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 Post subject: Call doesn't Hangup
PostPosted: Wed Oct 26, 2011 1:44 pm 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Hi,

I have a wholesales setup and i can connect to the system using a sip client and make call and the calls are logged and billed correctly. but the problem is when the remote party hangup the calls, a2b doesn't regonise this and its keeps on charging the calls. the weird thing, it only happen when i call to a landline number, its perfectly working fine to a mobile. i though it would be a signalling problem so have tried with different carrier, its the same result. it only behave this way, when remote landline line hangup.

i have attached a sip debug log, for both calls to landline and mobile. much appreciate your help.

Running asterisk 1.8
a2billing 1.9.4
Using AGI in dial plan.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Fri Oct 28, 2011 12:30 am 
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Joined: Fri Jan 18, 2008 9:54 pm
Posts: 135
Hi,

In the file sip.conf you can find a section called "RTP Timers"
Make sure to enable the (rtptimeout , rtpholdtimeout) and choose the suitable values.

This might help, as asterisk will disconnect the call automatically when RTP is absent

Regards,


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 Post subject: Re: Call doesn't Hangup
PostPosted: Fri Nov 04, 2011 8:57 pm 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Thank you Swift for the replay,

rtptimeout does hang up the calls according to the defined seconds but this could cause it to terminate the call when one party is silent for that number of seconds. i'm using this for wholesales deployment, this could lead to inaccurate billing. is there any other solution? or is it something to do with asterisk 1.8.8.0?

Thanks.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Fri Nov 04, 2011 11:02 pm 
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Joined: Mon Jan 08, 2007 6:56 pm
Posts: 345
You seem to have a NAT issue in your setup.

You should not use "nat=yes" indiscriminately. Instead use nat=no or let the default nat=no take effect.

Use nat=yes in the peer config when you have reason to believe the remote peer is behind a NAT. For asterisk 1.8 you can use nat=comedia if you are not sure and it tries to detect it.

When devices and asterisk are behind the same NAT and they communicate with the asterisk these devices will have nat=no.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Sun Nov 06, 2011 2:44 pm 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Hi,

I have tried with different setting but same result. If Nat is an issue then why is working perfectly when calling mobile number and detect hangup. im confused. I Have SIP nat set to No in Asterisk (i don't have the comedia option in freepbx) and have following Sip setting in a2billing:

A2B in Realtime:
id 1
id_cc_card 1
name 6016369339
accountcode 6016369339
regexten 6016369339
amaflags billing
callgroup
callerid
canreinvite yes
context custom-a2billing-sip
DEFAULTip
dtmfmode RFC2833
fromuser
fromdomain
host dynamic
insecure port,invite
language
mailbox
md5secret
nat no
deny
permit
mask
pickupgroup
port 5060
qualify no
restrictcid
rtptimeout
rtpholdtimeout
secret xxxxxxxxx
type friend
username 6016369339
disallow ALL
allow ulaw,alaw,gsm,g729
musiconhold
regseconds 1320593255
ipaddr xx.xx.xx.xx
cancallforward yes
fullcontact
setvar
regserver
lastms 0
defaultuser
auth
subscribemwi
vmexten
cid_number
callingpres
usereqphone
incominglimit
subscribecontext
musicclass
mohsuggest
allowtransfer no
autoframing
maxcallbitrate
outboundproxy
rtpkeepalive 0
useragent Zoiper Communicator 2.05.11136 rev.11135


Thanks


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 Post subject: Re: Call doesn't Hangup
PostPosted: Tue Nov 08, 2011 3:18 am 
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Joined: Mon Jan 08, 2007 6:56 pm
Posts: 345
Check the NAT setting in the trunk definition and the general section of sip.conf


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 Post subject: Re: Call doesn't Hangup
PostPosted: Tue Nov 08, 2011 5:57 am 
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Joined: Fri Jan 18, 2008 9:54 pm
Posts: 135
Hi all,

I am not sure it is related to NAT ,, as NAT can not be that precise and selective (heppening only with certain destinations)

However, I agree that you need to enable the NAT option if your sip clients are behind NAT.

As for the rtptimeout parameter,, it is not a silent detector,, it is an RTP detector.
If one party is silent, you will still be receiving RTP of noise ..

Issues will start if one party is using silence suppression ,, i guess


Hope you help me understand something here :
Quote:
when the remote party hangup the calls, a2b doesn't regonise this and its keeps on charging the calls.


What about asterisk ,, does it detect the end of the call ??
Can you show us CDR for both Astersik and A2B for the same call ?

Regards


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 Post subject: Re: Call doesn't Hangup
PostPosted: Tue Nov 08, 2011 10:49 am 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Hi,

Thanks for the rpttimeout explanation. what i mean by " when the remote party hangup the calls, a2b doesn't regonise this and its keeps on charging the calls." is for example if A (a2b) --CAll--B (Landline no) and B Hang up, A (a2b) doesn't detect the hang up.

Asterisk also doesn't detect the end of the calls, the call log varies as asterisk record the call duration from the moment a number is dialled to call end, where a2b record once the call is answered.

Here is the call Log:

Asterisk: 2011-11-08 10:44:42 SIP/601636... 6016369339 6016369339 00442084706xxx ANSWERED 00:36

A2B: 2011-11-08 10:44:48 6016369339 00442084706xxx 442084706xxx United Kingd 00:30

Thanks.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Tue Nov 08, 2011 1:28 pm 
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Joined: Fri Jan 18, 2008 9:54 pm
Posts: 135
Hello,

Hope you can elaborate here.
The CDR is showing that the call was around 30 something seconds,, where is the problem ,, what was the length of the actual call ?


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 Post subject: Re: Call doesn't Hangup
PostPosted: Tue Nov 08, 2011 6:03 pm 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Hi,

Hope i make it clear this time, when a call is made via a2b, and the call receiver end the call by hanging up the phone, it doesn't detect the hang up signal. the call need to be ended by the caller that Initiated the call in the first place..thats when a2b record the call and bill the account. Like i said earlier in my post this only happens, when calls are made to landline numbers.

Thanks.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Wed Nov 09, 2011 7:27 am 
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Joined: Fri Jan 18, 2008 9:54 pm
Posts: 135
Hi,

We need to be able to detect who is causing it.

when the other party hangs up, there should be no rtp from its side,,

at the Asterisk CLI prompt issue the command :

Code:
rtp set debug on


And watch what happens.


Also,,
Enable the rtptimeout parameter and make a test call but do not hang up from your side and see if the Asterisk can detect the case and automatically cut the call.

It could be a signalling issue related to that certain destination and all the carriers you tried are reaching it through the same sub-route.


Regards,


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 Post subject: Re: Call doesn't Hangup
PostPosted: Wed Nov 09, 2011 2:41 pm 
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Joined: Wed Nov 25, 2009 12:01 pm
Posts: 90
Hi,

I have set the rtp debug on but when the other party hangs up, i am still receiving rtp from other side.

i've also set the following in the asterisk as you have said,
Reinvite Behavior= no
rtptimeout =10
rtpholdtimeout=30

Asterisk doesn't detect the case and doesn't automatically cut the call.

i also get the following warning even though i have set the rtp timer in the sip.conf

[Nov 9 13:58:25] WARNING[13585]: chan_sip.c:26865 build_peer: '' is not a valid RTP hold time at line 0. Using default.
[Nov 9 13:58:25] WARNING[13585]: chan_sip.c:26699 build_peer: no value given for outbound proxy on line 0 of sip.conf.

Im begnning to wonder if its the carrier side? i want to be asolutley sure before i take this up issue to them.


Thank you for the support.


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 Post subject: Re: Call doesn't Hangup
PostPosted: Thu Nov 10, 2011 7:32 am 
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Joined: Fri Jan 18, 2008 9:54 pm
Posts: 135
Hello roby ,

The rtptimeout will not be effective until rtp stops coming from one of the sides,, as long as you keep receiving rtp then Asterisk will not cut the call. This is the correct behavior.

One thing, you need to pay attention to the rtpkeepalive parameter and disable it.

Normally rtp should stop coming from the other side when that side hangs up. This implies that your case is mostly caused by the other side,, the carrier, the operator, or any gateway between the Asterisk and the final destination.

I recommend that you contact them.
Actually whenever you are in doubt, contact (politely) all the involved parts, if the issue is not caused by them at least there is a chance that they guide you where to look.

As for the warning, check the related line and make sure there is no odd characters and that the comments are starting with (;)

Regards,


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