[Asterisk-Users] Cisco FXO as PSTN gateway (updated request for assistance)

[Fran Boon]

Olle E. Johansson wrote:
>> I have been compiling information on this configuration onto the Wiki:
>> http://voip-info.org/wiki-Asterisk+cisco+FXO
>> I can call out to the PSTN just fine, but inbound calls all appear in 
>> my default [bogon-calls] context, not in [pstn-incoming]
> As I understand it, the Cisco is not registred with Asterisk as a peer.

It /appears/ to be:

redcusr01*CLI> sip show peers
Name/username    Host                 Mask             Port     Status
1001/1001        10.129.3.128    (D)  255.255.255.255  5060     Unmonitored
PSTN             10.129.3.254         255.255.255.255  5060     Unmonitored

> Could you please mail a SIP DEBUG output of an incoming INVITE from the
> Cisco to Asterisk?

redcusr01*CLI> SIP DEBUG
SIP Debugging Enabled
redcusr01*CLI>

Sip read:
INVITE sip:1001 at 10.129.3.239:5060 SIP/2.0
Via: SIP/2.0/UDP  10.129.3.254:5060
From: <sip:10.129.3.254>;tag=23D83A04-449
To: <sip:1001 at 10.129.3.239>
Date: Sun, 07 Mar 1993 23:02:53 GMT
Call-ID: B62068A2-1A6611CC-8066E2A8-2B58B3FE at 10.129.3.254
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 3055577250-442896844-2154029736-727233534
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:10.129.3.254>;party=calling;screen=no;privacy=off
Timestamp: 731545373
Contact: <sip:10.129.3.254:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 7818 6792 IN IP4 10.129.3.254
s=SIP Call
c=IN IP4 10.129.3.254
t=0 0
m=audio 19058 RTP/AVP 0 101
c=IN IP4 10.129.3.254
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

20 headers, 11 lines
Using latest request as basis request
Sending to 10.129.3.254 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format telephone-event
Capabilities: us - 4, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 1001 in bogon-calls
list_route: hop: <sip:10.129.3.254:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying


All guidance very much welcomed :)

Other options I'm considering to fix this are:
(1) Using SER to take the incoming calls from the Cisco
(2) Using H.323 to take the incoming calls from the Cisco

Commetns on these 2 options also welcomed :)

Best Wishes,
Fran.