provided by Star2Billing S.L.

User-Agent: OpenSER (1.3.0-notls (i386/linux))

Thanx i have done just that and it worked!. However now i face another problem, everytime im trying to register x-lite i get "403 forbiden" message what can be the cause of this?
wbr,

LU

Make sure that you are using the right domain when registering.

Ok asibiby, my setup is as follows:

Openser is at domain "A"
A2billing and Mysql Database are at domain "B" IP

Thefore my Openser database/mysql is pointed at domain "B" IP


Should i proxy challenge/ proxy authorize with domain "A" IP or domain "B" IP?

wbr,
LU

I am doing it by proxy challenging with the same domain as openser. So in this case, it would be Domain A.

If you think about it, the users do not have to know about the Asterisk Box domain. They should only see your OpenSER and then OpenSER will dispatch the call through you Asterisk server farm.

Make sure that the openser authorized domains include the one that you intend to use and that the account that you have added to the subscriber is also using the right domain.

Hi Asiby
I can now register two xlite phones located in two different PCs in my internal LAN. How ever whenever i attempt to make peer to peer calls btn them i got message 'Call failed Forbiden'

In my sip.conf file openser is set as type =friend and in a2billing all card/customers type=friend as well the context is set as 'default' throughout. I even recharge MY XLITES with 5 usd each of them just incase credit is needed to make p2p calls!!

Please advice me the best way to tweak this setup to life.
cheers,
LU.

Hi mates,
Look like this development is slowly becoming in active!
Can someone share her/his sip.conf and extensions.conf settings with me, my setup is giving me a hell of problems. as you can see below. I need your pointers in this. How do i create extensions for a2billing? how do i authentificate in asterisk?

#ngrep snapshot of openser

U 168.172.200.70:59888 -> 168.172.200.87:5060
INVITE sip:[email protected] SIP/2.0.
Via: SIP/2.0/UDP 168.172.200.70:59888;branch=z9hG4bK-d87543-2635895dcf5de608-1--d87543-;rport.
Max-Forwards: 70.
Contact: <sip:[email protected]:59888>.
To: "1189900745"<sip:[email protected]>.
From: "kaunda"<sip:[email protected]:5060>;tag=a059a611.
Call-ID: OGNiM2NlODRlNWE1NjA2NzJhZDMwZWYzZDMzMjYzNTk..
CSeq: 2 INVITE.
Session-Expires: 95.
Min-SE: 90.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
Content-Type: application/sdp.
Proxy-Authorization: Digest username="7718804597",realm="168.172.200.87",nonce="488d464fc85197859446c5618b7dafe7184c5e9b",uri="sip:[email protected]",response="f375ed25769da8338c2116a112f99afc",algorithm=MD5.
Supported: timer.
User-Agent: X-Lite release 1011s stamp 41150.
Content-Length: 373.
.
v=0.
o=- 6 2 IN IP4 168.172.200.70.
s=CounterPath X-Lite 3.0.
c=IN IP4 168.172.200.70.
t=0 0.
m=audio 2054 RTP/AVP 107 119 100 106 0 105 98 8 101.
a=fmtp:101 0-15.
a=rtpmap:107 BV32/16000.
a=rtpmap:119 BV32-FEC/16000.
a=rtpmap:100 SPEEX/16000.
a=rtpmap:106 SPEEX-FEC/16000.
a=rtpmap:105 SPEEX-FEC/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:101 telephone-event/8000.
a=sendrecv.

#
U 168.172.200.87:5060 -> 168.172.200.70:59888
SIP/2.0 500 Server Internal Error.
Via: SIP/2.0/UDP 168.172.200.70:59888;branch=z9hG4bK-d87543-2635895dcf5de608-1--d87543-;rport=59888.
To: "1189900745"<sip:[email protected]>;tag=9a17bd4180f96d7136f8b30b25c6947e.86e9.
From: "kaunda"<sip:[email protected]:5060>;tag=a059a611.
Call-ID: OGNiM2NlODRlNWE1NjA2NzJhZDMwZWYzZDMzMjYzNTk..
CSeq: 2 INVITE.
Server: OpenSER (1.3.0-notls (i386/linux)).
Content-Length: 0.

#Asterisk log

[May 30 15:10:04] NOTICE[2679] chan_sip.c: Failed to authenticate user "7718804597"<sip:[email protected]:5060>;tag=$

#my sip.conf

[openser]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
type=friend
context=a2billing
host=168.172.200.87
fromdomain=168.172.200.87
insecure=invite
port=5060
disallow=all
allow=ulaw
canreinvite=no
nat=yes
no=qualify


#my extensions.conf
[general]

static=yes
writeprotect=no
autofallback=no

[a2billing]

exten => 1189900745,1,Dial(SIP/[email protected],30)
exten => 7718804597,2,Dial(SIP/[email protected],30)
exten => _XXXXXXXXXX,3,Dial(SIP/${EXTEN}@168.172.200.87,30)

It's been a while since I've looked at this, but I returned to the issue today as my SIP user base is growing to the point that I fear I may have problems soon.

Having a break from the issue, has been useful.

The reason I parked the project was because of all sorts of issues with attended transfers, MOH to say the least.

Well guess what, after a few hours work I now have a fully functional system, with just one exception, MWI which I'm working on,

To recap.

I replaced the subscribers table with a view of the same name looking at sip buddies

I've removed the include for additional_a2billing_sip.conf as I'm only using the database. This get's around the proxy auth problems

The media problems are now resolved as all traffic goes thru asterisk.

I'll keep everyone posted and share my configs after a short production run. As I said, once I fugure out a way to enable MWI I'm done.

Regards

Well what do you know, it now all works!!!


2 days of testing show a fully functional Openser<->asterisk<-> A2B solution WITH voicemail notification(WMI)!!!


I'll post the solution shortly, once I've cleaned it up (to many 'what the $#@^!~s!! is happening here!!)

Need a little help though. It's been a while since I looked at the A2B code. Can someone give me a clue as to where additional_sip.conf is written out to the config file? I need to make some changes for Openser

regards

its a good news that you have everything working. We're waiting for your good works post.

Please post your configuration !
I would like to test it as well !

Will do shortly. Just cleaning it up for the book.

regards

Excellent

I've got a few servers in my test lab here ready to go. If you need to play on boxes for it, let me know if I can help.

It all works, only thing I need to do is find the code that writes sip_additional.conf and disable the writing out of the secret. (Openser uses the db)

regards

Do you have the code now? I can send you the code that I am using to write my sip_additional.conf