[asterisk-users] asterisk / sipura call breaking up on silence?!

R_O_L_A_N_D at hotmail.com R_O_L_A_N_D at hotmail.com
Sun Nov 30 14:04:31 CST 2008


Hi Philipp,

wht I mean with "a bit"  1 minute. almost always the same.. it varies 
between 1 minute and 1 minute and 6 seconds.

this is the output on CLI:

Nov 30 21:57:10] WARNING[23213]: chan_sip.c:1946 retrans_pkt: Maximum 
retries exceeded on transmission 
NzQxZGExNjZlOWQyYzhhOTdmZWY4ZmI1M2U1OTdiZWY. for seqno 102 (Critical 
Request)
[Nov 30 21:57:10] WARNING[23213]: chan_sip.c:1970 retrans_pkt: Hanging up 
call NzQxZGExNjZlOWQyYzhhOTdmZWY4ZmI1M2U1OTdiZWY. - no reply to our critical 
packet.
  == Spawn extension (spa, 02, 3) exited non-zero on 'SIP/179-0824c8b0'


I;ve added the following under my general context in sip.conf but the 
problem remains the same:

rtpholdtimeout = 150 ; Max number of seconds of inactivity before 
terminating a call on hold or with no activity
rtptimeout= 30 ; Number of seconds, to wait for RTP traffic before classify 
the connection as discontinued


any advice on how to move on with this problem?!


best,

Roland
--------------------------------------------------
From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
Sent: Saturday, November 29, 2008 10:18 PM
To: "Asterisk Users" <asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] asterisk / sipura call breaking up on 
silence?!

> R_O_L_A_N_D at hotmail.com schrieb:
>
>> I'm facing a problem that's occurring more often..
>>
>> my setup is as such :
>>
>> PSTN- Sipura 3102- Asterisk- siphones.
>>
>>
>> whenever a call takes place both outbound as well as inbound, if there 
>> were
>> a bit of silence, the channel gets closed.
>> if there were a bit of latency, the system detects it as silence, and it
>> closes as well..
>
> How much is "a bit"? A second? 10 seconds?
>
> If the phone supports VAD[1]: disable that.
> Make sure rtptimeout in sip.conf is at least 60 [seconds].
>
> [1] http://en.wikipedia.org/wiki/Voice_activity_detection
> [2] http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout
>
>   Philipp Kempgen
>
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