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@init-1002:1] NoOp("SIP/1007-0000000e",
"---------{EXTEN}@DigiVoice----------") in new stack
-- Executing [6049091588@init-1002:2] Dial("SIP/1007-0000000e",
"SIP/6049091588@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:6049091...@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>
Contact: <sip:6042881234@207.3.45.123:5060>
Call-ID: 10edc49d63h7572719ef88980172b787@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@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>
Call-ID: 10edc49d63h7572719ef88980172b787@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787@207.3.45.123:5060
CSeq: 102 INVITE
User-Agent: DV VOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6049091588@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@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>;tag=as254dc64b
Contact: <sip:6042881234@207.3.45.123:5060>
Call-ID: 10edc49d63h7572719ef88980172b787@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@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@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787@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@207.3.45.123>;tag=as124d4e17
To: <sip:6049091...@voip.digitalvoice.ca>;tag=as254dc64b
Call-ID: 10edc49d63h7572719ef88980172b787@207.3.45.123:5060
CSeq: 103 BYE
User-Agent: DV VOIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6049091588@209.205.85.162>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog
'10edc49d63h7572719ef88980172b787@207.3.45.123:5060' Method: INVITE
Really destroying SIP dialog
'5753977127869b982367f17c0f2a10fc@127.0.0.1' Method: REGISTER
--
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
