[Asterisk-Users] SIP to PSTN gateway

[Rasta Man]

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 at yyy.yy.113.56;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.226.166:5060
From: "TOMS" <sip:TOMS at yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771 at yyy.yy.113.56;user=phone>
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a at xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: CSCO/4
Contact: <sip:TOMS at 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 at 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 at yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771 at yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a at xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:94732771 at 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 at yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771 at yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a at xxx.xxx.226.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:94732771 at 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 at yyy.yy.113.56:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.226.166:5060
From: "TOMS" <sip:TOMS at yyy.yy.113.56>;tag=0003e362fc0a001a407788de-1f443a6c
To: <sip:94732771 at yyy.yy.113.56;user=phone>;tag=as4bfbf62f
Call-ID: 0003e362-fc0a0042-22647abb-4243e54a at xxx.xxx.226.166
CSeq: 101 ACK
User-Agent: CSCO/4
Content-Length: 0


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


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