Remote phones lose connection

reynolwi

Joined
May 5, 2008
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#1
Ok... I got the phones connected after all my static problems got solved but now Ive noticed a different problem. They all register and connect to elastix and work perfectly with no static, have 2way audio, no audio distortion so everything is working great for the first few minutes. Then few minutes later all phones but 1 drops connection then few minutes after that the last phone drops its connection. Its not the same phone either that stays online the longest its a different phone that stays connected longer everytime. After about 20 minutes of elastix retransmitting trying to reconnect each extension all the phones re-register and everything works again. All the phones are Grandstream phones... 2 GXP2000, 1 GXP2020, and 2 BT200. I copied the CLI info and pasted it below...

From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1882c600
To: <sip:3013@10.25.19.102:5064;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 51ed443e69f4686247a237371ba02e9f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
== Manager 'admin' logged off from 127.0.0.1
Retransmitting #3 (NAT) to 74.197.181.236:33470:
OPTIONS sip:3013@10.25.19.102:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK25238a32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1882c600
To: <sip:3013@10.25.19.102:5064;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 51ed443e69f4686247a237371ba02e9f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (NAT) to 74.197.181.236:33470:
OPTIONS sip:3013@10.25.19.102:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK25238a32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1882c600
To: <sip:3013@10.25.19.102:5064;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 51ed443e69f4686247a237371ba02e9f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '51ed443e69f4686247a237371ba02e9f@10.25.18.100' Method: OPTIONS
Reliably Transmitting (NAT) to 74.197.181.236:43927:
OPTIONS sip:3012@10.25.19.103:5063;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK6c422b32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1738cdfb
To: <sip:3012@10.25.19.103:5063;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (NAT) to 74.197.181.236:43927:
OPTIONS sip:3012@10.25.19.103:5063;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK6c422b32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1738cdfb
To: <sip:3012@10.25.19.103:5063;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 74.197.181.236:58037:
OPTIONS sip:3014@10.25.19.104:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK1fa25d8b;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as3e27bd85
To: <sip:3014@10.25.19.104:5065;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 518f84bb017461a71a2a8edf07354a5f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #2 (NAT) to 74.197.181.236:43927:
OPTIONS sip:3012@10.25.19.103:5063;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK6c422b32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1738cdfb
To: <sip:3012@10.25.19.103:5063;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (NAT) to 74.197.181.236:58037:
OPTIONS sip:3014@10.25.19.104:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK1fa25d8b;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as3e27bd85
To: <sip:3014@10.25.19.104:5065;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 518f84bb017461a71a2a8edf07354a5f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #3 (NAT) to 74.197.181.236:43927:
OPTIONS sip:3012@10.25.19.103:5063;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK6c422b32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1738cdfb
To: <sip:3012@10.25.19.103:5063;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 74.197.181.236:49298:
OPTIONS sip:3010@10.25.19.100:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK3253f8b1;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as046b11ec
To: <sip:3010@10.25.19.100:5061;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 3f78997e6cd93759151e6288312551d6@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #2 (NAT) to 74.197.181.236:58037:
OPTIONS sip:3014@10.25.19.104:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK1fa25d8b;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as3e27bd85
To: <sip:3014@10.25.19.104:5065;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 518f84bb017461a71a2a8edf07354a5f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (NAT) to 74.197.181.236:43927:
OPTIONS sip:3012@10.25.19.103:5063;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK6c422b32;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as1738cdfb
To: <sip:3012@10.25.19.103:5063;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '28ba8afa793d877a1a7f6d110f206ef7@10.25.18.100' Method: OPTIONS
Retransmitting #1 (NAT) to 74.197.181.236:49298:
OPTIONS sip:3010@10.25.19.100:5061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK3253f8b1;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as046b11ec
To: <sip:3010@10.25.19.100:5061;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 3f78997e6cd93759151e6288312551d6@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
Retransmitting #3 (NAT) to 74.197.181.236:58037:
OPTIONS sip:3014@10.25.19.104:5065;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.25.18.100:5060;branch=z9hG4bK1fa25d8b;rport
From: "Unknown" <sip:Unknown@10.25.18.100>;tag=as3e27bd85
To: <sip:3014@10.25.19.104:5065;transport=udp>
Contact: <sip:Unknown@10.25.18.100>
Call-ID: 518f84bb017461a71a2a8edf07354a5f@10.25.18.100
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 22 Jun 2009 02:51:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
elastix*CLI>
Disconnected from Asterisk server
Executing last minute cleanups


The firewall at the remote location where these phones are is a pfSense box. The phones are set with a different SIP port and I set the local rtp port of each phone different as well. The sip ports are spaced 1 port apart and the rtp ports are 10 apart.

so ext. 3010 is SIP 5061 & RTP 10020, ext. 3011 is SIP 5062 & RTP 10030, and so on. There are 5 extensions at that location. All the phones connect externally so they all point to a public url which is set in the sip_nat.conf file. On the pfSense box i have each sip port opened and pointed to each phones respective static ip address and then for each phone ip i also have all the rtp ports opened (10000-20000). so each phone has its respective sip port and all the rtp ports open.

I am going to guess its the pfsense box causing this because I noticed when i was watching the CLI each phone extension is showing a different port number on the external ip it registered with and those ports are not open nor why its showing that instead of its sip port... Retransmitting #3 (NAT) to 74.197.181.236:58037:
OPTIONS sip:3014@10.25.19.104:5065;transport=udp SIP/2.0.
Its a different port number for every phone and I do not see where it is coming up with this port number since every phone has its own sip and rtp port.

Is this what is causing it? If it is how do I solve this? This is the first time Ive come across this and im a litte confused. Ive gone back and checked all the phones, elastix, even the pfsense box and just dont understand why they drop connection and the 20 minutes later they reconnect like nothing happened and then the cycle starts over and they start dropping connections again after a few minutes.
 

bucasia

Joined
Feb 15, 2009
Messages
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#2
If you've not got it set already you could try setting "qualify=yes" on the extension/s.
 

reynolwi

Joined
May 5, 2008
Messages
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#3
qualify=yes is set on all the extensions and nat=yes is also set on each extension. They all reconnect and then the cycle starts over again after a few minutes is what is throwing me off.
 

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