Call Drop after 32 seconds

flatpack · **Posted:** Mon Apr 05, 2010 7:35 pm

hi,
i think this is rgith place to ask this question on my test system everything is working fine. it anounce balance and ask for number to dial then after dialing number call stay connected aprox 32seconds but then hangup can some one please tell me what i m doing wrong?

here is results from astersik

Code:

[root@billing ~]# asterisk -r
Asterisk 1.4.30, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.30 currently running on billing (pid = 3482)
Verbosity is at least 3
-- Executing [00999999999999@a2billing:1] Answer("SIP/7209687053-00000010", "") in new stack
-- Executing [00999999999999@a2billing:2] Wait("SIP/7209687053-00000010", "1 ") in new stack
-- Executing [00999999999999@a2billing:3] DeadAGI("SIP/7209687053-00000010", "a2billing.php|1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/7209687053-00000010> Playing 'digits/19' (language 'en')
-- Playing 'euros' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/7209687053-00000010> Playing 'digits/70' (language 'en')
-- <SIP/7209687053-00000010> Playing 'digits/8' (language 'en')
-- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
-- <SIP/7209687053-00000010> Playing 'prepaid-enter-dest' (language 'en')
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/7209687053-00000010> Playing 'digits/9' (language 'en')
-- <SIP/7209687053-00000010> Playing 'digits/hundred' (language 'en')
-- <SIP/7209687053-00000010> Playing 'digits/80' (language 'en')
-- <SIP/7209687053-00000010> Playing 'digits/5' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/7209687053-00000010> Playing 'digits/50' (language 'en')
-- Playing 'prepaid-seconds' (escape_digits=#) (sample_offset 0)
-- AGI Script Executing Application: (DIAL) Options: (sip/voipb/009999999999 39|60|HRrL(59150000:61000:30000))
-- Limit Data for this call:
> timelimit = 59150000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound =
> warning_sound = timeleft
> end_sound =
-- Called voipb/00999999999999
-- SIP/voipb-00000011 is making progress passing it to SIP/7209687053-000000 10
-- SIP/voipb-00000011 answered SIP/7209687053-00000010
-- <SIP/7209687053-00000010> Playing 'prepaid-enter-dest' (language 'en')
-- Playing 'prepaid-invalid-digits' (escape_digits=#) (sample_offset 0)
billing*CLI>

xianlonghe · **Joined:** Tue Apr 06, 2010 12:29 am **Posts:** 4

try changing dialcommand_param to be

,60,iRL(%timeout%:61000:30000)

which removes "H" from its original setting.

Hope it helps!

xianlonghe · **Joined:** Tue Apr 06, 2010 12:29 am **Posts:** 4

If calls are made by a SIP friend or IAX friend, change those dialcommand as well and try..

For example, if calls are made from a SIP friend, then you change dialcommand_param_sipiax_friend in addition.

flatpack · **Posted:** Tue Apr 06, 2010 1:03 pm

Code:

dialcommand_param/dialcommand_param_sipiax_friend        |60|HRrL(%timeout%:61000:30000)  

changed to

Code:

dialcommand_param/dialcommand_param_sipiax_friend         |60|iRrL(%timeout%:61000:30000)  

tried this as well

Code:

dialcommand_param/dialcommand_param_sipiax_friend        |60|iRL(%timeout%:61000:30000)  

changed these settings even left blank but still same problem

manugmedia · **Joined:** Wed Feb 13, 2008 11:11 am **Posts:** 224

Hi flatpack,

Code:

hi, 
i think this is rgith place to ask this question on my test system everything is working fine. it anounce balance and ask for number to dial then after dialing number call stay connected [b]aprox 32seconds[/b] but then hangup can some one please tell me what i m doing wrong?

Would start looking at the network setup, what does asterisk full log say. seems like RTP OK problem. and what phones are you using to test the call.

Rgds
Eamonn

http://www.eatelco.com

flatpack · **Posted:** Sat Apr 10, 2010 3:00 pm

provider

droping call all time, i installed seperate freepbx for testing this just in case i m doing something wrong. but damn provider droping call after 32 seconds. may be its codec issue but he never came back with explanation.
thanks guys,

zl-Dez · **Joined:** Thu Apr 29, 2010 2:45 pm **Posts:** 9

Hello guys, i've got almost the same problem, except i was testing in at lab.
So i have 2 asterisks, with 2 SIP phones, one of them Linksys SPA962, the second is X-lite.
There is a SIP-trunk between 2 asterisk. The call is going fine, and its all ok till 30-32 secs, then my call drops. Debug didnt show the reason, or i didnt find it. I've tried different settings from dialparam, rates, agi-conf, tried usin AGI, DeadAGI, all the same - 32 seconds then drop.

I've downloaded and installed the lastest version of a2b, but there is no subforum with 1.6 version so i'll post it here.
Here is the log guys. Help me with it:

Code:

SIP.CONF for 1st asterisk with billing:

Code:

[trunk]
type=friend
context=a2billing
host=10.0.0.103
fromuser=8804257720
disallow=all
allow=alaw
nat=no
canreinvite=no
dtmfmode=auto
insecure=port,invite

Extensions.conf for 1st asterisk:

Code:

[a2billing]
; CallingCard application
exten => _X.,1,Answer
exten => _X.,2,Wait(2)
exten => _X.,3,AGI(a2billing.php)
exten => _X.,4,Wait(2)
exten => _X.,5,Hangup

Extensions.conf for 2nd asterisk:

Code:

[trunk]
exten => 937631862,1,Dial(SIP/1000)

Sip.conf for 2nd asterisk:

Code:

[trunk]
type=friend
context=trunk
host=10.0.0.201
disallow=all
allow=alaw
nat=no
canreinvite=no
dtmfmode=auto
insecure=port,invite


[1000]
username=1000
type=friend
context=phones
host=dynamic
secret=1000
disallow=all
allow=alaw
accountcode=1000

Plz, help me to solve the problem.
And sorry for my English, i am from Ukraine, so its not native.

acaratt · **Joined:** Fri Mar 05, 2010 7:00 am **Posts:** 133

Hello zl-Dez,

Did you fix this issue? if not, look into Xlite advance option, just remove the check mark at the middle of the setting. I believe says Network....... it is a know issue with the configuration.

Kind regards,

Al

flatpack · **Posted:** Sun Jun 13, 2010 9:39 pm

my provider never came back with any solution, you made my day, thanks m8, this worked without any probelm, now can some one tell me why this was? call is not inactive...

acaratt wrote:

Hello zl-Dez,

Did you fix this issue? if not, look into Xlite advance option, just remove the check mark at the middle of the setting. I believe says Network....... it is a know issue with the configuration.

Kind regards,

Al

acaratt · **Joined:** Fri Mar 05, 2010 7:00 am **Posts:** 133

Hello,

I am glad you are able to make calls. What do you mean with ..call is not active?

I f you referes why the check mark was enable? In the last version is default setting, that did not happen in previous release.

Kind reagrds,

Al

provided by Star2Billing S.L.

Call Drop after 32 seconds

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