[Asterisk-Users] Cisco and Asterisk

[Ben Miller]

Well, I'm just leaving demo in for testing.  Once I get things working
I'll be changing all that to city names most likely.

I don't want the call to it the Cisco then redirect to the Asterisk
box.  If I hit extension 602 right now, it works fine.  What I'm trying
to do is dial out to another real phone number through the Cisco's
FXO ports (one of which is connected to an FXS.)

On Mon, Mar 14, 2005 at 02:02:08PM +0100, Tomasz Bukowski wrote:
> Hi!
> First of all , (apart from solving your problem) you really should get
> rid of the whole [demo] context from extensions.conf, and place your
> stuff in your own context (i.e. [local]) (just for convenience and
> "security"). Getting back to the problem - as I see it you want to dial
> out through Cisco gw by dialing 1XXX
> To do so you must send the whole number to the gateway, so the gateway
> could do something (anything) with it.
> Your extensions.conf should be more like:
> exten => _1XXX,1,Dial(SIP/${EXTEN}@voice-gw)
> Dialing 1602 on your system-phone will result with sending the number
> 1602 to the gateway, which will then (according to your current
> dial-peer configuration) strip the leading 1 and send 602 back to
> Asterisk to dial your laptop.
> Hope it helps
> Brgs
> Tomek
> 
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Ben Miller
> > Sent: Friday, March 11, 2005 12:40 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Cisco and Asterisk
> > 
> > Hey all,
> > 
> > I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can
> get
> > a bit of help here.
> > 
> > First I'll explain my setup, and then my problem.
> > 
> > Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2
> FXO
> > ports.  I have an analog phone line plugged into the first port
> > (voice-port 1/0/0).  I've got it setup so that calls coming into that
> > analog line are transferred to my Asterisk server via SIP.
> > 
> > In the second port on my FXO card, I have a phone cable plugged into a
> > phone-system phone (the kind you have in the office plugged into your
> > phone system, the extra port on it acts as an FXS so a normal phone
> can
> > be plugged into it and can dial out by hitting 9,9 and then a number).
> > 
> > Incoming calls come into my * box fine, and I can hit digits on the
> > phone and have different thing happen.  For example, I setup XLite on
> my
> > work laptop and I've got an extension setup to dial my laptop.  What
> I'm
> > trying to do, though, is setup an extension that will connect back to
> my
> > router and let me make an outgoing call on the second voice port.
> Every
> > time I try to do this, I get SIP errors in the * CLI:
> > 
> > Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back from
> > 206.222.200.46.
> > 
> > 206.222.200.46 is the IP of my router.  I'm pretty sure that I'm just
> > missing some config in my router, but I've been googling the past few
> > days and can't get anything that's helping.  Thus, I turn to you to
> help
> > me out, if possible.
> > 
> > I work for an ISP and what we eventually want to do is setup VoIP for
> > our broadband customers so they can do unlimited dialing to various
> > cities where we have routers, and we'll just through some voice ports
> > into those routers and get some lines hooked up.
> > 
> > Here is my relevant config:
> > 
> > sip.conf:
> > 
> > [general]
> > context=default
> > port=5060
> > bindaddr=0.0.0.0
> > srvlookup=yes
> > disallow=all
> > allow=ulaw
> > dtmfmode=inband
> > nat=never
> > promiscredir = yes      ; If yes, allows 302 or REDIR to non-local SIP
> > address
> > 
> > [voice-gw]			; This is what I've setup for my Cisco
> > 				; has the voice ports
> > context=demo
> > type=friend
> > host=206.222.200.46             ; IP address of Cisco gateway
> > dtmfmode=inband
> > disallow=all
> > allow=ulaw
> > nat=no
> > qualify=yes
> > 
> > [ben]				; my work laptop
> > context=demo
> > type=friend
> > username=ben
> > host=dynamic
> > disallow=all
> > allow=ulaw
> > 
> > 
> > extensions.conf:
> > 
> > [general]
> > static=yes
> > writeprotect=no
> > 
> > ; You can include other config files, use the #include command
> (without
> > the ';')
> > ; Note that this is different from the "include" command that includes
> > contexts within
> > ; other contexts. The #include command works in all asterisk
> > configuration files.
> > ;#include "filename.conf"
> > 
> > ; The "Globals" category contains global variables that can be
> referenced
> > ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for
> Environmental
> > variable
> > ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
> > ;
> > [globals]
> > CONSOLE=Console/dsp                             ; Console interface
> for demo
> > ;CONSOLE=Zap/1
> > ;CONSOLE=Phone/phone0
> > IAXINFO=guest                                   ; IAXtel
> username/password
> > ;IAXINFO=myuser:mypass
> > TRUNK=Zap/g2                                    ; Trunk interface
> > TRUNKMSD=1                                      ; MSD digits to strip
> > (usually 1 or 0)
> > ;TRUNK=IAX2/user:pass at provider
> > 
> > ;
> > ; Any category other than "General" and "Globals" represent
> > ; extension contexts, which are collections of extensions.
> > ;
> > ; Extension names may be numbers, letters, or combinations
> > ; thereof. If an extension name is prefixed by a '_'
> > ; character, it is interpreted as a pattern rather than a
> > ; literal.  In patterns, some characters have special meanings:
> > ;
> > ;   X - any digit from 0-9
> > ;   Z - any digit from 1-9
> > ;   N - any digit from 2-9
> > ;   [1235-9] - any digit in the brackets (in this example,
> 1,2,3,5,6,7,8,9)
> > ;   . - wildcard, matches anything remaining (e.g. _9011. matches
> > ;       anything starting with 9011 excluding 9011 itself)
> > ;
> > ; For example the extension _NXXXXXX would match normal 7 digit
> dialings,
> > ; while _1NXXNXXXXXX would represent an area code plus phone number
> > ; preceeded by a one.
> > ;
> > ; Each step of an extension is ordered by priority, which must
> > ; always start with 1 to be considered a valid extension.
> > ;
> > ; Contexts contain several lines, one for each step of each
> > ; extension, which can take one of two forms as listed below,
> > ; with the first form being preferred.  One may include another
> > ; context in the current one as well, optionally with a
> > ; date and time.  Included contexts are included in the order
> > ; they are listed.
> > ;
> > ;[context]
> > ;exten => someexten,priority,application(arg1,arg2,...)
> > ;exten => someexten,priority,application,arg1|arg2...
> > ;
> > ; Timing list for includes is
> > ;
> > ;   <time range>|<days of week>|<days of month>|<months>
> > ;
> > ;include => daytime|9:00-17:00|mon-fri|*|*
> > ;
> > ; ignorepat can be used to instruct drivers to not cancel dialtone
> upon
> > ; receipt of a particular pattern.  The most commonly used example is
> > ; of course '9' like this:
> > ;
> > ;ignorepat => 9
> > ;
> > ; so that dialtone remains even after dialing a 9.
> > ;
> > 
> > ;
> > ; Here are the entries you need to participate in the IAXTEL
> > ; call routing system.  Most IAXTEL numbers begin with 1-700, but
> > ; there are exceptions.  For more information, and to sign
> > ; up, please go to www.gnophone.com or www.iaxtel.com
> > ;
> > [iaxtel700]
> > exten =>
> > _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
> > 
> > ;
> > ; The SWITCH statement permits a server to share the dialplain with
> > ; another server. Use with care: Reciprocal switch statements are not
> > ; allowed (e.g. both A -> B and B -> A), and the switched server needs
> > ; to be on-line or else dialing can be severly delayed.
> > ;
> > [iaxprovider]
> > ;switch => IAX2/user:[key]@myserver/mycontext
> > 
> > [trunkint]
> > ;
> > ; International long distance through trunk
> > ;
> > exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9011.,2,Congestion
> > 
> > [trunkld]
> > ;
> > ; Long distance context accessed through trunk
> > ;
> > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91NXXNXXXXXX,2,Congestion
> > 
> > [trunklocal]
> > ;
> > ; Local seven-digit dialing accessed through trunk interface
> > ;
> > exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9NXXXXXX,2,Congestion
> > 
> > [trunktollfree]
> > ;
> > ; Long distance context accessed through trunk interface
> > ;
> > exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91800NXXXXXX,2,Congestion
> > exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91888NXXXXXX,2,Congestion
> > exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91877NXXXXXX,2,Congestion
> > exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91866NXXXXXX,2,Congestion
> > 
> > [international]
> > ;
> > ; Master context for international long distance
> > ;
> > ignorepat => 9
> > include => longdistance
> > include => trunkint
> > 
> > [longdistance]
> > ;
> > ; Master context for long distance
> > ;
> > ignorepat => 9
> > include => local
> > include => trunkld
> > 
> > [local]
> > ;
> > ; Master context for local, toll-free, and iaxtel calls only
> > ;
> > ignorepat => 9
> > include => default
> > include => parkedcalls
> > include => trunklocal
> > include => iaxtel700
> > include => trunktollfree
> > include => iaxprovider
> > ;
> > ; You can use an alternative switch type as well, to resolve
> > ; extensions that are not known here, for example with remote
> > ; IAX switching you transparently get access to the remote
> > ; Asterisk PBX
> > ;
> > ; switch => IAX2/user:password at bigserver/local
> > 
> > [macro-stdexten];
> > ;
> > ; Standard extension macro:
> > ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as
> well
> > ;   ${ARG2} - Device(s) to ring
> > ;
> > exten => s,1,Dial(${ARG2},20)                                   ; Ring
> > the interface, 20 seconds maximum
> > exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump
> > based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> > 
> > exten => s-NOANSWER,1,Voicemail(u${ARG1})               ; If
> > unavailable, send to voicemail w/ unavail announce
> > exten => s-NOANSWER,2,Goto(default,s,1)                 ; If they
> press
> > #, return to start
> > 
> > exten => s-BUSY,1,Voicemail(b${ARG1})                   ; If busy,
> send
> > to voicemail w/ busy announce
> > exten => s-BUSY,2,Goto(default,s,1)                             ; If
> > they press #, return to start
> > 
> > exten => _s-.,1,Goto(s-NOANSWER,1)                              ;
> Treat
> > anything else as no answer
> > 
> > exten => a,1,VoicemailMain(${ARG1})                             ; If
> > they press *, send the user into VoicemailMain
> > 
> > [demo]
> > ;
> > ; We start with what to do when a call first comes in.
> > ;
> > exten => s,1,Wait,1                     ; Wait a second, just for fun
> > exten => s,2,Answer                     ; Answer the line
> > exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5
> seconds
> > exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10
> seconds
> > exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory
> message
> > exten => s,6,BackGround(demo-instruct)  ; Play some instructions
> > 
> > exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
> > exten => 2,2,Goto(s,6)
> > 
> > exten => 3,1,SetLanguage(fr)            ; Set language to french
> > exten => 3,2,Goto(s,5)                  ; Start with the
> congratulations
> > 
> > exten => 1001,1,Goto(default,s,1)
> > ;
> > ; We also create an example user, 1234, who is on the console and has
> > ; voicemail, etc.
> > ;
> > exten => 201,1,Playback(transfer,skip)          ; "Please hold
> while..."
> > exten => 1234,1,Playback(transfer,skip)         ; "Please hold
> while..."
> > ;                                       ; (but skip if channel is not
> up)
> > exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
> > ;
> > exten => 1235,1,Voicemail(u1234)                ; Right to voicemail
> > ;
> > exten => 1236,1,Dial(Console/dsp)               ; Ring forever
> > exten => 1236,2,Voicemail(u1234)                ; Unless busy
> > 
> > ;
> > ; # for when they're done with the demo
> > ;
> > exten => #,1,Playback(demo-thanks)              ; "Thanks for trying
> the
> > demo"
> > exten => #,2,Hangup                     ; Hang them up.
> > 
> > ;
> > ; A timeout and "invalid extension rule"
> > ;
> > exten => t,1,Goto(#,1)                  ; If they take too long, give
> up
> > exten => i,1,Playback(invalid)          ; "That's not valid, try
> again"
> > 
> > ;
> > ; Create an extension, 500, for dialing the
> > ; Asterisk demo.
> > ;
> > exten => 500,1,Playback(demo-abouttotry); Let them know what's going
> on
> > exten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default)     ; Call
> > the Asterisk demo
> > exten => 500,3,Playback(demo-nogo)      ; Couldn't connect to the demo
> site
> > exten => 500,4,Goto(s,6)                ; Return to the start over
> message.
> > 
> > ;
> > ; Create an extension, 600, for evaulating echo latency.
> > ;
> > exten => 600,1,Playback(demo-echotest)  ; Let them know what's going
> on
> > exten => 600,2,Echo                     ; Do the echo test
> > exten => 600,3,Playback(demo-echodone)  ; Let them know it's over
> > exten => 600,4,Goto(s,6)                ; Start over
> > 
> > 
> > exten => 601,1,Dial(SIP/1 at voice-gw)
> > exten => 602,1,Dial(SIP/ben)
> > 
> > 
> > [default]
> > include => demo
> > 
> > Cisco 3640 config:
> > 
> > !
> > voice-port 1/0/0
> >   input gain 10
> >   output attenuation 11
> >   no comfort-noise
> >   connection plar 1001
> > !
> > voice-port 1/0/1
> >   input gain 10
> >   output attenuation 11
> >   no comfort-noise
> >   connection trunk 1
> > !
> > !
> > mgcp profile default
> > !
> > !
> > !
> > dial-peer cor custom
> > !
> > !
> > !
> > dial-peer voice 100 pots
> >   destination-pattern 9....
> >   port 1/0/0
> >   forward-digits 4
> > !
> > dial-peer voice 2 voip
> >   destination-pattern 1...
> >   session protocol sipv2
> >   session target ipv4:206.222.200.19:5060
> >   codec g711ulaw
> >   no vad
> > !
> > dial-peer voice 601 pots
> >   port 1/0/1
> > !
> > 
> > voice-gw#sh ver
> > Cisco Internetwork Operating System Software
> > IOS (tm) 3600 Software (C3640-P7-M), Version 12.2(15)T14, RELEASE
> > SOFTWARE (fc4)
> > Technical Support: http://www.cisco.com/techsupport
> > Copyright (c) 1986-2004 by cisco Systems, Inc.
> > Compiled Sat 28-Aug-04 10:54 by cmong
> > Image text-base: 0x60008950, data-base: 0x61802000
> > 
> > ROM: System Bootstrap, Version 11.1(19)AA, EARLY DEPLOYMENT RELEASE
> > SOFTWARE (fc1)
> > ROM: 3600 Software (C3640-P7-M), Version 12.2(15)T14, RELEASE SOFTWARE
> > (fc4)
> > 
> > voice-gw uptime is 1 day, 7 hours, 27 minutes
> > System returned to ROM by reload
> > System image file is "slot0:c3640-p7-mz.122-15.T14.bin"
> > 
> > cisco 3640 (R4700) processor (revision 0x00) with 126976K/4096K bytes
> of
> > memory.
> > Processor board ID 09301319
> > R4700 CPU at 100Mhz, Implementation 33, Rev 1.0
> > X.25 software, Version 3.0.0.
> > Bridging software.
> > 1 FastEthernet/IEEE 802.3 interface(s)
> > 2 Voice FXO interface(s)
> > DRAM configuration is 64 bits wide with parity disabled.
> > 125K bytes of non-volatile configuration memory.
> > 8192K bytes of processor board System flash (Read/Write)
> > 16384K bytes of processor board PCMCIA Slot0 flash (Read/Write)
> > 
> > %Error: No PCMCIA Slot1 flash chip information available
> > 
> > Configuration register is 0x2102
> > 
> > 
> > If anyone has any suggestions at all, they would be greatly
> appreciated.
> >   Thanks in advance.
> > 
> > -Ben Miller
> > ruiner at netslacking.net
> > 
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> 
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