Hello.
I'm trying to use Asterisk in combination with SER, to make the
routing proccess to my PSTN-Gateways. I made a simple test defining some
extension in my extension.conf, when i made a call my SER (SIP) Server
forward the call to Asterisk, this proccess is ok, but when the call is
answered i see an INVITE going out from Asterisk to my SER Server, this
invite is then passed to my endpoint. I think this is not normal (or maybe
yes?). I'am attaching my extension.conf and my sip.conf.
extension.conf
[general]
static=yes
writeprotect=no
[OUTGOING]
;exten => _00562.,1,Dial(SIP/${EXTEN}@my.ip.gw.1)
exten => _00569.,1,Dial(SIP/${EXTEN}@my.ip.gw.2)
-----------------------------------
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
;context = sip ;default Default for incoming calls
context=OUTGOING
autocreatepeer=yes
---------------------------
Here is part of the debug from Asterisk (sip debug ip (xxx.xxx.148.246))
We're at xxx.xxx.148.232 port 19848
Answering with capability 0x2(GSM)
Answering with capability 0x4(ULAW)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.148.246;branch=0
Via: SIP/2.0/UDP xxx.xxx.148.242:5060;branch=z9hG4bK2142c11da4177
Record-Route: <sip:005622408196@xxx.xxx.148.246;ftag=2142c11da4;lr=on>
From: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4
To: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7
Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242
CSeq: 177 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:005622408196@xxx.xxx.148.232>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13611 13611 IN IP4 xxx.xxx.148.232
s=session
c=IN IP4 xxx.xxx.148.232
t=0 0
m=audio 19848 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to xxx.xxx.148.246:5060
sipquest*CLI>
Sip read:
ACK sip:005622408196@xxx.xxx.148.232 SIP/2.0
Record-Route: <sip:005622408196@xxx.xxx.148.246;ftag=2142c11da4;lr=on>
Via: SIP/2.0/UDP xxx.xxx.148.246;branch=0
Via: SIP/2.0/UDP xxx.xxx.148.242:5060;branch=z9hG4bK2142c11da4177
From: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4
To: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7
Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242
CSeq: 177 ACK
Content-Length: 0
Max-Forwards: 69
10 headers, 0 lines
set_destination: Parsing
<sip:005622408196@xxx.xxx.148.246;ftag=2142c11da4;lr=on> for address/port
to
send to
set_destination: set destination to xxx.xxx.148.246, port 5060
We're at xxx.xxx.148.232 port 19848
Answering with capability 0x4(ULAW)
Answering with non-codec capability 0x1(G723)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:5555832351@xxx.xxx.148.242 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport
Route: <sip:5555832351@xxx.xxx.148.242>
From: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7
To: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4
Contact: <sip:005622408196@xxx.xxx.148.232>
Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 13611 13612 IN IP4 xxx.xxx.154.50
s=session
c=IN IP4 xxx.xxx.154.50
t=0 0
m=audio 24670 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
=silenceSupp:off - - - -
(no NAT) to xxx.xxx.148.246:5060
sipquest*CLI>
Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport=5060
From: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7
To: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4
Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 xxx.xxx.148.246:5060 "Noisy feedback tells: pid=26918
req_src_ip=xxx.xxx.148.232 req_src_port=5060
in_uri=sip:5555832351@xxx.xxx.148.242 out_uri=sip:5555832351@xxx.xxx.148.242
via_cnt==1"
9 headers, 0 lines
sipquest*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.148.232:5060;branch=z9hG4bK37068ebe;rport=5060
From: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7
To: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4
Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242
CSeq: 102 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:5555832351@xxx.xxx.148.242
Content-Type: application/sdp
Content-Length: 207
v=0
o=5555832351 1114766125 1114766125 IN IP4 xxx.xxx.148.242
s=AddPac Gateway SDP
c=IN IP4 xxx.xxx.148.242
t=1114766125 0
m=audio 23268 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
10 headers, 8 lines
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xxx.xxx.148.242:23268
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer -
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
list_route: hop: <sip:5555832351@xxx.xxx.148.242>
set_destination: Parsing <sip:5555832351@xxx.xxx.148.242> for address/port
to send to
set_destination: set destination to xxx.xxx.148.242, port 5060
Hope that someone can help me.
Thanks in advance.
Best Regards
Ricardo Martinez
