Re: [asterisk-users] Remote extensions call drops after 20 seconds.
Thanks Steve. I think my problem is NAT. I'm using iptables, but I don't sure if I'm doing right steps. In the principal router I've forwarded the ports, but in my firewall (iptables on PBX server) I'm not sure. 201.237.180.154 is my remote place. #El NAT para el 5060 y el 1-3 (rtp) iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j MASQUERADE iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT Can somebody help me to configure my NAT on iptables ? Maybe an example. Thank you again. On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro stot...@totarotechnologies.com wrote: Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.comwrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] Remote extensions call drops after 20 seconds.
Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should be a NAT issue? On Thu, Mar 13, 2014 at 8:43 AM, alp...@gmail.com alp...@gmail.com wrote: Thanks Steve. I think my problem is NAT. I'm using iptables, but I don't sure if I'm doing right steps. In the principal router I've forwarded the ports, but in my firewall (iptables on PBX server) I'm not sure. 201.237.180.154 is my remote place. #El NAT para el 5060 y el 1-3 (rtp) iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j MASQUERADE iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT Can somebody help me to configure my NAT on iptables ? Maybe an example. Thank you again. On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro stot...@totarotechnologies.com wrote: Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.comwrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.comwrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr
Re: [asterisk-users] Remote extensions call drops after 20 seconds.
Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
[asterisk-users] Remote extensions call drops after 20 seconds.
Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don't work for the specific environment. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Thank you Eric for your reply. How Can I fix it? In server side, I opened RTP ports. On Wednesday, December 18, 2013, Eric Wieling wrote: Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don't work for the specific environment. -Original Message- From: asterisk-users-boun...@lists.digium.com javascript:; [mailto: asterisk-users-boun...@lists.digium.com javascript:;] On Behalf Of alp...@gmail.com javascript:; Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com javascript:; Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
here's a checklist... First, RTP port range not port forwarded correctly on the NAT router (check rtp.conf). Then, on sip.conf: externip not correctly setup (it should be the public IP of the NAT router)? nat setting not enabled for any outbound trunk and the extensions (nat=yes) ? localnet not properly setup (to include subnets of local, un-nat'd extensions) ? canreinvite not disabled for any outbound trunk and for the extensions? rgds On Wed, Dec 18, 2013 at 8:34 PM, alp...@gmail.com alp...@gmail.com wrote: Thank you Eric for your reply. How Can I fix it? In server side, I opened RTP ports. On Wednesday, December 18, 2013, Eric Wieling wrote: Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don't work for the specific environment. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
Rodrigo, thanks for reply. 1- RTP ports is forwarded correctly on the NAT router. 2- externip is my public ip. 3- All my extensions have nat=yes by default. 4- localnet is setup. 5- canreinvite is disabled. It could be a codec mistake? On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: here's a checklist... First, RTP port range not port forwarded correctly on the NAT router (check rtp.conf). Then, on sip.conf: externip not correctly setup (it should be the public IP of the NAT router)? nat setting not enabled for any outbound trunk and the extensions (nat=yes) ? localnet not properly setup (to include subnets of local, un-nat'd extensions) ? canreinvite not disabled for any outbound trunk and for the extensions? rgds On Wed, Dec 18, 2013 at 8:34 PM, alp...@gmail.com alp...@gmail.comwrote: Thank you Eric for your reply. How Can I fix it? In server side, I opened RTP ports. On Wednesday, December 18, 2013, Eric Wieling wrote: Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don't work for the specific environment. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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] Remote extensions call drops after 20 seconds.
What version of Asterisk?directmedia=no should be used in versions of Asterisk 1.8 and later. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 4:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Rodrigo, thanks for reply. 1- RTP ports is forwarded correctly on the NAT router. 2- externip is my public ip. 3- All my extensions have nat=yes by default. 4- localnet is setup. 5- canreinvite is disabled. It could be a codec mistake? On Wed, Dec 18, 2013 at 2:58 PM, Rodrigo Borges Pereira rodrigoborgespere...@gmail.com wrote: here's a checklist... First, RTP port range not port forwarded correctly on the NAT router (check rtp.conf). Then, on sip.conf: externip not correctly setup (it should be the public IP of the NAT router)? nat setting not enabled for any outbound trunk and the extensions (nat=yes) ? localnet not properly setup (to include subnets of local, un-nat'd extensions) ? canreinvite not disabled for any outbound trunk and for the extensions? rgds On Wed, Dec 18, 2013 at 8:34 PM, alp...@gmail.com alp...@gmail.com wrote: Thank you Eric for your reply. How Can I fix it? In server side, I opened RTP ports. On Wednesday, December 18, 2013, Eric Wieling wrote: Calls dropping after 20 seconds is often directmedia enabled when it should not be enabled or RTP keepalives enabled when they should not be enabled. Dropping around 20 mins is often Session Timers being enabled when they don't work for the specific environment. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 3:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Remote extensions call drops after 20 seconds. Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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 -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- 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
Re: [asterisk-users] Remote extensions call drops after 20 seconds.
On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.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
