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[cyril.scetbon at free.fr]

Hi guys,

I receiving call through a gateway without any problem but I can't
transfer the call. Asterisk is complaining about not being able to
translate a path and getting 403 error from gateway.

Here is my sip configuration :

[412345679]
context=accueil
host=192.168.19.10
username=412345679
type=peer
insecure=very

[sipout]
type=peer
host=192.168.19.10

in extension.conf I'm trying to transfer the call :

exten => _*,1,Dial(SIP/sipout/612345678)

but asterisk does not agree

Audio is at 192.168.19.10 port 19322
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.19.1:5060:
INVITE sip:612345678 at 192.168.19.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" <sip:412345678 at 192.168.19.10>;tag=as123c4a09
To: <sip:612345678 at 192.168.19.1>
Contact: <sip:412345678 at 192.168.19.10>
Call-ID: 11f2e3ee0d049f121749878a4252d806 at 192.168.19.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 20:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3385 3385 IN IP4 192.168.19.10
s=session
c=IN IP4 192.168.19.10
t=0 0
m=audio 19322 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
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 sipout/612345678
[Apr 22 22:35:23] WARNING[3431]: channel.c:3337
ast_channel_make_compatible: No path to translate from
SIP/sipout-081909d0(4) to SIP/412345679-081885e8(
www*CLI>
<--- SIP read from 192.168.19.1:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" <sip:412345678 at 192.168.19.10>;tag=as123c4a09
To: <sip:612345678 at 192.168.19.1>;tag=46E30A54-C9A
Date: Tue, 22 Apr 2008 20:35:20 GMT
Call-ID: 11f2e3ee0d049f121749878a4252d806 at 192.168.19.10
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 192.168.19.1:5060:
ACK sip:612345678 at 192.168.19.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.19.10:5060;branch=z9hG4bK2fa34bcb;rport
From: "412345678" <sip:412345678 at 192.168.19.10>;tag=as123c4a09
To: <sip:612345678 at 192.168.19.1>;tag=46E30A54-C9A
Contact: <sip:412345678 at 192.168.19.10>
Call-ID: 11f2e3ee0d049f121749878a4252d806 at 192.168.19.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Apr 22 22:35:23] WARNING[3392]: chan_sip.c:11995
handle_response_invite: Received response: "Forbidden" from '"412345678"
<sip:412345678 at 192.168.19.10>;tag=as123c4a09'
-- SIP/sipout-081909d0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Really destroying SIP dialog
'11f2e3ee0d049f121749878a4252d806 at 192.168.19.10' Method: INVITE
www*CLI>
<--- SIP read from 192.168.19.1:55654 --->
BYE sip:412345679 at 192.168.19.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.19.1:5060;branch=z9hG4bK1895912ED
From: <sip:412345678 at 192.168.19.1>;tag=46E29598-2618
To: <sip:412345679 at 192.168.19.10>;tag=as73c6a52d
Date: Tue, 22 Apr 2008 20:34:51 GMT
Call-ID: 6451B9F0-FE211DD-A59DADFE-87DC36C1 at 192.168.19.1
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 12
Timestamp: 1208896525
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.19.1 : 5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.19.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.19.1:5060;branch=z9hG4bK1895912ED;received=192.168.19.1
From: <sip:412345678 at 192.168.19.1>;tag=46E29598-2618
To: <sip:412345679 at 192.168.19.10>;tag=as73c6a52d
Call-ID: 6451B9F0-FE211DD-A59DADFE-87DC36C1 at 192.168.19.1
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:412345679 at 192.168.19.10>
Content-Length: 0


<------------>
Really destroying SIP dialog
'6451B9F0-FE211DD-A59DADFE-87DC36C1 at 192.168.19.1' Method: BYE
Executing last minute cleanups
== Destroying musiconhold processes


Regards
--
Cyril SCETBON