Help with Asterisk and Callmanager (SIP)


#1

Hello

I have been searching this formum and found a topic regarding using “SIP trunk”.
I have followed the instruction that says:
----------------------Instructions----------------------------------

  1. In ccmadmin select Device->Gateway->add->SIP Trunk->SIP
  2. Fill in Device Name, Device Pool, and IP address
  3. In * setup sip.conf with something like:
    callmgr (in brackets… wiki?)
    type=friend
    host=XXX.XXX.XXX.XXX
  4. In extension.conf:
    exten => ???,1,Dial(SIP/???@XXX.XXX.XXX.XXX,30,rT)
    substituting some 4 digit ext. for the ??? and IP for XXX

In my sip.conf I have written:

[callmgr]
type=friend
host=myccmip (IP OF MY CCM)

In my extensions.conf I have written:
exten => 5041,1,Dial(SIP/5041@172.30.1.121)

In my ccm 5041 is a device of type phone

In CCM 4.1 I have added:

In ccmadmin select Device->Trunk->SIP

I have put this values in my sip trunk:
Devicename: myIPofCCM
Devicepool: the same pool as my phones are in
Call classification: Use System default
Destination adress: Ip of my Asterisk
Outgoing trasport type: UDP

When I try to call from my Asterisk to a phone registrated in CCM I get this message in my Asterisk:

--------------------------------- Message------------------------------------------------
– Executing Goto(“SIP/(My phone nr calling from)-8580”, “(myextension)|5041|1”) in new stack
– Goto ((myextension),5041,1)
– Executing Dial(“SIP/(My phone nr calling from)-8580”, “SIP/5041@(IP of CCM)”) in new stack
– Called 5041@(IP of CCM)
– SIP/IP of CCM-bc0e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Got SIP response 481 “Call Leg/Transaction Does Not Exist” back from (IP of CCM)

Where
(My phone nr calling from)= my userid making the call
(myextension)= the extension call made from
(IP of CCM)= Ip of my CCM

I dont know what I am doing wrong and why I should set up [callmgr] in my sip.cfg when not using it?

Thanks for helping me!

/D_S


#2

It would appear that what you are trying to call on the CCM with Asterisk does not exist. Although I am having a hard time following all of your substituion for the real configuration.

If you could post the real configuration, without passwds and IP address, it might be easier to follow. Also, any logs available on the CCM side?


#3

Hello again here is the original message with no substitutions…

[callmgr]
type=friend
host=172.30.1.121 ;(IP OF MY CCM)

In my extensions.conf I have written:
exten => 5041,1,Dial(SIP/5041@172.30.1.121)

In my ccm 5041 is a device of type phone

In CCM 4.1 I have added:

In ccmadmin select Device->Trunk->SIP

I have put this values in my sip trunk in CCM:
Devicename: 172.30.1.121 (=IP of my CCM)
Devicepool: mypool (=the same pool as my phones are in)
Call classification: Use System default
Destination adress: 172.30.1.126 (=Ip of my Asterisk)
Outgoing trasport type: UDP

When I try to call from my Asterisk to a phone registrated in CCM I get this message in my Asterisk:

--------------------------------- Message------------------------------------------------
– Executing Goto(“SIP/1234567-8580”, “myextension|5041|1”) in new stack
– Goto (myextension,5041,1)
– Executing Dial(“SIP/1234567-8580”, “SIP/5041@172.30.1.121”) in new stack
– Called 5041@(172.30.1.121)
– SIP/IP of CCM-bc0e is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Got SIP response 481 “Call Leg/Transaction Does Not Exist” back from (172.30.1.121)

Thanks again!

/D_S


#4

From following the errors it appears everything is correct on the Asterisk side. Based on the fact you are getting that error message back from CCM it would appear that something is amiss there. Any logs available?


#5

Hello

Where do I find logs regarding generated calls in CCM, I have been searching in c:\Program Files\cisco\Trace but it seems to be alot of catalogues with no “informative names” likwise the naming of the files (?)

/D_S


#6

[quote=“d_s”]Hello

Where do I find logs regarding generated calls in CCM, I have been searching in c:\Program Files\cisco\Trace but it seems to be alot of catalogues with no “informative names” likwise the naming of the files (?)

/D_S[/quote]

Unfortunately, I do not know the CCM system. But from the error it appears your problem is somewhere in there.


#7

Hello

I got it to work! I changed the “call-route” (in extensions.cfg) to another registered phone in my CCM and it worked directly so it has to be something with my configuration of my first phone that isnt correct.

Thanks for your help anyway!

Regards/Mattias


#8

Hello again

Now it works to establish a call from asterisk-phone->CCM-phone but I can only speak from CCM-phone to Asterisk-phone. I cant hear anything in CCM phone when speaking in Asterisk phone

My Asteriskphone is configurated as below (in sip.cfg)

[myuser]
type=friend
context=mycontext
language=se
nat=yes
qualify=5000
canreinvite=no
host=dynamic
username=myuser
secret=8j5l56j85080
callerid = “Don Corleone” <5080>
dtmfmode=rfc2833
mailbox=5080@mycontext
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
disallow=all
;allow=g729
allow=alaw

In my CCM I have set in the SIP-trunk to use 711alaw as Prefered Originating Codec

…so what am I missing?

/D_S


#9

Is there a firewall somewhere in between, or the firewall setup on the Asterisk box? Sounds like an RTP stream is not getting from somewhere to somewhere.

Look at /etc/asterisk/rtp.conf to know which ports you are dealing with.


#10

Hello,

no there is no firewall between them and my rtp ports are set to (in rtp.cfg)
rtpstart=5000
rtpend=20000

as they are set to as Default;

/D_S