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" <[EMAIL PROTECTED]> Sent: Saturday, November 29, 2008 10:18 PM To: "Asterisk Users" <[email protected]> Subject: Re: [asterisk-users] asterisk / sipura call breaking up on silence?! > [EMAIL PROTECTED] 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 > > -- > http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com > Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > -- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
