Asterisk Setup for Community VoIP

== CDR updated on Zap/5-1
-- Executing Dial("Zap/5-1", "SIP/00442074127332@voipuser") in new stack
-- Called 00442074127332@voipuser
Jan 12 13:43:39 NOTICE[2399]: chan_sip.c:6750 handle_response: Failed to authenticate on INVITE to '"XXXXXXXXX" <sip:stephen@voipuser.org>;tag=as1a8c50f3'

then it sits there till i hangup then gives

-- Channel 0/2, span 2 got hangup
== Spawn extension (default, 9900442074127332, 1) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
-- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

this is using an ISDN phone connected up (using zapbri) works fine for anything else.

just to update i am registered ok

Host Username Refresh State
voipuser.org:5060 stephen 105 Registered

working now, typo in teh password in my [voipuser] section of sip.conf

have configured * as per the instructions above, but cannot dial out. I notice that asterisk doesn't return a "Registered to..." message for voipuser.org, and when I dial out, I see the following:

Code:

-- Called 0016504750863 [!at] voipuser.org (replace the [!at] with a @) Jan 12 04:44:59 NOTICE[135385088]: chan_sip.c:6807 handle_response: Failed to authenticate on INVITE to '"Ben Calvert" <sip:Unknown@64.81.53.18>;tag=as74f3f6de'   == Spawn extension (outgoing, 4750863, 1) exited non-zero on 'SIP/lorax-11c9'  -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

lorax is the kphone install on my laptop, and obviously i'm Ben Calvert.

Below is my sip config:

Code:

[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=yes                   ; Enable DNS SRV lookups on outbound calls disallow=all                    ; First disallow all codecs allow=ulaw                      ; Allow codecs in order of preference allow=ilbc                      ; Note: codec order is respected only in [general] nat=yes                         ; NAT settings register => flying_walrus:PASSWIRD:flying_walrus@voipuser.org [lorax] type=friend ;callerid="Ben Calvert (650) 475-0863" context=outgoing host=dynamic dtmfmode=inband secret=PASSWORD qualify=yes nat=yes allow=ulaw allow=alaw [voipuser] type=friend context=voicepulse-in username=flying_walrus secret=PASSWORD host=voipuser.org fromuser=flying_walrus fromdomain=voipuser.org insecure=very qualify=no disallow=all allow=ulaw allow=alaw allow=gsm

stephen :

working now, typo in teh password in my [voipuser] section of sip.conf

Ah, that's great!

By the way, you should dial as if you are in the UK, so to dial that uk number, you want 020xxxxxxxx.

--ian

flying_walrus :

Jan 12 04:44:59 NOTICE[135385088]: chan_sip.c:6807 handle_response: Failed to authenticate on INVITE to '"Ben Calvert" <sip:Unknown@64.81.53.18>;tag=as74f3f6de' == Spawn extension (outgoing, 4750863, 1) exited non-zero on 'SIP/lorax-11c9' -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

That's the exact errors that stephen was reporting and he just came back saying he had an incorrect password.

Are you sure your password is correct?

Your sip password is your voipuser.org password at the time you registered for the outbound.

--ian

either it just takes some time to register, or i was suffering from middle of the night syndrome. Thanks for the help

FYI
If the 'realm' keyword is defined in the [general] session of sip.conf, then asterisk does not manage to register with voipuser.org, whereas fwd.pulver.com looks hapyy with and without it.
But there are some authentication issues that I need to investigate.

I have had one way voice all day(unless DMZ) but I finally got it, here is what I ended up with.
Many of these options may not be needed, but I am too tired to test now that it is working.
Just a side note...IAX would solve this mess very nicely.

NAT: port forward 10000-20000 UDP
Port forward 5060 TCP/UDP
port forward 5082 UDP

Here is my sip.conf

[general]
nat=yes
externip=67.172.***.***
localnet=192.168.0.77/255.255.255.0

register => ******:******@voipuser.org/08449869358

[voipuser]
disallow=all
allow=ulaw
type=peer
context=incoming_voipuser
username=******
fromuser=******
secret=******
host=voipuser.org
fromdomain=voipuser.org
insecure=very
qualify=no
nat=yes
;port=5060
;bindaddr=0.0.0.0

___________________________________
It seems to have been the commenting out the last two lines that solved the problem.

Here is the voipuser part of my extensions.conf

[outgoing_voipuser]
exten => _81.,1,Dial(SIP/${EXTEN:2}@voipuser,60)
exten => _81.,2,Congestion


Here is my rtp.conf

[general]
rtpstart=10000
rtpend=20000

Hi Drroller,

Thanks for sharing your experiences.

--ian

no, the iax server is different.

Regards,

Tjardick

How do we configure our Asterisk box for voipuser.org IAX?
If the IAX configuration was included in the first post on this thread it might be used more.

voipuser.org only supports IAX for the UK incoming number correct?

Quote:

voipuser.org only supports IAX for the UK incoming number correct?

Correct.

Quote:

If the IAX configuration was included in the first post on this thread it might be used more.

If you want to copy/paste me the config you mean, I'll edit and add it into the top post.

Thanks,

Dean

Is the IAX server the same as the SIP server?

Hi,

A few posts above got a bit mixed up because of the server time thing so it's a bit hard to follow

Outbound is only SIP, not IAX.

IAX is provided for incoming and there is a forum discussing this (with a sticky thread describing the setup) here:
http://www.voipuser.org/forum_view_9.html

--ian