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

[Philipp Kempgen]

R_O_L_A_N_D at hotmail.com schrieb:

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

> 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

> --------------------------------------------------
> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>

>> R_O_L_A_N_D at hotmail.com schrieb:

>>> 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].

On first sight this looks like a SIP signaling / NAT problem.
You could try
qualify=yes
rtpkeepalive=15


   Philipp Kempgen

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