Chanunavail error

Asterisk Asterisk Support
1

Hi All,

Got 2 Asterisk boxes. Can’t make a call from one of them Below is log of call:

[size=85]moree*CLI>

<— SIP read from UDP:1.1.1.100:5060 —>
INVITE sip:2000@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-5d495820a6db378d-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1000@1.1.1.100:5060
To: "2000"sip:2000@1.1.1.1
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 236

v=0
o=- 12997360570299945 1 IN IP4 1.1.1.100
s=CounterPath X-Lite 5.0.0
c=IN IP4 1.1.1.100
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 11 lines) —
Sending to 1.1.1.100:5060 (NAT)
Using INVITE request as basis request - NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
Found peer ‘1000’ for ‘1000’ from 1.1.1.100:5060

<— Reliably Transmitting (NAT) to 1.1.1.100:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-5d495820a6db378d-1—d8754z-;received=1.1.1.100;rport=5060
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
To: "2000"sip:2000@1.1.1.1;tag=as35775bba
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6ec0b831"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:1.1.1.100:5060 —>
ACK sip:2000@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-5d495820a6db378d-1—d8754z-;rport
Max-Forwards: 70
To: "2000"sip:2000@1.1.1.1;tag=as35775bba
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:1.1.1.100:5060 —>
INVITE sip:2000@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-6dfe75d8b84b11ff-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:1000@1.1.1.100:5060
To: "2000"sip:2000@1.1.1.1
From: "Moree Kwa Dell “sip:1000@1.1.1.1;tag=274e611d
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“6ec0b831”,uri="sip:2000@1.1.1.1”,response=“dc391cb681fefe2a36f7262782f73bae”,algorithm=MD5
Content-Length: 236

v=0
o=- 12997360570299945 1 IN IP4 1.1.1.100
s=CounterPath X-Lite 5.0.0
c=IN IP4 1.1.1.100
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 11 lines) —
Sending to 1.1.1.100:5060 (NAT)
Using INVITE request as basis request - NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
Found peer ‘1000’ for ‘1000’ from 1.1.1.100:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.100:5062
Looking for 2000 in internal (domain 1.1.1.1)
list_route: hop: sip:1000@1.1.1.100:5060

<— Transmitting (NAT) to 1.1.1.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-6dfe75d8b84b11ff-1—d8754z-;received=1.1.1.100;rport=5060
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
To: "2000"sip:2000@1.1.1.1
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:2000@1.1.1.1:5060
Content-Length: 0

<------------>
– Executing [2000@internal:1] Dial(“SIP/1000-0000000b”, “SIP/2000@astHP”) in new stack
== Using SIP RTP CoS mark 5
Really destroying SIP dialog ‘49584b1748baf3752a4443703626cce4@(null)’ Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/1000-0000000b’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (NAT) to 1.1.1.100:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-6dfe75d8b84b11ff-1—d8754z-;received=1.1.1.100;rport=5060
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
To: "2000"sip:2000@1.1.1.1;tag=as40d2bbba
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
Content-Length: 0

<------------>

<— SIP read from UDP:1.1.1.100:5060 —>
ACK sip:2000@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.100:5060;branch=z9hG4bK-d8754z-6dfe75d8b84b11ff-1—d8754z-;rport
Max-Forwards: 70
To: "2000"sip:2000@1.1.1.1;tag=as40d2bbba
From: "Moree Kwa Dell "sip:1000@1.1.1.1;tag=274e611d
Call-ID: NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘NDUyOTlkYWUzMGI1N2M1YWRjOWE1ZWZhMjUyMjEyMmI.’ Method: ACK

<— SIP read from UDP:1.1.1.100:5060 —>

<------------->

moree*CLI>[/size]

Can anyone assist in making this call work? Have been frustrated for the past one week and would really like to see it work.

2

One of:

astHP doesn’t exist in sip.conf;
astHP has a dynamic address but there is no current registration for it;
astHP has qualify enabled and is failing to respond (even negatively) to Asterisk’s OPTION packets used to probe the connectivity.

sip show peers, users and registry may reveal more.