Hello. I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. I've tried many configuration in sip.conf, but no one solved the problem. Log from /var/log/asterisk/messages: [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call E3F0204B-B35811DE-BBA58889-6D53C3C3 at 83.211.2.220 - no reply to our critical packet. and from CLI: [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 for seqno 101 (Critical Response) [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 - no reply to our critical packet. == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) Any idea? Thanks in advance. -- Gianni Fioretta - gianni.fioretta at yetopen.it YetOpen S.r.l. - http://www.yetopen.it/ Via Previati 72 - 23900 Lecco - ITALY - Tel 0341 220 205 - Fax 178 607 8199 -------- D.Lgs. 196/2003 -------- Si avverte che tutte le informazioni contenute in questo messaggio sono riservate ed a uso esclusivo del destinatario. Nel caso in cui questo messaggio Le fosse pervenuto per errore, La invitiamo ad eliminarlo senza copiarlo, a non inoltrarlo a terzi e ad avvertirci non appena possibile. Grazie.
Hi We've had this a few times and never got to the bottom of exactly why it happens but stopped it by upgrading the phone and the router it was going through to the latest firmware versions. Hope that helps Ish Gianni Fioretta wrote:> Hello. > > I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. > The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. > I've tried many configuration in sip.conf, but no one solved the problem. > > Log from /var/log/asterisk/messages: > [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call E3F0204B-B35811DE-BBA58889-6D53C3C3 at 83.211.2.220 - no reply to our critical packet. > > and from CLI: > [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 for seqno 101 (Critical Response) > [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 - no reply to our critical packet. > == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' > > (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) > > Any idea? > > Thanks in advance. > >-- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
--- On Thu, 10/15/09, Gianni Fioretta <gianni.fioretta at yetopen.it> wrote:> I have a problem with Asterisk, sometimes it hangs up an > external call after 20 seconds, apparently without any > reason. > The call comes from a SIP serverHi, This may or may not apply to your case: https://issues.asterisk.org/view.php?id=14652 Vieri
Gianni Fioretta wrote:> Hello. > > I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason. > The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer. > I've tried many configuration in sip.conf, but no one solved the problem. > > Log from /var/log/asterisk/messages: > [Oct 8 15:49:05] WARNING[10659] chan_sip.c: Hanging up call E3F0204B-B35811DE-BBA58889-6D53C3C3 at 83.211.2.220 - no reply to our critical packet. > > and from CLI: > [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 for seqno 101 (Critical Response) > [Oct 8 15:52:26] WARNING[10659]: chan_sip.c:1972 retrans_pkt: Hanging up call 59E62874-B35911DE-9A598915-C5A6B3AA at 195.62.226.16 - no reply to our critical packet. > == Spawn extension (incoming, 03411885583, 4) exited non-zero on 'SIP/03411885583-081e0d78' > > (peer 101 was not connected at this time, but Asterisk also hags up with all the peers connected) > > Any idea? > > Thanks in advance. > >This is a very weird Asteriskism that we see from time to time. Some SIP servers don't route ACK packets properly (or there will be an ACK loop). The nature of ACK packets is.... tenuous at best in the SIP world. Many clients don't even send them. Asterisk relies heavily on ACK packets to determine if a call is currently connected. If it doesn't receive one, it hangs up the call, even if the rest of the packets have been routed properly and the call is working fine. There's no configuration to turn this off, but there is a way to remove the check in the code. I can't recall the appropriate line to comment out, though. Perhaps someone else knows? In an ideal world, when Asterisk sent an ACK, whatever server/client it was connected to would respond accordingly. It is, however, not an ideal world, so this doesn't always happen. N.
----- "Kevin P. Fleming" <kpfleming at digium.com> ha scritto: | Da: "Kevin P. Fleming" <kpfleming at digium.com> | A: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> | Inviato: Luned?, 19 ottobre 2009 14:03:53 | Oggetto: Re: [asterisk-users] Calls hang up after 20 seconds | | SIP wrote: | | > In an ideal world, when Asterisk sent an ACK, whatever server/client | it | > was connected to would respond accordingly. It is, however, not an | ideal | > world, so this doesn't always happen. | | This is not correct; there are no responses to SIP ACK messages. In | addition. ACK messages are *required* for proper SIP operation; lack | of | an ACK to a response from Asterisk absolutely requires that Asterisk | assume that either the response was never delivered to the requester, | or | that that requester has stopped responding. In either case, the SIP | dialog/transaction in question must be terminated, because it is no | longer in a determinate state. | | If the SIP network does not route ACK responses properly, it is | broken. The SIP network from SIP server (ie EuteliaVoIP) to Asterisk? Internal network works correctly, internal calls are ok. Can I do something to favour the route of ACK responses with my firewall? Maybe opening, or forwarding something? Now port 5060 is opened in TCP and UDP, and ports from 10000 to 20000 are opened in UDP only. Another "Asterisksm": if I restart Asterisk, initially calls works... after 1 or 2 hours every call hangs up after 20 seconds. Any suggestion would be appreciated. | | -- | Kevin P. Fleming | Digium, Inc. | Director of Software Technologies | 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA | skype: kpfleming | jabber: kpfleming at digium.com | Check us out at www.digium.com & www.asterisk.org | | _______________________________________________ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | AstriCon 2009 - October 13 - 15 Phoenix, Arizona | Register Now: http://www.astricon.net | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users -- Gianni Fioretta - gianni.fioretta at yetopen.it YetOpen S.r.l. - http://www.yetopen.it/ Via Previati 72 - 23900 Lecco - ITALY - Tel 0341 220 205 - Fax 178 607 8199 -------- D.Lgs. 196/2003 -------- Si avverte che tutte le informazioni contenute in questo messaggio sono riservate ed a uso esclusivo del destinatario. Nel caso in cui questo messaggio Le fosse pervenuto per errore, La invitiamo ad eliminarlo senza copiarlo, a non inoltrarlo a terzi e ad avvertirci non appena possibile. Grazie.