[asterisk-users] SIP trunk trouble. Please help.

J Gao jgao at veecall.com
Mon Aug 15 17:54:06 CDT 2011 

Hello,

I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5. 
The problem is can't make any outbound/inbound. It always get "Number is 
not valid 701".

I tried to figure out the reason the call got dropped and couldn't find 
out the solution. I noticed that in the SIP debug there are two IP (from 
the provider) involved:
209.205.85.162
209.205.85.130

It seems my Asterisk sent INVITE to the first IP, but the provider want 
use 2nd one. How can I make it works? I never seen this thing before. 
(BTW, if I test this account on a Linsys ATA it works just fine!)

Here is my sip.conf setting and the debug out put.

Thanks for help!

Jian


-------------------------------
  sip.conf

[DigiVoice]
defaultuser=6042881234
fromuser=6042881234
authuser=6042881234
type=friend
secret=password
insecure=port,invite
canreinvite=yes
host=voip.digitalvoice.ca
context=from-trunk
qualify=yes
nat=yes


-----------------------------------
    -- Registered SIP '1007' at 24.20.99.133:9082
   == Using UDPTL CoS mark 5
   == Using SIP RTP CoS mark 5
     -- Executing [6049091588 at init-1002:1] NoOp("SIP/1007-0000000e", 
"---------{EXTEN}@DigiVoice----------") in new stack
     -- Executing [6049091588 at init-1002:2] Dial("SIP/1007-0000000e", 
"SIP/6049091588 at DigiVoice,60,T") in new stack
   == Using UDPTL CoS mark 5
   == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 209.205.85.130:5060:
INVITE sip:6049091588 at voip.digitalvoice.ca SIP/2.0
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK37896aa9;rport
Max-Forwards: 70
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>
Contact: <sip:6042881234 at 207.3.45.123:5060>
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Mon, 15 Aug 2011 22:22:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 2052747645 2052747645 IN IP4 207.3.45.123
s=Asterisk PBX 1.8.5.0
c=IN IP4 207.3.45.123
t=0 0
m=audio 18910 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
     -- Called SIP/6049091588 at DigiVoice

<--- SIP read from UDP:209.205.85.130:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK37896aa9;rport=5060
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 102 INVITE
Server: DigitalVoice.ca
Content-Length: 0

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

<--- SIP read from UDP:209.205.85.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK37896aa9;rport=5060
Record-Route: <sip:209.205.85.130;lr=on>
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 102 INVITE
User-Agent: DV VOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6049091588 at 209.205.85.162>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 6322 6322 IN IP4 209.205.85.162
s=session
c=IN IP4 209.205.85.162
t=0 0
m=audio 16818 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x4 
(ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 209.205.85.162:16818
list_route: hop: <sip:209.205.85.130;lr=on>
set_destination: Parsing <sip:209.205.85.130;lr=on> for address/port to 
send to
set_destination: set destination to 209.205.85.130:5060
Transmitting (NAT) to 209.205.85.130:5060:
ACK sip:6049091588 at 209.205.85.162 SIP/2.0
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK6999efa4;rport
Route: <sip:209.205.85.130;lr=on>
Max-Forwards: 70
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>;tag=as254dc64b
Contact: <sip:6042881234 at 207.3.45.123:5060>
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
     -- SIP/DigiVoice-0000000f answered SIP/1007-0000000e
Scheduling destruction of SIP dialog 
'10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060' in 23936 ms 
(Method: INVITE)
set_destination: Parsing <sip:209.205.85.130;lr=on> for address/port to 
send to
set_destination: set destination to 209.205.85.130:5060
Reliably Transmitting (NAT) to 209.205.85.130:5060:
BYE sip:6049091588 at 209.205.85.162 SIP/2.0
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK0ae463bf;rport
Route: <sip:209.205.85.130;lr=on>
Max-Forwards: 70
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 103 BYE
User-Agent: Asterisk
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
   == Spawn extension (init-1002, 6049091588, 2) exited non-zero on 
'SIP/1007-0000000e'

<--- SIP read from UDP:209.205.85.130:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.3.45.123:5060;branch=z9hG4bK0ae463bf;rport=5060
From: "User Name" <sip:6042881234 at 207.3.45.123>;tag=as124d4e17
To: <sip:6049091588 at voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060
CSeq: 103 BYE
User-Agent: DV VOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6049091588 at 209.205.85.162>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 
'10edc49d63h7572719ef88980172b787 at 207.3.45.123:5060' Method: INVITE
Really destroying SIP dialog 
'5753977127869b982367f17c0f2a10fc at 127.0.0.1' Method: REGISTER


--

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