Phones loosing registration periodicaly

Asterisk Asterisk Support
1

Hi,

we use a * 1.2.7.1 with 4 GXP2000. The * is inside a DMZ, all GXP’s behind the FW. After a while the GXP’s are no longer answering any phonecall.

The Debug from * shows:

[code]Scheduling destruction of call ‘6d2be8c07e134b417ab1bd2f20905664@192.168.202.3’ in 32000 ms
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0

— (8 headers 0 lines)—
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060;tag=6f69876048d1f00d
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Contact: sip:121@192.168.202.121:5060
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0

— (10 headers 0 lines)—
Retransmitting #1 (no NAT) to 192.168.202.121:5060:
CANCEL sip:121@192.168.202.121:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060
Contact: sip:111@192.168.202.3
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 CANCEL
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0

accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060;tag=6f69876048d1f00d
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 CANCEL
User-Agent: Grandstream GXP2000 1.1.0.13
Contact: sip:121@192.168.202.121:5060
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer
Content-Length: 0

— (11 headers 0 lines)—
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060;tag=6f69876048d1f00d
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 INVITE
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0

— (8 headers 0 lines)—
Transmitting (no NAT) to 192.168.202.121:5060:
ACK sip:121@192.168.202.121:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060;tag=6f69876048d1f00d
Contact: sip:111@192.168.202.3
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0

Destroying call '6d2be8c07e134b417ab1bd2f20905664@192.168.202.3’
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
SIP/2.0 481 No Such Call
Via: SIP/2.0/UDP 192.168.202.3:5060;branch=z9hG4bK57bea36b;rport
From: “J. Fritz” sip:111@192.168.202.3;tag=as31487c93
To: sip:121@192.168.202.121:5060;tag=933b0a9ad858a725
Call-ID: 6d2be8c07e134b417ab1bd2f20905664@192.168.202.3
CSeq: 102 CANCEL
User-Agent: Grandstream GXP2000 1.1.0.13
Content-Length: 0
[/code]

what does: “SIP/2.0 481 No Such Call” mean?

Also a 407 error comes up some times…

asterisk -rx’restart now’ resolves the problem for some hours, but later on the same error comes up.

All Phones displaying error 603.

Does anyone have any idea and can help us?

Thanks in advance
Jo

2

Some additional info:

The Asterisk shows the following, if we try to call to a Phone with the 603 error:

[code]<-- SIP read from 192.168.202.121:5060:
INVITE sip:111@192.168.202.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKLoxVhocu1MRwYwci
Max-Forwards: 7
User-Agent: Vlines IP-Phone
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
Contact: sip:121@192.168.202.121:5060
CSeq: 1 INVITE
Supported: replaces
Content-Type: application/sdp
Content-Length: 271

v=0
o=- 17843079 67413006 IN IP4 192.168.202.121
s=SIP CALL
c=IN IP4 192.168.202.121
t=0 0
m=audio 5004 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (12 headers 12 lines)—
Using INVITE request as basis request - zBnNbqW7WPQQjKGB@192.168.202.121
Sending to 192.168.202.121 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.202.121:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKLoxVhocu1MRwYwci;received=192.168.202.121
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3;tag=as407edab8
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
CSeq: 1 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:111@192.168.202.3
Proxy-Authenticate: Digest realm=“asterisk”, nonce="61259d5a"
Content-Length: 0

Scheduling destruction of call ‘zBnNbqW7WPQQjKGB@192.168.202.121’ in 15000 ms
Found user '121’
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
ACK sip:111@192.168.202.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKLoxVhocu1MRwYwci
Max-Forwards: 7
User-Agent: Vlines IP-Phone
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3;tag=as407edab8
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
Contact: sip:121@192.168.202.121:5060
CSeq: 1 ACK
Content-Length: 0

— (10 headers 0 lines)—
accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
INVITE sip:111@192.168.202.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKEQe6v5uXus6Wtclv
Max-Forwards: 7
User-Agent: Vlines IP-Phone
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
Contact: sip:121@192.168.202.121:5060
Proxy-Authorization: Digest username=“121”, realm=“asterisk”, nonce=“61259d5a”, uri="sip:111@192.168.202.3", response=“b6ac2bc8026e693f632c2abbcff30763”, algorithm=MD5
CSeq: 2 INVITE
Supported: replaces
Content-Type: application/sdp
Content-Length: 271

v=0
o=- 37097570 08849947 IN IP4 192.168.202.121
s=SIP CALL
c=IN IP4 192.168.202.121
t=0 0
m=audio 5004 RTP/AVP 8 0 3 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

— (13 headers 12 lines)—
Using INVITE request as basis request - zBnNbqW7WPQQjKGB@192.168.202.121
Sending to 192.168.202.121 : 5060 (non-NAT)
Found user '121’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.202.121:5004
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 111 in 121 (domain 192.168.202.3)
list_route: hop: sip:121@192.168.202.121:5060
Transmitting (no NAT) to 192.168.202.121:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKEQe6v5uXus6Wtclv;received=192.168.202.121
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
CSeq: 2 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:111@192.168.202.3
Content-Length: 0

Reliably Transmitting (no NAT) to 192.168.202.121:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKEQe6v5uXus6Wtclv;received=192.168.202.121
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3;tag=as2a08f1cf
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
CSeq: 2 INVITE
User-Agent: Vlines accessVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:111@192.168.202.3
Content-Length: 0

accessvoip*CLI>
<-- SIP read from 192.168.202.121:5060:
ACK sip:111@192.168.202.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.202.121:5060;branch=z9hG4bKEQe6v5uXus6Wtclv
Max-Forwards: 7
User-Agent: Vlines IP-Phone
From: “121” sip:121@192.168.202.3;tag=BNhD4HmRKN9HMgqf
To: “111” sip:111@192.168.202.3;tag=as2a08f1cf
Call-ID: zBnNbqW7WPQQjKGB@192.168.202.121
Contact: sip:121@192.168.202.121:5060
Proxy-Authorization: Digest username=“121”, realm=“asterisk”, nonce=“61259d5a”, uri="sip:111@192.168.202.3", response=“a0a87e853f3ce7fbbd51006b98a27dff”, algorithm=MD5
CSeq: 2 ACK
Content-Length: 0
[/code]

sip.conf:

[general]
useragent=Vlines accessVoIP
port=5060
insecure=very
context=default
tos=lowdelay
defaultexpire=1200
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=g729


[111]
type=friend
username=111
secret=xxxxxx
callerid="xxxxxx" <111>
host=dynamic
mailbox=111@default
canreinvite=no
context=111
callgroup=1
pickupgroup=1

[121]
type=friend
username=121
secret=xxxxx
callerid="xxxxxxx" <121>
host=dynamic
mailbox=121@default
canreinvite=no
context=121
callgroup=1
pickupgroup=1

[131]
type=friend
username=131
secret=xxxxxx
callerid="xxxxxx" <131>
host=dynamic
mailbox=131@default
canreinvite=no
context=131
callgroup=1
pickupgroup=1

[101]
type=friend
username=101
secret=xxxxxxx
callerid="xxxxx" <101>
host=dynamic
mailbox=101@default
canreinvite=no
context=101
callgroup=1
pickupgroup=1
3

Does nobody have any idea what could be wrong here?

4

Hi there,

has this been resolved?

I am facing the same issue.
GXP-2000 which is losing registration after the first "re-"register message because Asterisk (1.4.0) says it’s a 481.
Only the first registration after I reboot the phone works.

Anyone?

Thanks for the support!

5

Hi,

I am facing similar problem. My CLI is bombared with following messages from all the phones. Some phones are behind NAT.

The phones are not restarted. I figured out this message will be displayed only when the phone that is not registered registers itself. So what it means is that phones are momentarily and periodically losing its registration. Can someone suggest why this is happening?

Thanks,
Harish

6

phones re register usually every 60 seconds so the server knows where they are, usually very important if they are remote. based on your log snippet, they are registering every 120 seconds. nothing unusual

7

Thanks for the reply. I understand that the phones periodically re-registers itself say every 120s. On the CLI the registration (with verbosity level 7) messages are NOT displayed when a re-registration message comes from an already registered phone.

So what is happening is somewhere near the end of 120s the PBX is losing the registration momentarily. I can confirm this running sip show peers periodically.

Do you think if for some reason the internet is slow and round trip delay is slightly larger, this could happen.

8

Stick a qualify-yes line in each phones block in your sip.conf file.

If any of the phones are behind a NAT, set nat=yes as well

Here is one of my entries:

;Matthew Kleinmann
[1155]
type=friend
secret=wouldentyouliketoknow
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=sip-phones
callerid=“Matthew Kleinmann” <1155>
mailbox=1155

On the phones on the outside, make sure stun is turned on and point the phones at a stun server. stunserver.org works. Make sure the nat setting for the phone is turned on as well. Both of those settings are per line on the gxp2000’s in the line configuration menu.