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

Sun Jan 16 05:00:28 MST 2005

> >>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="21
680b72deb8cb966868d671528fc431"
> 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="21
680b72deb8cb966868d671528fc431"
> Content-Length: 0

Kind of tough guessing at this without the config files that go with
it. Just a pure guess based on the modifyied debug stuff...
      -- Call accepted by 66.225.202.72 (format ULAW)
      -- Format for call is ULAW
      -- Hungup 'IAX2/NuFone/7'
implies NuFone isn't accepting ULAW across and IAX link, or the number
you are sending to NuFone is not valid.