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 Post subject: Problem with SIP
PostPosted: Thu Nov 22, 2007 6:12 pm 
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Joined: Thu Nov 22, 2007 5:52 pm
Posts: 17
Hello,

for my first message I want to say a BIG THANk's for Areski and all members of this forum whom make a great job and helpness.

So I have trixbox, installed in centos, a2billing v1.3.0, for my first test, I have 1 customer and the card work fine with a DID number.

But when I try to activate a the SIP of this customer, I have a problem ???

By clicking generate additional_a2billing_sip.conf , it's ok, I see in this file my extension. but when I click in Click here to reload Asterisk Server the server freeze and I wait 30 seconds and I have the error message.

So I tried to reload the asterisk server from trixbox, it's ok, but I can't connect the SIP client(with sipphone) to asterisk server ???

Where is the problem please ?

I use normally the softphone to connect other extensions and make a call with it, so the problem is in the a2billing context and where is it ??

Thank you


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 Post subject:
PostPosted: Thu Nov 22, 2007 7:34 pm 
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Location: Devon, UK
Possibly the Asterisk manager credentials in a2billing.conf are not correct. What do the Asterisk console and your Apache log files say at this moment?

Does manually issuing the 'reload' command to Asterisk change things? Or to approach this from a different direction, is the SIP friend in question in additional_a2billing_sip.conf?


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 Post subject:
PostPosted: Fri Nov 23, 2007 4:14 pm 
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Posts: 17
Hello,

thank's for your help stavros.

The Asterisk manager credentiel is incorrect, what do you meen ??? I haven't understand sorry :oops:


The SIP friend is in additional_a2billing_sip.conf with all parameters, no problem in this file.

Thank you


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 Post subject:
PostPosted: Fri Nov 23, 2007 4:36 pm 
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nazimou wrote:
The Asterisk manager credentiel is incorrect, what do you meen ??? I haven't understand sorry :oops:
I think the reason you are getting the 30 second timeout and an error when you try to reload Asterisk from A2B might be because you haven't set the Asterisk manager user/pass in a2billing.conf correctly.

Quote:
The SIP friend is in additional_a2billing_sip.conf with all parameters, no problem in this file.
So your SIP friend should be able to register. What error are you getting? What does the Asterisk console say when the SIP client tries to register. What does 'sip show peers' say?


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 Post subject:
PostPosted: Fri Nov 23, 2007 5:15 pm 
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Joined: Thu Nov 22, 2007 5:52 pm
Posts: 17
Hello,

what is the login/password of asterisk ? is the same of my trixbox or there is another ones ? and I must change it in
; MANAGER CONNECTION PARAMETERS

123456 is the sip account into the additional_a2billig_sip.conf
Code:
[123456]
type=friend
username=123456
accountcode=123456
amaflags=billing
secret=123456
nat=yes
dtmfmode=RFC2833
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
host=dynamic
context=a2billing
regseconds=0
cancallforward=yes




(Another question please : where is the a2billing context defined ???)


I have in the logs

Code:
Nov 23 17:56:42 DEBUG[2155] db.c: Unable to find key '123456' in family 'IAX/Registry'

Nov 23 18:05:44 VERBOSE[3012] logger.c:
<-- SIP read from mon.add.res.sip:5060:
REGISTER sip:add.res.se.asterisk SIP/2.0
Via: SIP/2.0/UDP mon.add.res.sip:5060;rport;branch=z9hG4bK4E712E2CB41DA95886A9E5BD
From: friend
To: friend
Contact: "friend"
Call-ID: [email protected]
CSeq: 48386 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-PRO build 1082
Content-Length: 0

Nov 23 18:05:44 VERBOSE[3012] logger.c: --- (11 headers 0 lines)Nov 23 18:05:44 VERBOSE[3012] logger.c: --- (11 headers 0 lines)---
Nov 23 18:05:44 DEBUG[3012] acl.c: ##### Testing mon.add.res.sip with 192.168.1.0
Nov 23 18:05:44 DEBUG[3012] chan_sip.c: Target address mon.add.res.sip is not local, substituting externip
Nov 23 18:05:44 DEBUG[3012] chan_sip.c: Allocating new SIP dialog for [email protected] - REGISTER (No RTP)
Nov 23 18:05:44 VERBOSE[3012] logger.c: Using latest REGISTER request as basis request
Nov 23 18:05:44 VERBOSE[3012] logger.c: Sending to mon.add.res.sip : 5060 (NAT)
Nov 23 18:05:44 VERBOSE[3012] logger.c: Transmitting (NAT) to mon.add.res.sip:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP mon.add.res.sip:5060;branch=z9hG4bK4E712E2CAF8742B9B41DA95886A9E5BD;received=mon.add.res.sip;rport=5060
From: friend
To: friend ;tag=as53c16a58
Call-ID: [email protected]
CSeq: 48386 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0


---
Nov 23 18:05:44 NOTICE[3012] chan_sip.c: Registration from 'friend ' failed for 'mon.add.res.sip' - Username/auth name mismatch




Thank you


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 Post subject:
PostPosted: Fri Nov 23, 2007 5:41 pm 
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Location: Devon, UK
nazimou wrote:
is the same of my trixbox ?
You can use the same password if you want but it's less secure that way.
Quote:
and I must change it in ; MANAGER CONNECTION PARAMETERS
That's the correct place to put it, yes.
Quote:
(Another question please : where is the a2billing context defined ???)
In extensions.conf usually, or extensions_custom.conf on Trixbox. To find what to put in this context you need to read the install documentation more thoroughly.

Your problems getting a SIP peer to register are off-topic for this forum, but I'll say that 'sip show peers' might be a useful command to use at the Asterisk console. It also seems that your X-Pro isn't sending "123456" for the username, it's sending "Friend".


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 Post subject:
PostPosted: Fri Nov 23, 2007 6:13 pm 
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Joined: Thu Nov 22, 2007 5:52 pm
Posts: 17
thank you,

In the sip show peers I don't see the peer 123456.

The a2billing context I think that in my case is custom-callingcard. (I have changed it in the sip-friend)

Code:
[custom-callingcard]
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,DeadAGI,a2billing.php
exten => s,4,Wait,2
exten => s,5,Hangup



Now I have changed the soft I use the x-lite not pro, and he send the 123456 correctly, but I can't make a call, I listen 'the number you have dialed is not in service...' in the logs I see that I'm in the from-sip-external context.


Code:
[from-sip-external]

exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Congestion
exten => s,n,Hangup
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)


Thnak's for all your helps, sir


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 Post subject:
PostPosted: Fri Nov 23, 2007 6:23 pm 
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Posts: 2890
Location: Devon, UK
nazimou wrote:
In the sip show peers I don't see the peer 123456.
Sounds like you forgot to include the necessary "#include" line in sip.conf.

It doesn't seem like your installation is complete. Have you read the Trixbox install guide?


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 Post subject:
PostPosted: Sat Nov 24, 2007 5:07 pm 
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Joined: Thu Nov 22, 2007 5:52 pm
Posts: 17
Hello,

So now I have installed the last stable version of trixbox and the v1.3.0 of a2billing.

I had 1 error in installing the v1.3.0 after this command

Code:
mv * /var/lib/asterisk/sounds/

mv : cannot overwrite directory '/var/lib/asterisk/sounds/es'
mv : cannot overwrite directory '/var/lib/asterisk/sounds/fr'


and when I see in the /var/lib/asterisk/sounds/fr directory, I see another directory fr/ with lot of gsm files, where there isn't in the /var/lib/asterisk/sounds/fr ?? the same with es/

Can I mv all the files from /var/lib/asterisk/sounds/fr/fr/ to /var/lib/asterisk/sounds/fr/ ?

The second problem is also with the sipphone, this time I call make a call with the sip friend but I listen a first ring in the sipphone, after nothing, and in the 22 seconds of the call I listen another time the ring and if I response at this time it's ok but between the 5 second and 22 second I can hear nothing.

I think that is from rtp port, but why he start and after 20 seconds he rering ?

I confirm that I have open the rtp from 10000 -> 50000 I change it in asterisk (rtp.conf)

But NADA
:shock:

Any suggestion please ?


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 Post subject:
PostPosted: Sun Nov 25, 2007 1:02 am 
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Posts: 2890
Location: Devon, UK
nazimou wrote:
Can I mv all the files from /var/lib/asterisk/sounds/fr/fr/ to /var/lib/asterisk/sounds/fr/ ?
Yes, I think that was the intent of the install document.

Quote:
The second problem is also with the sipphone, this time I call make a call with the sip friend but I listen a first ring in the sipphone, after nothing, and in the 22 seconds of the call I listen another time the ring and if I response at this time it's ok but between the 5 second and 22 second I can hear nothing.
If you have "r" (not "R") in your dialcommand_param then Asterisk will enforce alerting as soon it dials out. Look up the "R" and "r" flags on voip-info. I use "R" not "r" for the reasons given there.

Quote:
I think that is from rtp port, but why he start and after 20 seconds he rering ?
I think you've probably been billed for those 20 seconds too. This is most likely a problem at your carrier: false answer. It's not unusual on cheap grey routes to mobile destinations.


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