Asterisk Integration with callmanager

Asterisk Asterisk Support
#1

Hallo,

Am doing in integration between our cisco callmanager and an asterisk server.

My callmanager is version 7.1.5.30000-1.I have configured a sip trunk on callmanager,i also have route pattern that i have confgured on callmanager that points to this trunk.

Calls made from sip clints to callmanager are successful and without any problem.

However calls that are made from callmanager ip phones to any sip phone are unsuccessful with a busy tone.

I collected the trace files on callmanager and are as below:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.105.105.1:5060;branch=z9hG4bK24c1f5f7fe7;received=10.105.105.1
From: “Koiser Nicholas” sip:752298@10.105.105.1;tag=afedfdba-4104-41f2-89de-b2d0a4304ac9-21288001
To: sip:921055@10.104.104.104;tag=as709d1736
Call-ID: b950e780-df016ed4-18f-169690a@10.105.105.1
CSeq: 101 INVITE
Server: FPBX-2.8.1(1.6.2.11)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5093e39c"
Content-Length: 0

10.105.105.1-cisco callmanager
10.104.104.104-asterisk server.
752298-extension on callmanager
921055-extension on asterisk(92 is the access code)

What would cause the calls made from callmanager to sip phones to fail?

Please assist.

Rgds,
Cliff.

#2

Getting unauthorised is normal behaviour. The question is what happens (or what happened before).

The most common case is that the client submits the request without authentication, gets unauthorised and then re-submits with the specified type of authentication…

I can’t see if CCM already offered suitable authentication, but if it didn’t and it didn’t retry after the unauthorised, you have CCM configuration problem.

#3

Have you solved this problem?

I have a similar problem except reversed. I can call to a phone on asterisk from my call manager but I cannot call from asterisk to callmanager.

Maybe we can exchange ideas and arrive at a solution.

Thanks

Mitch

#4

Hi!!!

I have the same problem of cliffline: i call from asterisk to callmanager but not from callmanager to asterisk. This is my configuration:

Cisco Callmanager 4.1.3
SIP Trunk parameters:

[code]Device Name* = 'my device name’
Description = 'my device name’
Device Pool* = Default
Call Classification* = OnNet
Media Resource Group List = < None >
Location = < None >
AAR Group = < None >
Media Termination Point Required = checked

Destination Address* = 10.0.70.248 (ip asterisk)
Destination Address is an SRV = unchecked

Destination Port = 5064
Incoming Port* = 5064
Outgoing Transport Type* = UDP
Preferred Originating Codec* = 711ulaw

Significant Digits* = All
Connected Line ID Presentation* = Allowed
Connected Name Presentation* = Allowed
Calling Search Space = 'my CSS’
AAR Calling Search Space = < None >
Prefix DN = blank
Redirecting Number Delivery - Inbound = unchecked

Calling Party Selection* = Originator
Calling Line ID Presentation* = Allowed
Calling Name Presentation* = Allowed
Caller ID DN = blank
Caller Name = blank
Redirecting Number Delivery - Outbound = unchecked

MLPP Domain (e.g., “0000FF”) = blank
MLPP Indication Not available on this device
MLPP Preemption Not available on this device [/code]

Route pattern configuration:

[code]Route Pattern* = 'my route pattern’
Partition = < None >
Description = blank
Numbering Plan* = North American Numbering Plan
Route Filter = < None >
MLPP Precedence = Default
Gateway or Route List* = sip trunk to asterisk

Route this pattern = checked
Block this pattern = — Not Selected —
Call Classification* = OffNet
Allow Device Override = unchecked
Provide Outside Dial Tone = checked
Allow Overlap Sending = unchecked
Urgent Priority = unchecked
Require Forced Authorization Code = unchecked
Authorization Level = 0
Require Client Matter Code = unchecked

Use Calling Party’s External Phone Number Mask = unchecked
Calling Party Transform Mask = blank
Prefix Digits (Outgoing Calls) = blank
Calling Line ID Presentation = Default
Calling Name Presentation = Default

Connected Line ID Presentation = Default
Connected Name Presentation = Default

Discard Digits = < None >
Called Party Transform Mask = blank
Prefix Digits (Outgoing Calls) = blank

Carrier Identification Code = blank
Network Service Protocol = — Not Selected —
Network Service = — Not Selected —
Service Parameter Name = — Not Exist —
Service Parameter Value = blank[/code]

Asterisk 1.8
Trunk configuration:

[code]Trunk Name: CallManagerInOut
Outbound Caller ID: CallManagerInOut
CID Options: Allow Any CID

Outgoing Settings
Trunk Name: CallManagerOut
PEER Details:
type=friend
qualify=yes
nat=no
insecure=very
host=10.0.70.231
port=5064
fromdomain=10.0.70.231
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no

Incoming Settings
USER Context: CallManagerIn
USER Details:
type=friend
qualify=yes
nat=no
insecure=very
host=10.0.70.231
port=5064
fromdomain=10.0.70.231
dtmf=rfc2833
disallow=all
context=from-internal
canreinvite=no
allow=ulaw

Register String: blank[/code]

Inbound Routes

Any suggestion?