A2billing 1.8 not passing on callerid from PiaF

[ghurty]

Ive been working on a guide based on my pitfalls and I hopefully am almost done.

I have a2billing set up with sip clients registering correctly. I am using asterisk realtime.

The only problem appears to be passing the outbound callerid from the "customer" to the trunk.

It appears to be trying to use the account number as the callerid.

On the a2billing forum jroper suggested not using "Fromuser" as in that would cause such a issue.

And doing that in v 1.3 solved the problem.

However if I do that in 1.8, I get the no service recording. If I put back the fromuser line, it works.


THanks

[2010-10-13 20:44:52] VERBOSE[3466] netsock.c: == Using SIP RTP TOS bits 184
[2010-10-13 20:44:52] VERBOSE[3466] netsock.c: == Using SIP RTP CoS mark 5
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [1XXXXXXXXXX@from-sip-external:1] NoOp("SIP/192.168.1.43-0000001d", "Received incoming SIP connection from unknown peer to 1XXXXXXXXXX") in new stack
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [1XXXXXXXXXX@from-sip-external:2] Set("SIP/192.168.1.43-0000001d", "DID=1XXXXXXXXXX")  in new stack
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [1XXXXXXXXXX@from-sip-external:3] Goto("SIP/192.168.1.43-0000001d", "s,1") in new stack
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Goto (from-sip-external,s,1)
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.1.43-0000001d", "0?checklang:noanonymous") in new stack
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Goto (from-sip-external,s,5)
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:5] Set("SIP/192.168.1.43-0000001d", "TIMEOUT(absolute)=15") in new stack
[2010-10-13 20:44:52] VERBOSE[5209] func_timeout.c: Channel will hangup at 2010-10-13 20:45:07.289 EDT.
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:6] Answer("SIP/192.168.1.43-0000001d", "") in new stack
[2010-10-13 20:44:52] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:7] Wait("SIP/192.168.1.43-0000001d", "2") in new stack
[2010-10-13 20:44:54] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:8] Playback("SIP/192.168.1.43-0000001d", "ss-noservice") in new stack
[2010-10-13 20:44:54] VERBOSE[5209] file.c: -- <SIP/192.168.1.43-0000001d> Playing 'ss-noservice.gsm' (language 'en')
[2010-10-13 20:44:59] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:9] PlayTones("SIP/192.168.1.43-0000001d", "congestion") in new stack
[2010-10-13 20:44:59] VERBOSE[5209] pbx.c: -- Executing [s@from-sip-external:10] Congestion("SIP/192.168.1.43-0000001d", "5") in new stack

Here is the trunk settings from freepbx:

username=YYYYYYYYYYY
type=friend
secret=6481526359
host=XX.XX.XXX.XXX
context=a2billing ; change for proper context
fromuser=YYYYYYYY
allow=g729 ; we support ulaw,alaw,ilbc,gsm,g723.1,g726,g729a
trustrpid=yes
sendrpid=yes
canreinvite=no

 

[jroper]

Trunk settings as below work for me:-

username=YYYYYYYYYYY
type=friend
secret=6481526359
host=XX.XX.XXX.XXX
allow=g729 ; we support ulaw,alaw,ilbc,gsm,g723.1,g726,g729a
trustrpid=yes
sendrpid=yes
canreinvite=no

In inbound settings = YYYYYYYYYYY
secret=6481526359
context=from-trunk

and register string
YYYYYYYYYYY:[email protected]

Make sure you have no caller ID set in A2Billing Voip settings.

fromuser is clearly going to deliver your account number and cannot be used.

Joe

 

[ghurty]

Thank you.

Got that working.

But now I bumped into what appears to be a bug:
After trying to have a incoming DID point to a particular "customer", The username in the sip config list always changes from the username to "s".

And then the only way to recieve the call is to change on the "customer" box the trunk name to be "s".

If it is anything but "s" you get (where "test" is the name of trunk):
[2010-10-14 09:57:22] WARNING[3045] chan_sip.c: username mismatch, have <test>, digest has <s>
[2010-10-14 09:59:42] NOTICE[3045] chan_sip.c: Failed to authenticate user "XXXXXXXXX" <sip:[email protected]>;tag=as36a8058d

If I change the username to something else, it reverts back to me an "s" on the first incoming call.

I have the trunk settings exactly like what you posted, I have the user context to be the account number.

I tried this with multiple sip accounts being created.


Thanks

 

[jroper]

Hi

How are you sending the number to PBX. There are a number of valid ways, e.g.

sip/123456 where 123456 is the account number or
sip/123456/987654321 where 987654321 is the DID or
sip/987654321@IP-Address-Of-PBX but you would have to allow anonymous SIP calls to allow that.

Have you considered using IAX, it is far more friendly, and usually just works.

Joe

 

[rjm]

Ive been working on a guide based on my pitfalls and I hopefully am almost done.

Please let me know when you have that done!