[Asterisk-Users] ATA186: SIP/2.0 503 Service Unavailable

[Brian Capouch]

Rich Adamson wrote:
>>I have done my homework on this, I hope.
>>
>>I have a customer with an ATA186 who uses Nufone as his IAX provider. 
>>His network operations center in the Bahamas was destroyed by the 
>>hurricanes, and I'm helping him rebuild.
> 
> 
> I can help, but I think it might require being on site.
> 
> Just kidding; its 9 degrees above zero here in Nebraska. :(
> 
> Will need a little bit more then what you've provided to even guess
> at the issue.
> 
> Have you executed a 'sip debug' and looked at the detail?
> 

It took me a while to get it sanitized--it's at a customer site.  No NAT 
anywhere, 1.2.3.4 and 1.2.3.41 are the Asterisk box and ATA186, 
respectively.  81 is the "dial prefix" to choose the carrier.  Also, 
iaxy calls in the same context, using the same exact dialstring, go out 
just fine. . .*very perplexing.*

Thx.

B.

****  Snip ****

hostname-II*CLI> sip debug

Sip read:
INVITE sip:811235551212 at 1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>
Call-ID: 1938257462 at 1.2.3.41
CSeq: 1 INVITE
Contact: <sip:ata7001 at 1.2.3.41:5060;transport=udp>
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Expires: 300
Content-Length: 246
Content-Type: application/sdp

v=0
o=ata7001 6010 6010 IN IP4 1.2.3.41
s=ATA186 Call
c=IN IP4 1.2.3.41
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 11 lines
Using latest request as basis request
Sending to 1.2.3.41 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.41:16384
Found description format PCMU
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x4(ULAW), peer - 
audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>;tag=as5307f0b3
Call-ID: 1938257462 at 1.2.3.41
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:811235551212 at 1.2.3.4>
Proxy-Authenticate: Digest realm="asterisk", nonce="5e9f7505"
Content-Length: 0


  to 1.2.3.41:5060
Scheduling destruction of call '1938257462 at 1.2.3.41' in 15000 ms
Found user 'ata7001'

Sip read:
ACK sip:811235551212 at 1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>;tag=as5307f0b3
Call-ID: 1938257462 at 1.2.3.41
CSeq: 1 ACK
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Content-Length: 0


8 headers, 0 lines


Sip read:
INVITE sip:811235551212 at 1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>
Call-ID: 1938257462 at 1.2.3.41
CSeq: 2 INVITE
Contact: <sip:ata7001 at 1.2.3.41:5060;transport=udp>
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest 
username="ata7001",realm="asterisk",nonce="5e9f7505",uri="sip:811235551212 at 1.2.3.4",response="21680b72deb8cb966868d671528fc431"
Expires: 300> sip no debug
Content-Length: 246
Content-Type: application/sdp

v=0
o=ata7001 6016 6016 IN IP4 1.2.3.41
s=ATA186 Call
c=IN IP4 1.2.3.41
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12 headers, 11 lines
Using latest request as basis request
Sending to 1.2.3.41 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 1.2.3.41:16384
Found description format PCMU
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x4(ULAW), peer - 
audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Found user 'ata7001'
Looking for 811235551212 in home
list_route: hop: <sip:ata7001 at 1.2.3.41:5060;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>;tag=as29aecdb3
Call-ID: 1938257462 at 1.2.3.41
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:811235551212 at 1.2.3.4>
Content-Length: 0


  to 1.2.3.41:5060
     -- Executing Dial("SIP/ata7001-76d6", 
"IAX2/user at NuFone/11235551212") in new stack
     -- Called user at NuFone/11235551212
     -- Call accepted by 66.225.202.72 (format ULAW)
     -- Format for call is ULAW
     -- Hungup 'IAX2/NuFone/7'
   == No one is available to answer at this time
     -- Executing Congestion("SIP/ata7001-76d6", "") in new stack
Transmitting (no NAT):ebug
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>;tag=as29aecdb3
Call-ID: 1938257462 at 1.2.3.41
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:811235551212 at 1.2.3.4>
Content-Length: 0


  to 1.2.3.41:5060
   == Spawn extension (home, 811235551212, 2) exited non-zero on 
'SIP/ata7001-76d6'


Sip read:
ACK sip:811235551212 at 1.2.3.4;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:ata7001 at 1.2.3.4;tag=2980654425
To: <sip:811235551212 at 1.2.3.4;user=phone>;tag=as29aecdb3
Call-ID: 1938257462 at 1.2.3.41
CSeq: 2 ACK
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest 
username="ata7001",realm="asterisk",nonce="5e9f7505",uri="sip:811235551212 at 1.2.3.4",response="21680b72deb8cb966868d671528fc431"
Content-Length: 0