[Asterisk-Users] problem client sip (ser) to client sip (asterisk)

Walter Willis walterwn at gmail.com
Mon Aug 22 16:27:43 MST 2005 

i am configure ser:

if (method=="INVITE") {
if (uri=~"sip:1[1-9][0-9]+ at .*") {
 rewritehostport("192.168.0.183:5080");
};
};

an asterisk:

sip.conf
; config Xlite
[1234]
;context=sip
context=from-ser
type=friend
auth=md5
username=1234
secret=chooseapassword
;fromdomain=sorcier.com.pe      ; para prueba de ser -asterisk
callerid="First Extension" <1234>
host=dynamic
canreinvite=no
;disallow=all
;allow=gsm
;allow=ulaw
;allow=alaw


;and conexion the ser to asterisk
;
[ser-sip]
type=friend                ; permitimos llamadas entrantes y
salientes. Usar peer si solo es MWI
context=ser-asterisk               ; este es el contexto que usan las
llamadas entrantes
;host=sorcier.com.pe       ; Este es tu hostname o IP del servidor SER
host=192.168.0.183
fromdomain=sorcier.com.pe  ; este es tu  SER_DOMAIN (nombre de dominio del SER)
;insecure=very              ; Permite que las llamadas que viene del
SER pasen a Asterisk
insecure=yes
;mailbox=user at context      ; esto es para listar las cuentas de voicemail


;i am copy the voip-info



and the file the extensions.conf
; Configuracion al servidor ser, para llamada de ida
[from-ser]
exten => _X.,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,tr)

[ser-asterisk]
; Ignora el dígito 0
;ignorepat => 0
; conexion a un telefono sip

;exten => _0X.,1,Dial(SIP/${EXTEN:1},90,Ttr)
;exten => _0X.,1,Dial(SIP/${EXTEN},20,Ttr)
;exten => _0X.,1,Dial(SIP/1234,20,Ttr)
;exten => _0X.,1,Dial(SIP/1234 at ser-sip,20,Ttr)
exten => _0X.,1,Dial(SIP/${EXTEN})


i am probe diferents combinations, but no work


debug with asterisk and view itis:

Sip read:
INVITE sip:1234 at 192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
To: <sip:1234 at sorcier.com.pe>
Contact: <sip:rbolivar at 192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299

v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
To: <sip:1234 at sorcier.com.pe>;tag=as1ca211c4
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
CSeq: 3143 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0


 to 192.168.0.183:5060


Sip read:
INVITE sip:1234 at 192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
To: <sip:1234 at sorcier.com.pe>
Contact: <sip:rbolivar at 192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299

v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 13 lines
Ignoring this request


Sip read:
ACK sip:1234 at 192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.0
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
To: <sip:1234 at sorcier.com.pe>;tag=as1ca211c4
CSeq: 3143 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0


8 headers, 0 lines
Destroying call 'E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185'


Sip read:
INVITE sip:1234 at 192.168.0.183:5080 SIP/2.0
Record-Route: <sip:192.168.0.183;ftag=78607191;lr=on>
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;rport=5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
To: <sip:1234 at sorcier.com.pe>
Contact: <sip:rbolivar at 192.168.0.185:5060>
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
CSeq: 3143 INVITE
Max-Forwards: 16
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 299

v=0
o=rbolivar 23151399 23151750 IN IP4 192.168.0.185
s=X-Lite
c=IN IP4 192.168.0.185
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 13 lines
Using latest request as basis request
Sending to 192.168.0.183 : 5060 (non-NAT)
Found peer 'ser-sip'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.185:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e
(gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe
(gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 1234 in ser-asterisk
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
Via: SIP/2.0/UDP
192.168.0.185:5060;branch=z9hG4bKD068BA3A6FB6431AB2B9DC33F1E26857
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
To: <sip:1234 at sorcier.com.pe>;tag=as5a7c3a50
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
CSeq: 3143 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 192.168.0.183:5080>
Content-Length: 0


 to 192.168.0.183:5060


Sip read:
ACK sip:1234 at 192.168.0.183:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.183;branch=z9hG4bK54ee.3e0c30a3.1
From: rbolivar <sip:rbolivar at sorcier.com.pe>;tag=78607191
Call-ID: E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185
To: <sip:1234 at sorcier.com.pe>;tag=as5a7c3a50
CSeq: 3143 ACK
User-Agent: Sip EXpress router(0.10.99-dev14-tcp (i386/linux))
Content-Length: 0


8 headers, 0 lines
Destroying call 'E059F9B8-5EE6-4B5C-8D34-D5E0034BED91 at 192.168.0.185'


Sip read:


0 headers, 0 lines



the client sip (SER) call to client sip (asterisk) and return error 404 



WHAT IS THE PROBLEM??? OR HOW TO FIX THE ERROR??

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