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Rasta Man

2003-Sep-03 12:58 UTC

[Asterisk-Users] SIP to PSTN gateway

Hello all,

taking examples from various pointers, I am attempting to put together an
outbound dialing example using SIP (Cisco 7960) with 2 X100P.  Everything seems
to be working without generating errors, but the problem is the phone hangs up
(102/Bye).   Any pointers/advice are much appreciated

Here is the section in extensions.conf:

extensions.conf

; From CISCO at work
;
exten => _9NXXXXXX,1,StripMSD,1
exten => _NXXXXXX,2,Dial(Zap/2/BYEXTENSION)


And the debug output:



Sip read:
INVITE sip:94732771@yyy.yy.113.56;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.226.166:5060
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: CSCO/4
Contact: <sip:TOMS@xxx.xxx.226.166:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 250
Accept: application/sdp

v=0
o=Cisco-SIPUA 20902 8155 IN IP4 xxx.xxx.226.166
s=SIP Call
c=IN IP4 xxx.xxx.226.166
t=0 0
m=audio 27632 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12 headers, 11 lines
Using latest request as basis request
Sending to xxx.xxx.226.166 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 2147483647, them - 268/0, combined - 268
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 94732771 in from-sip
list_route: hop: <sip:TOMS@xxx.xxx.226.166:5060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:94732771@yyy.yy.113.56>
Content-Length: 0


 to xxx.xxx.226.166:5060
We're at yyy.yy.113.56 port 11818
Answering with preferred capability 2147483647
Answering with non-codec capability 1
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:94732771@yyy.yy.113.56>
Content-Type: application/sdp
Content-Length: 167

v=0
o=root 25762 25762 IN IP4 yyy.yy.113.56
s=session
c=IN IP4 yyy.yy.113.56
t=0 0
m=audio 11818 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 to xxx.xxx.226.166:5060
Sip read:
ACK sip:94732771@yyy.yy.113.56:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.226.166:5060
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 101 ACK
User-Agent: CSCO/4
Content-Length: 0


8 headers, 0 lines
Sip read:
BYE sip:94732771@yyy.yy.113.56:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.226.166:5060
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 102 BYE
User-Agent: CSCO/4
Content-Length: 0


8 headers, 0 lines
Sending to xxx.xxx.226.166 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.226.166:5060;received=xxx.xxx.226.166
From: "TOMS"
<sip:TOMS@yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771@yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:94732771@yyy.yy.113.56>
Content-Length: 0


 to xxx.xxx.226.166:5060
Expression is '1'
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Fabian Müller

2004-Oct-17 10:33 UTC

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[Asterisk-Users] sip to pstn gateway

Hello,

here is what I am able to do:
- I am able to register a SIP Phone on an Asterisk server.
- I am able to call an extension on the remote Asterisk server with my SIP
  phone and hear the congratulation message.

Informations about the configuration:
- There is no phonecard (no digium card, no isdn card ...) in the
  Asterisk box but only a network interface card which connects the
  Asterisk server to a softswitch. This softswitch is the gateway to PSTN.
  Unfortunately I do not know anything about the softswitch. Is there
  something important that I should know about it?

Here is what I would like to have:
- I would like to be able to call a PSTN/ISDN phone with my SIP phone.
- That means when I take my SIP Phone and dial a telephone number that
  belongs to the PSTN Asterisk must route the SIP packages to the
  softswitch which in turn routes the call to the PSTN.
- When I dial another registered SIP phone Asterisk should connect the
  two sip phones so that they can speak to each other.

 -------------      ------------      --------------      --------
 | SIP phone | ---> | Asterisk | ---> | Softswitch | ---> | PSTN |
 -------------      ------------      --------------      --------
                        |
                        |      -------------
                        |----> | Sip phone |
                               -------------

I have no idea how to configure Asterisk to accomplish this task. I
started reading documents like ftp://ftp.isi.edu/in-notes/rfc3372.txt
and the sip RFC and Mailing List articles and so on but they did not
make me able to configure Asterisk in that way.

Does anybody know where I can find documents that describe how I can
do what I would like to have? Do I have to configure a SIP Proxy (SER
for example) on the Asterisk box or does it work without a SIP proxy
as well? Do I have to register the Asterisk box on the softswitch?
(Should this be possible at all?)

Thanks very much in advance for any kind of help.

Regards,

Fabian M?ller

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