Re: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE
This will typically happen over internet connections. If the qualify message is lost, or takes too long the * server will stop sending calls. This is the normal function of qualify. I find that in most cases it is a matter of the end user saturating his connection on his end, assuming you are not overloading yours. On Jul 16, 2006, at 10:13 PM, Tong wrote: According to your console output it looks like there is some major latency. What is the average ping time from your asterisk machine to the polycom phone? - Original Message - From: Rana Dutt To: Asterisk Users Sent: Sunday, July 16, 2006 6:51 PM Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE I have a customer with a Polycom 501 phone behind a NAT. His phone is connected to his Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail. This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174 Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280] type=peer username=280 secret=280 host=dynamic dtmfmode=rfc2833 callerid=John 280 context=company_x mailbox=280 nat=yes canreinvite=no qualify=5000 We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.1/389 - Release Date: 7/14/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE
I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail.This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280]type=peerusername=280secret=280host=dynamicdtmfmode=rfc2833callerid=John 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE
According to your console output it looks like there is some major latency. What is the average ping time from your asterisk machine to the polycom phone? - Original Message - From: Rana Dutt To: Asterisk Users Sent: Sunday, July 16, 2006 6:51 PM Subject: [asterisk-users] Polycom phone cycles between UNREACHABLE and REACHABLE I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail.This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280]type=peerusername=280secret=280host=dynamicdtmfmode=rfc2833callerid="John" 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.10.1/389 - Release Date: 7/14/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
