Problem: Can not dial out from WGW to GSM:
Gateway registers successfully to freeswitch
Username: gsm1_amt
Call routing rule active:
Name: sip2gsm
from: sip gsm1_amt
to: gsm-1.1
GSM-Phone number: 0043 677 613 084 68
Dialed number: 0043 664 640 5042
What Im doing wrong in the SIP INVITE message?
Please have a look at the SIP trace below.
Thanks Willi
---- SIP messages received on WGW:
<--- SIP read from UDP:178.254.54.161:5070 --->
INVITE sip:
[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 178.254.54.161:5070;rport;branch=z9hG4bKr5ZSr5B1pp45e
Route: <sip:
[email protected]:57166>
Max-Forwards: 69
From: "gsm1_amt" <sip:
[email protected]>;tag=D0m8jpepSy83B
To: <sip:
[email protected]>
Call-ID: ac65d2b5-3aad-1233-4e80-00163e281932
CSeq: 72269610 INVITE
Contact: <sip:
[email protected]:5070>
User-Agent: FreeSWITCH-mod_sofia/1.4.8+git~20140812T200233Z~912c3652d2~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 298
X-FS-Support: update_display,send_info
Remote-Party-ID: "gsm1_amt" <sip:
[email protected]>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1425186336 1425186337 IN IP4 178.254.54.161
s=FreeSWITCH
c=IN IP4 178.254.54.161
t=0 0
m=audio 25012 RTP/AVP 8 9 0 3 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<
>
--- (18 headers 13 lines) ---
Sending to 178.254.54.161:5070 (NAT)
Using INVITE request as basis request - ac65d2b5-3aad-1233-4e80-00163e281932
peer=gsm1_amt-fs.ctww.de.
Found peer 'gsm1_amt-fs.ctww.de' for 'gsm1_amt-fs.ctww.de' from 178.254.54.161:5070
user_and_ip=gsm1_amt-fs.ctww.de, res=0, sipmethod=5.
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 13
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 178.254.54.161:25012
Looking for gsm1_amt in sipinbound (domain 192.168.0.198)
<--- Reliably Transmitting (no NAT) to 178.254.54.161:5070 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.254.54.161:5070;branch=z9hG4bKr5ZSr5B1pp45e;received=178.254.54.161;rport=5070
From: "gsm1_amt" <sip:
[email protected]>;tag=D0m8jpepSy83B
To: <sip:
[email protected]>;tag=as2d2115cd
Call-ID: ac65d2b5-3aad-1233-4e80-00163e281932
CSeq: 72269610 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<
>
Scheduling destruction of SIP dialog 'ac65d2b5-3aad-1233-4e80-00163e281932' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:178.254.54.161:5070 --->
ACK sip:
[email protected]:5070 SIP/2.0
Via: SIP/2.0/UDP 178.254.54.161:5070;rport;branch=z9hG4bKr5ZSr5B1pp45e
Route: <sip:
[email protected]:57166>
Max-Forwards: 69
From: "gsm1_amt" <sip:
[email protected]>;tag=D0m8jpepSy83B
To: <sip:
[email protected]>;tag=as2d2115cd
Call-ID: ac65d2b5-3aad-1233-4e80-00163e281932
CSeq: 72269610 ACK
Content-Length: 0
<
>
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'ac65d2b5-3aad-1233-4e80-00163e281932' Method: ACK
WGW1002G
Model: VS-GGU-E2M0400