X.,5,Hangup

[SIPGATE2-in]
exten => h,1,Hangup
exten => ACCT2,1,Dial(SIP/804&SIP/805&SIP/850,20,tr)

[VOIPBUSTER]
exten => _X.,1,SetCallerID([MYCALLERID])
exten => _X.,2,Dial(SIP/${EXTEN:1}@voipbuster,20,tr)
exten => _X.,3,Congestion
exten => _X.,4,Busy
exten => _X.,5,Hangup

2

AS I asked for anybody to look for error’s, and I haven’t had any reply, could I ask if an * expert could confirm that this looks right!

Thanks in advance

Dave

3

I assume by “dial in” you mean make a call from the PSTN to your sipgate account - and through that to your Asterisk server.

[quote]
[sipgate2]
type=peer
[…][/quote]
I’m surprised you get any incoming calls! “Peer” should be “friend”. “Peer” is for outgoing calls only, “friend” is for incoming and outgoing.

I haven’t really looked any further through your configs.

I’ve got two accounts with sipgate.co.uk and i don’t have any problems with incoming calls on both numbers.

4

Sorry,

That should say friend, (although it is working as peer???) This was lifted from one of the configs after another modification for testing.

And yes, dial-in from PSTN.

Could I possibly see your dual sipgate config files?

Cheers

Dave

5

In sip.conf:

[general]
context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port is 5060
)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=no
tos=lowdelay               ; lowdelay,throughput,reliability,mincost,none
disallow=all
allow=g729                      ; Allow codecs in order of preference
allow=gsm                       ; Note: codec order is respected only in
allow=ilbc
allow=ulaw                      ;       [general]
useragent=~Asterisk PBX         ; Allows you to change the user agent string
canreinvite=no
nat=no                          ; NAT settings 

[ here are register statements that are the same as your example ]

localnet=192.168.0.0/255.255.255.0      ; All RFC 1918 addresses are local
localnet=127.0.0.0/255.255.255.0                                ; networks

[sipgate-1]
type=friend
host=sipgate.co.uk
username=9999999
fromuser=9999999
secret=xxxxxxx
fromdomain=sipgate.co.uk
qualify=3000
authuser=9999999
nat=no 
disallow=all
allow=g729
context=default
dtmfmode=info

[sipgate-2]
type=friend
host=sipgate.co.uk
username=9999999
fromuser=9999999
authuser=9999999
secret=xxxxxxx
host=sipgate.co.uk
qualify=no
nat=no
disallow=all
allow=gsm
allow=ilbc
insecure=very
fromdomain=sipgate.co.uk
dtmfmode=info

Yeah, they’re not precisely identical…

in extensions.conf:

[default]
exten => 9999999,1,Goto(2,1)

exten => 2,1,Dial(SIP/budgetone,15,r)

etc…

6

Thanks for this, you put me on the right track, although your config had the same problem (for me at least :open_mouth: ):

I (effectivley) had to add:

exten => ACCT1,1,Goto(2,1)
exten => ACCT2,1,Goto(2,1)
exten => 2,1,Dial(SIP/budgetone,15,r)

to make it work!

Thanks for the advice