[Asterisk-Users] Cisco 2620 as PRI gateway

Wed Feb 8 10:41:27 MST 2006

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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