sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!

11.5.3.4. sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!

1. Description

In sip.conf there is an option for every peer called qualify.

If qualify=yes or a numeric value, then asterisk will sometimes poke this peer by sending a "SIP OPTIONS" request to phones or other pbx's.

If they do not reply on time, they will be considered unreachable, and this message will be printed on the asterisk CLI.

When the phone is back online (first time it replies on time) then asterisk will tell you Peer 'XXX' is now REACHABLE, if we got a reply from the phone, but not on time, the message Peer 'XXX' is now too LAGGED will be printed on the CLI.

The timeout is set to 2000ms by default. (If you specify qualify=yes).
But you could also set it to any other value.

e.g. qualify=3000

2. Reasons for seeing this message:

When a phone is rebooted, or when a phone hangs, or when its shut down this message might pop up.

(Or when there is a too big delay on the network).

If all your phones become unreachable at the same time, its probably your asterisk server that has network problems instead of the phone.

When a phone is unreachable, asterisk will not try to call it. (So you might want to set this value not too low, or you might want to completely disable it).

If the phone that has unreachable messages all the time is behind a NAT, it might be that the UDP timeout is set too low on the firewall.

User Comments

erik de wild (info at tripple-o dot nl)
12 June 2008 09:42:02
Just for the record. I had an incident once with all phones (80) unreachable and this was because the poe (power over ethernet) feauture of a well branded switch failed for just a moment. All the phones started to reboot at once and the complete system was out of operation for a while

Steve Krzysiak (steve at prusourceone dot com)
20 May 2008 22:49:49
Thank you for this great explanation. It really cleared a few things up for me. My only concern now is what happens if the qualify time is set too high??

TORRENTE (jtorrente at praxsys dot fr)
18 April 2008 15:35:02
Hi, I would like to receive an e-mail when a peer becomes unreachable or reachable, anybody knows how to do ?
Many Thanks
BR
Joseph
jtorrente@praxsys.fr

Lancelot (amendoza7 at hotmail dot com)
10 December 2007 23:48:08
He implementado una VPN dentro un enlace internet (Peru) para comunicarme con los servidores de telefonia de una empresa en Mexico el cual utiliza la aplicacion zoiper, a pesar que mi firewall deja pasar todo TCP/IP y configurado correctamente el zoiper no se llega a registra en los Log se visualiza "Timeout registration for 3002@192.168.40.2 muy a pesar que si llego a ese servidor haciendo ping.
alguien me puede ayudar con este problemita.

Gracias.

Cristina (cristinanazzaro at gmail dot com)
17 July 2007 16:05:43
Salve ragazzi ho un problema con asterisk: lo status dei peers è unmonitored e lo state della registration è unregistraded ed ho smepre il problem notice 5362 chan_si.c:9610 handle_sesponce_register mi dice che non trova il mio peers..ma lo pinga???
Che problema potrebbe essere????

Cristina (cristinanazzaro at gmail dot com)
17 July 2007 16:05:31
Salve ragazzi ho un problema con asterisk: lo status dei peers è unmonitored e lo state della registration è unregistraded ed ho smepre il problem notice 5362 chan_si.c:9610 handle_sesponce_register mi dice che non trova il mio peers..ma lo pinga???
Che problema potrebbe essere????

toysoft (toysoft at toysoft dot net dot nospam)
28 July 2006 00:45:10
UNREACHABLE Problem. What can I do to fix ? Thank you. TS

Sending to 86.209.149.217 : 10019 (non-NAT)
Transmitting (NAT) to 86.209.149.217:10019:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.209.149.217:10019;branch=z9hG4bKf5ffffffa60affff;received=86.209.149.217
From: "Gordes" <sip:200@sip.saggiori.com>;tag=c0900000cabcffff
To: <sip:200@sip.saggiori.com>
Call-ID: e3770000d7230000@86.209.149.217
CSeq: 162 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200@192.168.1.220>
Content-Length: 0

---
Transmitting (NAT) to 86.209.149.217:10019:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.209.149.217:10019;branch=z9hG4bKf5ffffffa60affff;received=86.209.149.217
From: "Gordes" <sip:200@sip.saggiori.com>;tag=c0900000cabcffff
To: <sip:200@sip.saggiori.com>;tag=as706a6a10
Call-ID: e3770000d7230000@86.209.149.217
CSeq: 162 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200@192.168.1.220>
WWW-Authenticate: Digest realm="asterisk", nonce="180da9b0"
Content-Length: 0

---
Scheduling destruction of call 'e3770000d7230000@86.209.149.217' in 15000 ms
asterisk1*CLI>
<-- SIP read from 86.209.149.217:10019:
REGISTER sip:sip.saggiori.com SIP/2.0
Via: SIP/2.0/UDP 86.209.149.217:10019;branch=z9hG4bKecc8ffffd909ffff
From: "Gordes" <sip:200@sip.saggiori.com>;tag=c0900000cabcffff
To: <sip:200@sip.saggiori.com>
Contact: <sip:200@86.209.149.217:10019>
Authorization: Digest username="200", realm="asterisk", algorithm=MD5, uri="sip:sip.saggiori.com", nonce="180da9b0", response="09098ed331327b68df07fd2f92fa6cad"
Call-ID: e3770000d7230000@86.209.149.217
CSeq: 163 REGISTER
Expires: 300
User-Agent: Grandstream HT386 1.0.2.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

--- (13 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 86.209.149.217 : 10019 (NAT)
Transmitting (NAT) to 86.209.149.217:10019:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.209.149.217:10019;branch=z9hG4bKecc8ffffd909ffff;received=86.209.149.217
From: "Gordes" <sip:200@sip.saggiori.com>;tag=c0900000cabcffff
To: <sip:200@sip.saggiori.com>
Call-ID: e3770000d7230000@86.209.149.217
CSeq: 163 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:200@192.168.1.220>
Content-Length: 0

---
Transmitting (NAT) to 86.209.149.217:10019:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.209.149.217:10019;branch=z9hG4bKecc8ffffd909ffff;received=86.209.149.217
From: "Gordes" <sip:200@sip.saggiori.com>;tag=c0900000cabcffff
To: <sip:200@sip.saggiori.com>;tag=as706a6a10
Call-ID: e3770000d7230000@86.209.149.217
CSeq: 163 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 300
Contact: <sip:200@86.209.149.217:10019>;expires=300
Date: Thu, 27 Jul 2006 17:09:21 GMT
Content-Length: 0

---
Scheduling destruction of call 'e3770000d7230000@86.209.149.217' in 15000 ms
12 headers, 0 lines
Reliably Transmitting (NAT) to 86.209.149.217:10019:
OPTIONS sip:200@86.209.149.217:10019 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK7845c72d
From: "Unknown" <sip:Unknown@192.168.1.220>;tag=as0b4f216d
To: <sip:200@86.209.149.217:10019>
Contact: <sip:Unknown@192.168.1.220>
Call-ID: 04f722c94637407f782ab7f140f1832b@192.168.1.220
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 27 Jul 2006 17:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Retransmitting #1 (NAT) to 86.209.149.217:10019:
OPTIONS sip:200@86.209.149.217:10019 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.220:5060;branch=z9hG4bK7845c72d
From: "Unknown" <sip:Unknown@192.168.1.220>;tag=as0b4f216d
To: <sip:200@86.209.149.217:10019>
Contact: <sip:Unknown@192.168.1.220>
Call-ID: 04f722c94637407f782ab7f140f1832b@192.168.1.220
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 27 Jul 2006 17:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---

Zsolt SZALAI (zs dot szalai at gmail dot com)
30 March 2006 23:03:08
Some providers(eg. NeoPhoneX) requires quality=no setting to work. Try it, if you get such a message.

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