Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
If it's a reinvite problem, check the domain in the request URI. Might be something it shouldn't be, eg something to do with the VPN. On 11 September 2013 05:28, isr...@gmail.com wrote: Some providers send a reinvite after 15 min and if asterisk doesn't respond will disconnect the call Maybe playaround with canreinvite --Original Message-- From: Jeremy Kister Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes Sent: Sep 10, 2013 10:23 PM On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 drop calls after 15 minutes
Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See https://wiki.a Packet timed out after 6401ms with no response (or 6399ms or 6401ms) Qualify freq being 6 ms for the peers. For the SIP peers I see Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs I tried to use originate for session-timer in global SIP conf, no changes. Any hint about this matter would be appreciate. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
hi, it seems your vpn connection drop. is you vpn on WiFi of any other high latency network? On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI ad...@tootai.netwrote: Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See https://wiki.a Packet timed out after 6401ms with no response (or 6399ms or 6401ms) Qualify freq being 6 ms for the peers. For the SIP peers I see Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs I tried to use originate for session-timer in global SIP conf, no changes. Any hint about this matter would be appreciate. -- Daniel -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
Some providers send a reinvite after 15 min and if asterisk doesn't respond will disconnect the call Maybe playaround with canreinvite --Original Message-- From: Jeremy Kister Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes Sent: Sep 10, 2013 10:23 PM On 9/10/2013 7:05 AM, Administrator TOOTAI wrote: I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk Just for kicks, I would disable session-timers to see if the problem goes away. in the general section and/or each peer in sip.conf: session-timers=refuse -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
