[asterisk-users] Calls are dropped after 15 minutes
What version of asterisk are you on? Marlon Araujo > On Aug 10, 2016, at 13:00, asterisk-users-requ...@lists.digium.com wrote: > > Calls are dropped after 15 minutes -- _ -- 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] Calls are dropped after 15 minutes
The solution that fixed our problem was to Edit the sip_general_additional.conf file by adding the line "session-timers=refuse" Thank you to each one who gave suggestions. Keith Keith Heppner Rio Grande Bible Institute 4300 S Business Highway 281 Edinburg, TX 78539-9650 Office 956-380-8171 Cell 956-335-6576 fax 956-380-8258 www.riogrande.edu -- _ -- 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] Calls are dropped after 15 minutes
Set session-timers=refuse in sip.conf and do a sip reload. We had this problem with a handful of devices and this ultimately stopped the issue. Thanks, Derek B. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Stapleton Sent: Tuesday, August 02, 2016 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls are dropped after 15 minutes SIP re-invite (http://www.voip-info.org/wiki/view/SIP+method+invite+re-invite) may be an issue as well. -- _ -- 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] Calls are dropped after 15 minutes
SIP re-invite (http://www.voip-info.org/wiki/view/SIP+method+invite+re-invite) may be an issue as well. On Sat, Jul 30, 2016 at 2:07 PM, Keith Heppner wrote: > We have a problem in that calls are dropped after 15 minutes (on both > internal and out going calls, incoming calls do not seem to have that limit) > How do we fix it? > > This is the version on that PBX > > Kernel >Linux(x86_64)-2.6.18-371.1.2.el5 > > Elastix >elastix-2.4.0-8 >elastix-a2billing-1.9.4-5 >elastix-addons-2.4.0-10 >elastix-agenda-2.4.0-14 >elastix-asterisk-sounds-1.2.3-1 >elastix-email_admin-2.4.0-6 >elastix-endpointconfig2-2.4.0-2 >elastix-extras-2.4.0-5 >elastix-fax-2.4.0-4 >elastix-firstboot-2.4.0-4 >elastix-framework-2.4.0-19 >elastix-im-2.4.0-2 >elastix-my_extension-2.4.0-6 >elastix-pbx-2.4.0-18 >elastix-portknock-0.0.1-0 >elastix-reports-2.4.0-10 >elastix-security-2.4.0-9 >elastix-system-2.4.0-13 > > RoundCubeMail >RoundCubeMail-0.3.1-12 > > Mail >postfix-2.3.3-6.el5 >cyrus-imapd-2.3.7-12.el5_7.2 > > IM >openfire-3.7.1-1 > > FreePBX >freePBX-2.11.0-17 > > Asterisk >asterisk-11.13.0-0 >asterisk-perl-1.03-0 >asterisk-addons-11.13.0-0 > > FAX >hylafax-4.3.10-2rhel5 >iaxmodem-1.2.0-2 > > DRIVERS >dahdi-2.10.0.1-0 >rhino-0.99.6-3.b4 >wanpipe-util-7.0.10-2 > > Thank you, > > Keith > > > -- > _ > -- 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
Re: [asterisk-users] Calls are dropped after 15 minutes
I had a similar issue and i set a timeout which fixed the issue SIP/trunk/ ${EXTEN},216,t We only had this on one of our providers the rest we havent had the issue - Original Message - From: Steve Edwards To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sat, 30 Jul 2016 20:27:45 +0200 (SAST) Subject: Re: [asterisk-users] Calls are dropped after 15 minutes On Sat, 30 Jul 2016, Keith Heppner wrote: > We have a problem in that calls are dropped after 15 minutes (on both > internal and out going calls, incoming calls do not seem to have that > limit) How do we fix it? You may gain some insight from viewing the console output after bumping up the debug and verbose levels. You will probably resolve this by using tcpdump to capture packets and wireshark to see what's happening. I had a problem with a similar description that was resolved by refusing SIP session timers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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
Re: [asterisk-users] Calls are dropped after 15 minutes
I've seen calls drop after 10 mins when SIP session timers are enabled. Try setting them to refuse in sip.conf. On 07/30/2016 02:07 PM, Keith Heppner wrote: We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? -- if at first you don't succeed, skydiving isn't for you -- _ -- 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] Calls are dropped after 15 minutes
On Sat, 30 Jul 2016, Keith Heppner wrote: We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? You may gain some insight from viewing the console output after bumping up the debug and verbose levels. You will probably resolve this by using tcpdump to capture packets and wireshark to see what's happening. I had a problem with a similar description that was resolved by refusing SIP session timers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Calls are dropped after 15 minutes
We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have that limit) How do we fix it? This is the version on that PBX Kernel Linux(x86_64)-2.6.18-371.1.2.el5 Elastix elastix-2.4.0-8 elastix-a2billing-1.9.4-5 elastix-addons-2.4.0-10 elastix-agenda-2.4.0-14 elastix-asterisk-sounds-1.2.3-1 elastix-email_admin-2.4.0-6 elastix-endpointconfig2-2.4.0-2 elastix-extras-2.4.0-5 elastix-fax-2.4.0-4 elastix-firstboot-2.4.0-4 elastix-framework-2.4.0-19 elastix-im-2.4.0-2 elastix-my_extension-2.4.0-6 elastix-pbx-2.4.0-18 elastix-portknock-0.0.1-0 elastix-reports-2.4.0-10 elastix-security-2.4.0-9 elastix-system-2.4.0-13 RoundCubeMail RoundCubeMail-0.3.1-12 Mail postfix-2.3.3-6.el5 cyrus-imapd-2.3.7-12.el5_7.2 IM openfire-3.7.1-1 FreePBX freePBX-2.11.0-17 Asterisk asterisk-11.13.0-0 asterisk-perl-1.03-0 asterisk-addons-11.13.0-0 FAX hylafax-4.3.10-2rhel5 iaxmodem-1.2.0-2 DRIVERS dahdi-2.10.0.1-0 rhino-0.99.6-3.b4 wanpipe-util-7.0.10-2 Thank you, Keith -- _ -- 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