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:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 1.2.3.41:5060 From: sip:[EMAIL PROTECTED];tag=2980654425 To: <sip:[EMAIL PROTECTED];user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED];tag=2980654425
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5307f0b3
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="asterisk", nonce="5e9f7505"
Content-Length: 0
to 1.2.3.41:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'ata7001'
Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 1.2.3.41:5060 From: sip:[EMAIL PROTECTED];tag=2980654425 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5307f0b3 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Content-Length: 0
8 headers, 0 lines
Sip read:
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:[EMAIL PROTECTED];tag=2980654425
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Contact: <sip:[EMAIL PROTECTED]:5060;transport=udp>
User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest username="ata7001",realm="asterisk",nonce="5e9f7505",uri="sip:[EMAIL PROTECTED]",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:[EMAIL PROTECTED]:5060;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:[EMAIL PROTECTED];tag=2980654425
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as29aecdb3
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 1.2.3.41:5060
-- Executing Dial("SIP/ata7001-76d6", "IAX2/[EMAIL PROTECTED]/11235551212") in new stack
-- Called [EMAIL PROTECTED]/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:[EMAIL PROTECTED];tag=2980654425
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as29aecdb3
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 1.2.3.41:5060
From: sip:[EMAIL PROTECTED];tag=2980654425
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as29aecdb3
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest username="ata7001",realm="asterisk",nonce="5e9f7505",uri="sip:[EMAIL PROTECTED]",response="21680b72deb8cb966868d671528fc431"
Content-Length: 0
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