[asterisk-dev] txfax

[orlando gianluca]

Hi all, 
 
I'm a problem on a sip channel 
 
WARNING[6486]: pbx.c:5173 ast_pbx_outgoing_app: SIP/app-0827ef60 already
has a call record??
 
on this scenario
 
SIP/2.0 180 Ringing
To: <sip:0243887630 at xxx.xxx.xxx.xxx:5061>;tag=e0ba0994
Content-Length: 0
CSeq: 102 INVITE
Call-ID: 1597f2b21a374f4f5e075de413c96d81 at 138.132.186.27
Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:5090;branch=z9hG4bK3b2747d3;rport=5090;received=xxx.xxx.
xxx.xxx
From: asterisk <sip:asterisk at xxx.xxx.xxx.xxx:5090>;tag=as645f75d2
 

<------------->
--- (7 headers 0 lines) ---
AS_Adm*CLI>
<--- SIP read from xxx.xxx.xxx.xxx:5061 --->
SIP/2.0 200 OK
To: <sip:0243887630 at xxx.xxx.xxx.xxx:5061>;tag=e0ba0994
Contact:
<sip:app-13ole83puevks at xxx.xxx.xxx.xxx:5061;transport=udp;wlsscid=377926
2777bf1e0>
Content-Length: 102
CSeq: 102 INVITE
Call-ID: 1597f2b21a374f4f5e075de413c96d81 at 138.132.186.27
<mailto:1597f2b21a374f4f5e075de413c96d81 at 138.132.186.27> 
Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:5090;branch=z9hG4bK3b2747d3;rport=5090;received=xxx.xxx.
xxx.xxx
From: asterisk <sip:asterisk at xxx.xxx.xxx.xxx:5090>;tag=as645f75d2
Content-Type: application/sdp
 
v=0
o=- 0 4657 IN IP4 172.24.30.35
s=IMSS
c=IN IP4 138.132.184.40
t=0 0
m=audio 19034 RTP/AVP 0
 
<------------->
--- (9 headers 6 lines) ---
Found RTP audio format 0
Peer audio RTP is at port xxx.xxx.xxx.xxx:19034
Capabilities: us - 0x3f1fff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|
jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port xxx.xxx.xxx.xxx (MEDIA GATEWAY):19034
list_route: hop:
<sip:app-13ole83puevks at xxx.xxx.xxx.xxx:5061;transport=udp;wlsscid=377926
2777bf1e0>
set_destination: Parsing
<sip:app-13ole83puevks at xxx.xxx.xxx.xxx:5061;transport=udp;wlsscid=377926
2777bf1e0> for address/port to send to
set_destination: set destination to xxx.xxx.xxx.xxx, port 5061
Transmitting (no NAT) to xxx.xxx.xxx.xxx:5061:
ACK
sip:app-13ole83puevks at xxx.xxx.xxx.xxx:5061;transport=udp;wlsscid=3779262
777bf1e0 SIP/2.0
Via: SIP/2.0/UDP 138.132.186.27:5090;branch=z9hG4bK69f6882d;rport
From: "asterisk" <sip:asterisk at xxx.xxx.xxx.xxx:5090>;tag=as645f75d2
To: <sip:0243887630 at xxx.xxx.xxx.xxx:5061>;tag=e0ba0994
Contact: <sip:asterisk at xxx.xxx.xxx.xxx:5090>
Call-ID: 1597f2b21a374f4f5e075de413c96d81 at xxx.xxx.xxx.xxx
<mailto:1597f2b21a374f4f5e075de413c96d81 at 138.132.186.27>  
CSeq: 102 ACK
ser-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
 
 
---
[Sep  3 18:23:45] WARNING[6486]: pbx.c:5173 ast_pbx_outgoing_app:
SIP/tsb-0827ef60 already has a call record??
AS_Adm*CLI>

I'm tring to send fax over ip throught txfax (spandsp).
 
Best Regards
Lella
 
 


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