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<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>Yeah-- sorry...</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>"</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>dial-peer voice 635099 voip<BR> description calls sent
to Asterisk<BR> preference 1<BR> destination-pattern
[635-9]..<BR> progress_ind setup enable 3<BR> session target
ipv4:10.10.1.28<BR> dtmf-relay h245-alphanumeric</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>"</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>I had been trying to do this with H.323 -- the Call Manager
uses H.323</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>There are some sip commands available in that dial-peer
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>ACS-GW(config-dial-peer)#voice-class sip ?<BR>
rel1xx Type of reliable provisional response
support<BR> transport Configure transport related
parameters<BR> url url type in
request line of outgoing INVITE</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>Not sure how I set those---</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>This:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>voice-class codec 1<BR> voice-class h323
1</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>is what is in there for the Call Manager h.323 dial-peer
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>That's obviously NOT what I want for the Asterisk-SIP
connection... </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>but I don't know what I need to do regarding the 'sip
url' or 'sip transport' or 'sip rel1xx' commands, if
anything...</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>How does one debug SIP activity? I see the debugs for
calls--- but I don't know the related debugs for actively
watching--</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>like you would 'debug isdn q931' -- that's the
outgoing side of the router--</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2>what would be the debug for a SIP call 'arriving' at the
router??</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=956355613-09022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Juan
Salas<BR><B>Sent:</B> Wednesday, February 08, 2006 2:17 PM<BR><B>To:</B>
'Asterisk Users Mailing List - Non-Commercial Discussion'<BR><B>Subject:</B> RE:
[Asterisk-Users] Cisco 2620 as PRI gateway<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=361581119-08022006><FONT face=Arial color=#0000ff size=2>Did
you create the dial-peers in the2651?</FONT></SPAN></DIV>
<DIV><SPAN class=361581119-08022006></SPAN> </DIV>
<BLOCKQUOTE>
<DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Mensaje original-----<BR><B>De:</B> Tim Reimers
[mailto:tim.reimers@asheville.k12.nc.us]<BR><B>Enviado el:</B> Wednesday,
February 08, 2006 1:41 PM<BR><B>Para:</B> Asterisk Users Mailing List -
Non-Commercial Discussion<BR><B>Asunto:</B> RE: [Asterisk-Users] Cisco 2620 as
PRI gateway<BR><BR></FONT></DIV>
<DIV dir=ltr align=left>sip-ua<BR> sip-server ipv4:<asterisk server
ip address><BR></DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2>OK -</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2>So I added those lines to my 2651 with the IP of my
asterisk box...</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2>How would I set this up as a SIP trunk in
Asterisk?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2>I have done this, in building a SIP trunk in
AMP.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006><FONT face=Arial
color=#0000ff size=2>host=10.12.1.252<BR>type=friend</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>I
don't know if/how to specify a username/password (as was the defaults in
there- the router didn't support having that configured..)</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>So I
picked friend..</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2>Then, in call routing, I picked my "Outbound
Routing"</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>the
"9_outside" route of "9|."</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>Set
that to use the new 'gw-rtr' I'd created...</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>no
go...</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2>Debug ISDN q931 doesn't show anything going to the
router...</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>In
Asterisk- </FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2>" -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2><snipped from below></FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff size=2>The
router doesn't show anything...</FONT></SPAN></DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=352053216-08022006><FONT face=Arial color=#0000ff
size=2></FONT> </DIV>
<DIV dir=ltr align=left><SPAN class=352053216-08022006></SPAN><FONT face=Arial
color=#0000ff size=2>t<SPAN class=352053216-08022006>he below shows up in
Asterisk -vvvv mode</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=352053216-08022006> -- Executing Macro("SIP/6351-cc18",
"dialout-trunk|3|2439499|") in new stack<BR> -- Executing
GotoIf("SIP/6351-cc18", "1?3:2)") in new stack<BR> -- Goto
(macro-dialout-trunk,s,3)<BR> -- Executing
Macro("SIP/6351-cc18", "user-callerid") in new stack<BR> --
Executing DBget("SIP/6351-cc18", "AMPUSER=DEVICE/6351/user") in new
stack<BR> -- DBget: varname=AMPUSER, family=DEVICE,
key=6351/user<BR> -- DBget: set variable AMPUSER to
6351<BR> -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack<BR> --
DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname<BR> -- DBget: set variable AMPUSERCIDNAME
to Tim-Zyxel<BR> -- Executing GotoIf("SIP/6351-cc18", "0?5")
in new stack<BR> -- Executing SetCallerID("SIP/6351-cc18",
"Tim-Zyxel <6351>") in new stack<BR> -- Executing
NoOp("SIP/6351-cc18", "Using CallerID "Tim-Zyxel" <6351>") in new
stack<BR> -- Executing Macro("SIP/6351-cc18",
"record-enable|6351|OUT") in new stack<BR> -- Executing
GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new stack<BR> --
Goto (macro-record-enable,s,4)<BR> -- Executing
AGI("SIP/6351-cc18", "recordingcheck|20060208-115748|1139417868.14") in new
stack<BR> -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck<BR>
recordingcheck|20060208-115748|1139417868.14: Outbound recording not
enabled<BR> -- AGI Script recordingcheck completed,
returning 0<BR> -- Executing NoOp("SIP/6351-cc18", "No
recording needed") in new stack<BR> -- Executing
Macro("SIP/6351-cc18", "outbound-callerid|3") in new
stack<BR> -- Executing GotoIf("SIP/6351-cc18", "1?3") in new
stack<BR> -- Goto
(macro-outbound-callerid,s,3)<BR> -- Executing
DBget("SIP/6351-cc18", "USEROUTCID=AMPUSER/6351/outboundcid") in new
stack<BR> -- DBget: varname=USEROUTCID, family=AMPUSER,
key=6351/outboundcid<BR> -- DBget: set variable USEROUTCID
to 6351<BR> -- Executing GotoIf("SIP/6351-cc18", "0?6") in
new stack<BR> -- Executing SetCallerID("SIP/6351-cc18",
"6351") in new stack<BR> -- Executing NoOp("SIP/6351-cc18",
"CallerID set to 6351") in new stack<BR> -- Executing
SetGroup("SIP/6351-cc18", "OUT_3") in new stack<BR> --
Executing CheckGroup("SIP/6351-cc18", "") in new stack<BR>
-- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499") in new
stack<BR> -- Executing SetVar("SIP/6351-cc18",
"DIAL_TRUNK=3") in new stack<BR> -- Executing
AGI("SIP/6351-cc18", "fixlocalprefix") in new stack<BR> --
Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix<BR>
fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf<BR> -- AGI Script
fixlocalprefix completed, returning 0<BR> -- Executing
SetVar("SIP/6351-cc18", "OUTNUM=2439499") in new stack<BR>
-- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new
stack<BR> -- Executing GotoIf("SIP/6351-cc18", "0?16") in
new stack<BR> -- Executing Dial("SIP/6351-cc18",
"SIP/acs-gw-rtr/2439499") in new stack<BR> -- Called
acs-gw-rtr/2439499<BR> -- SIP/acs-gw-rtr-b33f is
circuit-busy<BR> == Everyone is busy/congested at this time
(1:0/1/0)<BR> -- Executing Goto("SIP/6351-cc18",
"s-CONGESTION|1") in new stack<BR> -- Goto
(macro-dialout-trunk,s-CONGESTION,1)<BR> -- Executing
NoOp("SIP/6351-cc18", "Dial failed due to CONGESTION") in new
stack<BR> -- Executing Macro("SIP/6351-cc18", "outisbusy")
in new stack<BR> -- Executing Playback("SIP/6351-cc18",
"allison7/all-circuits-busy-now") in new stack<BR> -- Got
SIP response 481 "Call Leg/Transaction Does Not Exist" back from
10.12.1.252<BR> -- Playing 'allison7/all-circuits-busy-now'
(language 'en')<BR> -- Executing Playback("SIP/6351-cc18",
"allison7/pls-try-call-later") in new stack<BR> -- Playing
'allison7/pls-try-call-later' (language 'en')<BR> --
Executing Macro("SIP/6351-cc18", "hangupcall") in new
stack<BR> -- Executing ResetCDR("SIP/6351-cc18", "w") in new
stack<BR> -- Executing NoCDR("SIP/6351-cc18", "") in new
stack<BR> -- Executing Wait("SIP/6351-cc18", "5") in new
stack<BR> -- Executing Hangup("SIP/6351-cc18", "") in new
stack<BR> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/6351-cc18' in macro 'hangupcall'<BR> == Spawn extension
(macro-outisbusy, s, 3) exited non-zero on 'SIP/6351-cc18' in macro
'outisbusy'<BR> == Spawn extension (from-internal, 92439499, 2) exited
non-zero on 'SIP/6351-cc18'<BR> -- Executing
Macro("SIP/6351-cc18", "hangupcall") in new stack<BR> --
Executing ResetCDR("SIP/6351-cc18", "w") in new stack<BR> --
Executing NoCDR("SIP/6351-cc18", "") in new stack<BR> --
Executing Wait("SIP/6351-cc18", "5") in new stack<BR> --
Executing Hangup("SIP/6351-cc18", "") in new stack<BR> == Spawn
extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/6351-cc18' in macro
'hangupcall'<BR> == Spawn extension (from-internal, h, 1) exited
non-zero on 'SIP/6351-cc18'<BR>acsasterisk*CLI>
<BR></SPAN><BR></DIV></FONT></SPAN><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Gary
Richardson<BR><B>Sent:</B> Tuesday, February 07, 2006 9:09 PM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Cisco 2620 as PRI gateway<BR></FONT><BR></DIV>
<DIV></DIV>I have a 2811 working as a SIP gateway. My IOS version is
12.3(11)T5. <BR><BR>Looking through my config I
notice:<BR><BR>sip-ua<BR> sip-server ipv4:<asterisk server ip
address><BR><BR>Everything else in the config file is for our h323 call
manager gear. I can't remember if I needed to add the above line to make a sip
server run on the router. In order to place a call to the PSTN, I
Dial(SIP/9XXXXXXXXXX@<ip address of my 2811>) and everything works.
<BR><BR>As for how much of this applies to a 2600.. you'll have to
see.<BR><BR>
<DIV><SPAN class=gmail_quote>On 2/6/06, <B class=gmail_sendername>Schochet,
Wes</B> <<A
href="mailto:wes.schochet@selectcomfort.com">wes.schochet@selectcomfort.com
</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">I
just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I
make <BR>this thing into MGCP gateway or even a SIP gateway for
asterisk? Seems like<BR>it should bee useful for
something!<BR><BR>I'm perfectly happy to do my homework, but also don't feel
thee need to<BR>reinvent the wheel! So, links with relevant info
would be appreciated. If <BR>there is a config for a 2621 being
used as a gateway out there somewhere, I<BR>wouldn't be too proud to take a
look at that either! Asterisk configs would<BR>be great
too!<BR><BR>Thanks,<BR><BR>Wes<BR>_______________________________________________
<BR>--Bandwidth and Colocation provided by <A
href="http://Easynews.com">Easynews.com</A> --<BR><BR>Asterisk-Users mailing
list<BR>To UNSUBSCRIBE or update options visit:<BR> <A
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