RE: [Asterisk-Users] Cisco 2620 as PRI gateway

Tim Reimers Thu, 09 Feb 2006 06:20:11 -0800

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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED] > 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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RE: [Asterisk-Users] Cisco 2620 as PRI gateway

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