[Asterisk-Users] Cisco 2620 as PRI gateway

Tim Reimers tim.reimers at asheville.k12.nc.us
Thu Feb 9 07:04:27 MST 2006 

Yeah-- sorry...
"
dial-peer voice 635099 voip
 description calls sent to Asterisk
 preference 1
 destination-pattern [635-9]..
 progress_ind setup enable 3
 session target ipv4:10.10.1.28
 dtmf-relay h245-alphanumeric
"
 
I had been trying to do this with H.323 -- the Call Manager uses H.323
 
There are some sip commands available in that dial-peer 
ACS-GW(config-dial-peer)#voice-class sip ?
  rel1xx     Type of reliable provisional response support
  transport  Configure transport related parameters
  url        url type in request line of outgoing INVITE
 
Not sure how I set those---
 
This:
voice-class codec 1
 voice-class h323 1
is what is in there for the Call Manager h.323 dial-peer 
 
That's obviously NOT what I want for the Asterisk-SIP connection... 
 
but I don't know what I need to do regarding the 'sip url' or 'sip
transport' or 'sip rel1xx' commands, if anything...
 
How does one debug SIP activity? I see the debugs for calls--- but I
don't know the related debugs for actively watching--
like you would 'debug isdn q931'  -- that's the outgoing side of the
router--
what would be the debug for a SIP call 'arriving' at the router??
 
 
 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Juan Salas
Sent: Wednesday, February 08, 2006 2:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cisco 2620 as PRI gateway


Did you create the dial-peers in the2651?
 

	-----Mensaje original-----
	De: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
	Enviado el: Wednesday, February 08, 2006 1:41 PM
	Para: Asterisk Users Mailing List - Non-Commercial Discussion
	Asunto: RE: [Asterisk-Users] Cisco 2620 as PRI gateway
	
	
	sip-ua
	  sip-server ipv4:<asterisk server ip address>
	
	OK -
	So I added those lines to my 2651 with the IP of my asterisk
box...
	 
	How would I set this up as a SIP trunk in Asterisk?
	I have done this, in building a SIP trunk in AMP.
	 
	host=10.12.1.252
	type=friend
	 
	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..)
	So I picked friend..
	 
	Then, in call routing, I picked my "Outbound Routing"
	the "9_outside" route of "9|."
	Set that to use the new 'gw-rtr' I'd created...
	 
	no go...
	 
	Debug ISDN q931 doesn't show anything going to the router...
	 
	In Asterisk- 
	"  -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
back from 10.12.1.252"
	<snipped from below>
	 
	The router doesn't show anything...
	 
	 
	 
	 
	the below shows up in Asterisk -vvvv mode
	 -- Executing Macro("SIP/6351-cc18", "dialout-trunk|3|2439499|")
in new stack
	    -- Executing GotoIf("SIP/6351-cc18", "1?3:2)") in new stack
	    -- Goto (macro-dialout-trunk,s,3)
	    -- Executing Macro("SIP/6351-cc18", "user-callerid") in new
stack
	    -- Executing DBget("SIP/6351-cc18",
"AMPUSER=DEVICE/6351/user") in new stack
	    -- DBget: varname=AMPUSER, family=DEVICE, key=6351/user
	    -- DBget: set variable AMPUSER to 6351
	    -- Executing DBget("SIP/6351-cc18",
"AMPUSERCIDNAME=AMPUSER/6351/cidname") in new stack
	    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER,
key=6351/cidname
	    -- DBget: set variable AMPUSERCIDNAME to Tim-Zyxel
	    -- Executing GotoIf("SIP/6351-cc18", "0?5") in new stack
	    -- Executing SetCallerID("SIP/6351-cc18", "Tim-Zyxel
<6351>") in new stack
	    -- Executing NoOp("SIP/6351-cc18", "Using CallerID
"Tim-Zyxel" <6351>") in new stack
	    -- Executing Macro("SIP/6351-cc18",
"record-enable|6351|OUT") in new stack
	    -- Executing GotoIf("SIP/6351-cc18", "0 > 0?2:4") in new
stack
	    -- Goto (macro-record-enable,s,4)
	    -- Executing AGI("SIP/6351-cc18",
"recordingcheck|20060208-115748|1139417868.14") in new stack
	    -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
	  recordingcheck|20060208-115748|1139417868.14: Outbound
recording not enabled
	    -- AGI Script recordingcheck completed, returning 0
	    -- Executing NoOp("SIP/6351-cc18", "No recording needed") in
new stack
	    -- Executing Macro("SIP/6351-cc18", "outbound-callerid|3")
in new stack
	    -- Executing GotoIf("SIP/6351-cc18", "1?3") in new stack
	    -- Goto (macro-outbound-callerid,s,3)
	    -- Executing DBget("SIP/6351-cc18",
"USEROUTCID=AMPUSER/6351/outboundcid") in new stack
	    -- DBget: varname=USEROUTCID, family=AMPUSER,
key=6351/outboundcid
	    -- DBget: set variable USEROUTCID to 6351
	    -- Executing GotoIf("SIP/6351-cc18", "0?6") in new stack
	    -- Executing SetCallerID("SIP/6351-cc18", "6351") in new
stack
	    -- Executing NoOp("SIP/6351-cc18", "CallerID set to 6351")
in new stack
	    -- Executing SetGroup("SIP/6351-cc18", "OUT_3") in new stack
	    -- Executing CheckGroup("SIP/6351-cc18", "") in new stack
	    -- Executing SetVar("SIP/6351-cc18", "DIAL_NUMBER=2439499")
in new stack
	    -- Executing SetVar("SIP/6351-cc18", "DIAL_TRUNK=3") in new
stack
	    -- Executing AGI("SIP/6351-cc18", "fixlocalprefix") in new
stack
	    -- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix
	  fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.conf
	    -- AGI Script fixlocalprefix completed, returning 0
	    -- Executing SetVar("SIP/6351-cc18", "OUTNUM=2439499") in
new stack
	    -- Executing Cut("SIP/6351-cc18", "custom=OUT_3|:|1") in new
stack
	    -- Executing GotoIf("SIP/6351-cc18", "0?16") in new stack
	    -- Executing Dial("SIP/6351-cc18", "SIP/acs-gw-rtr/2439499")
in new stack
	    -- Called acs-gw-rtr/2439499
	    -- SIP/acs-gw-rtr-b33f is circuit-busy
	  == Everyone is busy/congested at this time (1:0/1/0)
	    -- Executing Goto("SIP/6351-cc18", "s-CONGESTION|1") in new
stack
	    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
	    -- Executing NoOp("SIP/6351-cc18", "Dial failed due to
CONGESTION") in new stack
	    -- Executing Macro("SIP/6351-cc18", "outisbusy") in new
stack
	    -- Executing Playback("SIP/6351-cc18",
"allison7/all-circuits-busy-now") in new stack
	    -- Got SIP response 481 "Call Leg/Transaction Does Not
Exist" back from 10.12.1.252
	    -- Playing 'allison7/all-circuits-busy-now' (language 'en')
	    -- Executing Playback("SIP/6351-cc18",
"allison7/pls-try-call-later") in new stack
	    -- Playing 'allison7/pls-try-call-later' (language 'en')
	    -- Executing Macro("SIP/6351-cc18", "hangupcall") in new
stack
	    -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack
	    -- Executing NoCDR("SIP/6351-cc18", "") in new stack
	    -- Executing Wait("SIP/6351-cc18", "5") in new stack
	    -- Executing Hangup("SIP/6351-cc18", "") in new stack
	  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/6351-cc18' in macro 'hangupcall'
	  == Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'SIP/6351-cc18' in macro 'outisbusy'
	  == Spawn extension (from-internal, 92439499, 2) exited
non-zero on 'SIP/6351-cc18'
	    -- Executing Macro("SIP/6351-cc18", "hangupcall") in new
stack
	    -- Executing ResetCDR("SIP/6351-cc18", "w") in new stack
	    -- Executing NoCDR("SIP/6351-cc18", "") in new stack
	    -- Executing Wait("SIP/6351-cc18", "5") in new stack
	    -- Executing Hangup("SIP/6351-cc18", "") in new stack
	  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/6351-cc18' in macro 'hangupcall'
	  == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/6351-cc18'
	acsasterisk*CLI> 
	
	

________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary
Richardson
	Sent: Tuesday, February 07, 2006 9:09 PM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: Re: [Asterisk-Users] Cisco 2620 as PRI gateway
	
	
	I have a 2811 working as a SIP gateway. My IOS version is
12.3(11)T5. 
	
	Looking through my config I notice:
	
	sip-ua
	  sip-server ipv4:<asterisk server ip address>
	
	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. 
	
	As for how much of this applies to a 2600.. you'll have to see.
	
	
	On 2/6/06, Schochet, Wes <wes.schochet at selectcomfort.com >
wrote: 

		I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card
in it.  Can I make 
		this thing into MGCP gateway or even a SIP gateway for
asterisk?  Seems like
		it should bee useful for something!
		
		I'm perfectly happy to do my homework, but also don't
feel thee need to
		reinvent the wheel!  So, links with relevant info would
be appreciated.  If 
		there is a config for a 2621 being used as a gateway out
there somewhere, I
		wouldn't be too proud to take a look at that either!
Asterisk configs would
		be great too!
		
		Thanks,
		
		Wes
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