Bbpglobal


#1

Is there anybody who managed to configure Asterisk as a SIP client to BBPglobal? (http://www.bbpglobal.com)
Can you tell me how you did it? (in sip.conf)

I have Free World Dialup and Talkin2ya/Budgetphone (=Dutch SIP provider) working well. But I can’t get bbpglobal working in Asterisk. If I use SJphone, I am able to use BBPglobal, but not in Asterisk…

sip.conf:

[general]
allow=all
context = incoming
maxexpirey=180
defaultexpirey=160
tos=reliability
videosupport=yes
bindaddr=0.0.0.0
externip = 123.123.123.123
localnet = 192.168.12.0/255.255.255.0

register => 123456:secret1@fwd.pulver.com/s
register => 1234567:secret2@sip2.bbpglobal.com/s
register => 31123456789:secret3@budgetphone.nl/s


[fwd]
type=friend
secret=secret1
username=123456
host=fwd.pulver.com
fromdomain=fwd.pulver.com
insecure=very
context=incoming-fwd

[bbp]
type=friend
secret=secret2
host=sip2.bbpglobal.com
fromuser=1234567
fromdomain=bbpglobal.com
username=1234567
insecure=very
nat=no
qualify=yes
;port=5160 ;for sip1.bbpglobal.com

[budgetphone]
type=friend
host=budgetphone.nl
fromuser=31123456789
fromdomain=budgetphone.nl
username=31123456789
insecure=very
nat=no
secret=secret3
qualify=no
port=5060
context=incoming-budgetphone

“sip show registry” on the Astrisk CLI looks fine:

Host                            Username       Refresh State
budgetphone.nl:5060             31123456789        145 Registered
sip2.bbpglobal.com:5060         1234567            145 Registered
fwd.pulver.com:5060             123456             145 Registered

However when I dial one of their access numbers and enter my extensioncode, I get:
“The number you are dialing is busy or offline…”

When I log on to their network with SJphone and follow the above procedure, the SJphone rings. So something is wrong in the way Asterisk connect to BBPglobal…

To study the problem I did on the Asterisk CLI “sip debug ip sip2.bbpglobal.com”, resulting in:

May 29 19:58:13 NOTICE[21019]: chan_sip.c:4024 sip_reregister:    -- Re-registration for  1234567@sip2.bbpglobal.com
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip2.bbpglobal.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK75e62ff1
From: <sip:1234567@sip2.bbpglobal.com>;tag=as725422e4
To: <sip:1234567@sip2.bbpglobal.com>
Call-ID: 679f74e842857a95495619a41dbe9f59@192.168.12.1
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="1234567", realm="sip2.bbpglobal.com", algorithm=MD5, uri="sip:sip2.bbpglobal.com", nonce="854080120l", response="5666a7674b93234dc0a8104b1c99b2b4", opaque=""
Expires: 160
Contact: <sip:s@123.123.123.123>
Event: registration
Content-Length: 0

 (no NAT) to 210.9.75.106:5060
sammy*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK75e62ff1
To: <sip:1234567@sip2.bbpglobal.com>
From: <sip:1234567@sip2.bbpglobal.com>;tag=as725422e4
Date: Mon, 30 May 2005 04:34:30 GMT
Call-ID: 679f74e842857a95495619a41dbe9f59@192.168.12.1
CSeq: 106 REGISTER
Contact: <sip:s@123.123.123.123>;expires=160
Content-Length: 0


9 headers, 0 lines
May 29 19:58:14 NOTICE[21019]: chan_sip.c:6814 handle_response: Outbound Registration: Expiry for sip2.bbpglobal.com is 160 sec (Scheduling reregistration in 145000 ms)

Also when I dial myself on bbpglobal via their Australian pstn access number, I see only these registration messages, no extra communication from bbpglobal trying to connect to my Asterisk box…

Any suggestions how to fix this problem or how to find the cause of the problem?


#2

I just noticed that not all messages that I see after “sip debug ip sip2.bbpglobal.com” are the same. I get some “loop detected” messages:

Reliably Transmitting:
OPTIONS sip:210.9.75.106 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK5e181cfd
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as13f44472
To: <sip:210.9.75.106>
Contact: <sip:asterisk@123.123.123.123>
Call-ID: 0a432ec541c5e03169675bd0445139e6@123.123.123.123
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Sun, 29 May 2005 18:16:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 210.9.75.106:5060
sammy*CLI>

Sip read:
SIP/2.0 482 Loop Detected,Please Use DNS not ip address
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK5e181cfd
To: <sip:210.9.75.106>
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as13f44472
Call-ID: 0a432ec541c5e03169675bd0445139e6@123.123.123.123
CSeq: 102 OPTIONS
Content-Length: 0


7 headers, 0 lines
Destroying call '0a432ec541c5e03169675bd0445139e6@123.123.123.123'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:210.9.75.106 SIP/2.0
Via: SIP/2.0/UDP 145.94.59.172:5060;branch=z9hG4bK019e3dd1
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as73ed4910
To: <sip:210.9.75.106>
Contact: <sip:asterisk@123.123.123.123>
Call-ID: 32e2a83d404ea0d275e9221734d6f726@123.123.123
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Sun, 29 May 2005 18:17:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 210.9.75.106:5060
sammy*CLI>

Sip read:
SIP/2.0 482 Loop Detected,Please Use DNS not ip address
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK019e3dd1
To: <sip:210.9.75.106>
From: "asterisk" <sip:asterisk@123.123.123.123>;tag=as73ed4910
Call-ID: 32e2a83d404ea0d275e9221734d6f726@123.123.123.123
CSeq: 102 OPTIONS
Content-Length: 0

#3

Did you have any luck with this??

I have managed to get my BBPG Asterisk combination working for incoming calls.

If you need a hand, I can help out, although my knowledge is limited, but I enjoy experimenting with it.

Regards,
L8