[Asterisk-Users] MGCP and SIP clients

[Mathias Röhl]

Hi folks

I seted up the asterisk with an active ISDN B1 AVM Card (german vendor)
and it works fine, various SIP clients (IP fon snom, xlite under 
MacOSX) and also incoming and outgoing connectins. Ok. No problem.
After that I configured a CP7940G with a MGCP IOS. It's connected to the
asterisk too via switch. No NAT deivce between. And of course, I can
call out to PSTN and also to one of the SIP clients. No problem.
But when I try to call the MGCP fon from a SIP client or from outside, I
get this error message.

----------------------<snip>--------------------


*CLI> Jun 20 15:30:20 NOTICE[25423]: chan_mgcp.c:1509 find_subchannel:
Gateway '172.16.103.20' (and thus its endpoint 'd001/1') does not exist
Jun 20 15:30:20 WARNING[25423]: chan_mgcp.c:3368 mgcp_request: Unable to
find MGCP endpoint 'd001/1 at 172.16.103.20'
Jun 20 15:30:20 NOTICE[25423]: app_dial.c:759 dial_exec: Unable to
create channel of type 'MGCP'
--------------------<snip>-------------------------

Otherwise, vice versa it looks good.

----------------------------<snip>---------------------

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.98.52:5060;branch=z9hG4bK0fe6a009
From: "1412" <sip:1412 at 172.16.98.52>;tag=as2ae905a1
To: <sip:1407 at 172.16.103.10:5060;line=xo91ds1x>;tag=ycfd6114eu
Call-ID: 7916fc9b5a6e3323751554a33603cbb9 at 172.16.98.52
CSeq: 102 INVITE
Contact: <sip:1407 at 172.16.103.10:5060;line=xo91ds1x>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
---------------------<snip>----------------

But I'm wondering why, the number "1412" is the MGCP fon. But it works. 
And as an additional info, why must I use with the Cisco CP7940 channel
d001 and not (like with an IP10s swissvoice...) aaln ? may be someone
can explain me.

If there are missing infos please let me know

best regards & thx in advance

mathias