Asterisk and Cisco Callmanager: Voicemail-Issues

kw
Joined: Thu 18 of Mar, 2010

Asterisk and Cisco Callmanager: Voicemail-Issues

Posted:Thu 18 of Mar, 2010 (13:11 UTC)
Hi,

I recently connected Cisco CCM 5.1 to an Asterisk 1.4 by following this instructions:

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration

That went very well. I can call from Snom SIP-Phones attached to Asterisk to Cisco-Phones attached to the Callmanager and vice versa.

After that I integrated the Asterisk-Voicemail to CCM like this:

http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Voicemail+Integration


That almost works. The Callmanager is redirecting the calls to the Asterisk-Voicemail and Asterisk is accecpting and recording the messages.

But as one can see in the text there is a problem, with calls from a SIP-Phone to CCM being redirected to the Voicemail.

[quote]
If a SIP telephone registered to the Asterisk machine acting as voicemail calls through to a Callmanager user and subsequently is sent to voicemail, the call will be dropped. This can be resolved by configuring the device in sip.conf as something like [sipexten] (eg: [sip7222]) rather than [exten] (eg: [7222]). 
[/quote]

So I changed the devices to [sipxxxxx]. After that calling CCM and being redirected to Voicemail works. But unfortunately the SIP-Phones are no longer reachable when configured like this.

If I try to call [sip29200] for example, Asterisk's debug output shows me:

[quote]
    -- Executing [29200@default:1] Dial("SIP/29100-09ec2d08", "SIP/29200|20") in new stack
[Mar 18 10:02:35] WARNING[27009]: chan_sip.c:2921 create_addr: No such host: 29200
Really destroying SIP dialog '522511cf2b76a83765219ba1731bee89@192.168.51.203' Method: INVITE
[Mar 18 10:02:35] WARNING[27009]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [29200@default:2] VoiceMail("SIP/29100-09ec2d08", "29200|u") in new stack
Audio is at 192.168.52.50 port 12604
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
[/quote]

I think Asterisk doesn't know which IP corresponds to the Snom SIP-Phone 29200.

Here is my user.conf:

[quote]
[trunk_1]
host = 192.168.52.5
username =
secret =
context = default
group = null
hasexten = no
hasiax = no
hassip = yes
registeriax = no
registersip = yes
trunk
insecure = no
disallow = all
allow = ulaw,alaw,g729,g726


;Sip Users
[sip29200]
username = 29200
fullname = Snom
type=friend
secret=12345
host=dynamic
disallow = all
allow = alaw,ulaw,g729
hassip = yes
canreinvite = yes


[29100]
type=friend
secret=12345
host=dynamic
hassip = yes
 [/quote]

And my extensions.conf:

[quote]
[general]

[globals]
CCM => SIP/trunk_1

[default]
;SIP-Accouunts
exten => 29100,1,Dial(SIP/29100)
exten => 29200,1,Dial(SIP/29200,20)
exten => 29200,n,VoiceMail(29200,u)

;Callmanager
exten => _5XXXX,1,Macro(dialout-callmanager,${EXTEN},${CCM})

;VoiceMail:
exten => 10000,1,GotoIf($["${CALLERID(rdnis)}" = ""]?NoRedirect:Redirect)
exten => 10000,n(NoRedirect),NoOp(${CALLERID(num)})
exten => 10000,n,VoiceMailMain(${CALLERID(num)},s)
exten => 10000,n,Hangup()
exten => 10000,n(Redirect),NoOp(${CALLERID(rdnis)})
exten => 10000,n,Voicemail(${CALLERID(rdnis)},u)
exten => 10000,n,Hangup()


[macro-dialout-callmanager]
exten => s,1,ChanIsAvail(${ARG2})
exten => s,2,Dial(${CUT(AVAILCHAN||1)}/${ARG1})
exten => s,3,Hangup
exten => s,102,Congestion

[/quote]


Can somebody point me in the direction of the solution? ;)