Home » Forums » Community Edition » Help

Trixbox with Cisco Call Manager

jon0881
Posts: 28
 Member Since:
2011-03-01
Submitted by jon0881 on Tue, 03/01/2011 - 7:24am.

Hi,

Im coming from a Cisco background and have only just started with trixbox and asterisk. Ive ordered a Sipura SPA3000 to have a play about in the lab with some PSTN connectivity.

Im fed up of lining Cisco's pockets for unnecessary changes. One example of which is voicemail. We cannot buy more licences for our Unity and to upgrade is going to cost us a fortune. My main goal is to eventually integrate with Exchange but for now i would like to have about 100 mailboxes on trixbox.

Ive setup the trunks and got the calls going between the two (CUCM 6.1 and trixbox). Ive read all the info on VOIP-INFO on the integration and have changed the voicemail profiles on the cisco side. The called number on the cisco side is 30105555. Ive tried creating the extension on the trixbox and loads of other things but given my experience im limited quite alot to the gui. I am not fussed about passing the callerID, just getting the prompt for username and password will be enough. After that, how would i add extensions just for voicemail?

Can anyone help?

Regards

Jon

‹ Message waiting indicator on Cisco 79x1 DynDNS Truck does not update ip address ›


jmd87
Posts: 37
 Member Since:
2011-02-23
Hi, Im not quite sure what
Tue, 03/01/2011 - 7:53am

Hi,

Im not quite sure what you mean? Im new to Trixbox aswell But have got Cisco phones working with Trixbox and also Voicemail so i maybe able to help a bit :)

Thanks alot
Joe



jon0881
Posts: 28
 Member Since:
2011-03-01
Hi Joe, OK, i have Cisco
Tue, 03/01/2011 - 8:09am

Hi Joe,

OK, i have Cisco phones working with a Cisco Call Manager. Voicemail is provided by Cisco Unity but this is at capacity and i am bringing in some greenfield sites. I would like to have the voicemail profile for the Cisco phones (still registered to the call manager) be sent to the trixbox and utilise voicemail only.

Regards

Jon



jon0881
Posts: 28
 Member Since:
2011-03-01
Hi,Can anyone help with
Mon, 03/07/2011 - 7:58am

Hi,

Can anyone help with this?

Ive moved a little further. If i think of it logically and simply the call manager is sending the call when a call manager extension is not avaliable. The problem is that in the SIP debugs on the * box all i can see is from:(mymobnumber)@(CUCMIP)
to:30105555@(asteriskIP)

So basically its just transfering the call andnot providing the origional DN. I reviewed the SIP trunk config on the call manager and saw that the 'Redirecting Diversion Header Delivery - Outbound' was not ticked so i did that.

Now i can see reference to my CUCM extension in the * SIP debugs. below is a copy of them. The call still comes in as 'the number you have dialed has not been regognised'. I have configured 30105085 as a custom extension on trixbox with voicemail enabled.

UDP://10.48.12.5:5060 --->
INVITE sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Mon, 07 Mar 2011 14:45:09 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
Allow-Events: presence
Supported: timer,replaces
Min-SE: 1800
Diversion: "Jon " ;reason=unconditional;privacy=off;screen=yes
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To:
Contact:
Expires: 180
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc816b2a9fd
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70


--- (19 headers 0 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Sending to 10.48.12.5 : 5060 (no NAT)
Using INVITE request as basis request - 7e878e00-d741ef75-13030-50c300a@10.48.12.5
No user '907**MYMOBNUMBER**' in SIP users list
Found peer 'CISCO' for '907**MYMOBNUMBER**' from 10.48.12.5:5060
Looking for 30105555 in from-sip-external (domain 10.48.90.31)
RDNIS for this call is is 30105085 (reason unconditional;privacy=off;screen=yes)
list_route: hop:
trixbox1*CLI>

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc816b2a9fd;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
To:
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Length: 0


-- Executing [30105555@from-sip-external:1] NoOp("SIP/CISCO-0000000c", "Received incoming SIP connection from unknown peer to 30105555") in new stack
-- Executing [30105555@from-sip-external:2] Set("SIP/CISCO-0000000c", "DID=30105555") in new stack
-- Executing [30105555@from-sip-external:3] Goto("SIP/CISCO-0000000c", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/CISCO-0000000c", "1?from-trunk,30105555,1") in new stack
-- Goto (from-trunk,30105555,1)
-- Executing [30105555@from-trunk:1] Set("SIP/CISCO-0000000c", "__FROM_DID=30105555") in new stack
-- Executing [30105555@from-trunk:2] NoOp("SIP/CISCO-0000000c", "Received an unknown call with DID set to 30105555") in new stack
-- Executing [30105555@from-trunk:3] Goto("SIP/CISCO-0000000c", "s,a2") in new stack
-- Goto (from-trunk,s,2)
-- Executing [s@from-trunk:2] Answer("SIP/CISCO-0000000c", "") in new stack
Audio is at 10.48.90.31 port 15722
Video is at 10.48.90.31 port 18042
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
trixbox1*CLI>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc816b2a9fd;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
To: ;tag=as1a54d05d
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Content-Length: 397

v=0
o=root 1559253914 1559253914 IN IP4 10.48.90.31
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 10.48.90.31
b=CT:384
t=0 0
m=audio 15722 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18042 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv


trixbox1*CLI>
UDP://10.48.12.5:5060 --->
ACK sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Mon, 07 Mar 2011 14:45:09 GMT
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
Allow-Events: presence
Content-Length: 230
To: ;tag=as1a54d05d
Content-Type: application/sdp
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc8240d12f6c
CSeq: 101 ACK
Max-Forwards: 70

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.48.12.5
s=SIP Call
c=IN IP4 10.48.12.5
t=0 0
m=audio 28192 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 34


--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 34
Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x4 (ulaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80004 (ulaw|h263)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.48.12.5:28192
Peer doesn't provide video
-- Executing [s@from-trunk:3] Wait("SIP/CISCO-0000000c", "2") in new stack
-- Executing [s@from-trunk:4] Playback("SIP/CISCO-0000000c", "ss-noservice") in new stack
-- Playing 'ss-noservice.gsm' (language 'en')
trixbox1*CLI>
UDP://10.48.12.5:5060 --->
BYE sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Mon, 07 Mar 2011 14:45:09 GMT
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To: ;tag=as1a54d05d
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc8321a55ca1
CSeq: 102 BYE
Max-Forwards: 70


--- (10 headers 0 lines) ---
Sending to 10.48.12.5 : 5060 (no NAT)


SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3cc8321a55ca1;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-27979945
To: ;tag=as1a54d05d
Call-ID: 7e878e00-d741ef75-13030-50c300a@10.48.12.5
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


== Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/CISCO-0000000c'
-- Executing [h@from-trunk:1] Hangup("SIP/CISCO-0000000c", "") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/CISCO-0000000c'
Really destroying SIP dialog '7e878e00-d741ef75-13030-50c300a@10.48.12.5' Method: BYE
Reliably Transmitting (NAT) to 10.48.90.162:5060:

Best wishes

Jon



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
Your call is arriving from
Mon, 03/07/2011 - 8:57am

Your call is arriving from "an unknown peer" in from-sip-external context. You need to fix your trunk and make sure it is in the from-internal context.

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
So ive made the below
Tue, 03/08/2011 - 3:31am

So ive made the below changed to sip_additional.conf and ive got some kind of loop. At first i thought it might be the reinvite message trying to establish direct RTP so i disabled it but that wasnt the problem.


[CISCO]
type=friend
context=from-internal
host=10.48.12.5
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=no
qualify=yes

Here is a link to the 140 page log file!

http://www.usaupload.net/d/6ziv7xu6889

Thanks for all the help



jon0881
Posts: 28
 Member Since:
2011-03-01
I have a gut feel something
Tue, 03/08/2011 - 3:55am

I have a gut feel something is missing from extensions_additional.conf


[from-trunk-sip-CISCO]
include => from-trunk-sip-CISCO-custom
exten => _.,1,Set(GROUP()=OUT_2)
exten => _.,n,Goto(from-trunk,${EXTEN},1)

; end of [from-trunk-sip-CISCO]

[macro-record-enable]
include => macro-record-enable-custom
exten => s,1,GotoIf($["${BLINDTRANSFER}" = ""]?check)
exten => s,n,ResetCDR(w)
exten => s,n,StopMixMonitor()
exten => s,n(check),AGI(recordingcheck,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)},${UNIQUEID})
exten => s,n,MacroExit()
exten => s,1+998(record),MixMonitor(${MIXMON_DIR}${CALLFILENAME}.${MIXMON_FORMAT},,${MIXMON_POST})

; end of [macro-record-enable]

[outbound-allroutes]
include => outbound-allroutes-custom
include => outrt-001-9_outside
include => outrt-002-ToCisco
exten => foo,1,Noop(bar)

; end of [outbound-allroutes]

[outrt-001-9_outside]
include => outrt-001-9_outside-custom
exten => _9.,1,Macro(user-callerid,SKIPTTL,)
exten => _9.,n,Set(_NODEST=)
exten => _9.,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _9.,n,Macro(dialout-trunk,3,${EXTEN},,)
exten => _9.,n,Macro(outisbusy,)

; end of [outrt-001-9_outside]

[outrt-002-ToCisco]
include => outrt-002-ToCisco-custom
exten => _3010XXXX,1,Macro(user-callerid,SKIPTTL,)
exten => _3010XXXX,n,Set(_NODEST=)
exten => _3010XXXX,n,Macro(record-enable,${AMPUSER},OUT,)
exten => _3010XXXX,n,Macro(dialout-trunk,2,${EXTEN},,)
exten => _3010XXXX,n,Macro(outisbusy,)

; end of [outrt-002-ToCisco]



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
WHy are you editing the conf
Tue, 03/08/2011 - 7:17am

WHy are you editing the conf files and not using the GUI? You are not supposed, nor is there a need to do what you are doing.

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
The only reason im doing it
Tue, 03/08/2011 - 7:40am

The only reason im doing it is on the basis of the reading on voip-info. Im just trying a few different things but ive always made a not as to what ive done so i can undo it if ive made the problem worse.



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
What you are doing is for
Tue, 03/08/2011 - 3:23pm

What you are doing is for generic Asterisk and not for FreePBX systems (like trixbox), not only do you not need to edit these files it will never work and it will be erased every time you reload PBX Settings.

I guess the warning at the top of the file is not stern enough :-(

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
I realise that editing the
Tue, 03/08/2011 - 3:42pm

I realise that editing the files manually would have been overwritten. Essentially i was copy and pasting config into it and testing. It was never meant to be permanent.

I will plod on by searching more along the lines of freepbx instead. I was wondering why there was a difference in the syntax for context=!



steve15
Posts: 43
 Member Since:
2006-12-17
Cisco Phones 7960
Tue, 03/08/2011 - 9:12pm

If your loading Trixbox, don't bother, it does not work! The cisco phones will not work!

--

Steve15



bravonoj
Posts: 213
 Member Since:
2007-11-20
@Steve15 - he is not trying
Tue, 03/08/2011 - 10:12pm

@Steve15 - he is not trying to get Cisco phones working with Trixbox - he wants CUCM to point to Trixbox for voicemail. I am interested in this; please let me know how and if you ever get it working.



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
No difference in syntax, you
Tue, 03/08/2011 - 10:17pm

No difference in syntax, you want the trunk in the from-internal context.

You don't need to do anything other than build a simple trunk

type=friend
insecure=very
host=x.x.x.x
disallow=all
allow=ulaw 
context=from-internal

This will allow the trixbox to receive calls from CUCM and send calls to CUCM

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
AHA!!!! now im getting
Wed, 03/09/2011 - 3:58am

AHA!!!! now im getting somewhere. I neglected to notice the outbound route i setup for testing the SIP trunk to and from the Call Manager. This dial pattern matched the destination that the VM pilot number and RDNIS was using. So removing that got rid of the looping.

Now im so close....Scott, im sure you are cheesed by now but i would really appreciate your help to finish this one off :)

I can still esablish a call from CUCM to trixbox and have checked i am using g711ulaw. Fine

I call in and i get ringing tone, and can see in the logs that trixbox is meant to be playing me some annunciator messages. then it kills the call. The voicemail pilot on the Cisco side is 30105555 and there is no reference to this in trixbox. Of course i can see the RDNIS is looking for 30105085 which is a custom extension on trixbox with voicemail enabled.

10.48.12.5 = call manager
10.48.90.31 = trixbox
30105085 = cisco extension number + custom extension on trixbox
30105555 = the voicemail pilot number in Cisco that has a route pattern to send the call to the SIP trunk.

trixbox1*CLI>
trixbox1*CLI>
UDP://10.48.12.5:5060 --->
INVITE sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Wed, 09 Mar 2011 10:18:29 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28013152
Allow-Events: presence
Supported: timer,replaces
Min-SE: 1800
Diversion: "Jon " ;reason=unconditional;privacy=off;screen=yes
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To:
Contact:
Expires: 180
Call-ID: 929cce00-d77153f5-13d20-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3da2562a2dc55
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70


--- (19 headers 0 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Sending to 10.48.12.5 : 5060 (no NAT)
Using INVITE request as basis request - 929cce00-d77153f5-13d20-50c300a@10.48.12.5
No user '9***PSTNNUMBER***' in SIP users list
Found peer 'CISCO' for '9***PSTNNUMBER***' from 10.48.12.5:5060
Looking for 30105555 in from-internal (domain 10.48.90.31)
RDNIS for this call is is 30105085 (reason unconditional;privacy=off;screen=yes)
list_route: hop:
trixbox1*CLI>

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3da2562a2dc55;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28013152
To:
Call-ID: 929cce00-d77153f5-13d20-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
ession-Expires: 1800;refresher=uas
Contact:
Content-Length: 0


-- Executing [30105555@from-internal:1] ResetCDR("SIP/CISCO-00000112", "") in new stack
-- Executing [30105555@from-internal:2] NoCDR("SIP/CISCO-00000112", "") in new stack
-- Executing [30105555@from-internal:3] Wait("SIP/CISCO-00000112", "1") in new stack
-- Executing [30105555@from-internal:4] Playback("SIP/CISCO-00000112", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
Audio is at 10.48.90.31 port 12546
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3da2562a2dc55;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28013152
To: ;tag=as2659a90e
Call-ID: 929cce00-d77153f5-13d20-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 826204356 826204356 IN IP4 10.48.90.31
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 10.48.90.31
t=0 0
m=audio 12546 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


-- Playing 'silence/1.gsm' (language 'en')
-- Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- Playing 'check-number-dial-again.ulaw' (language 'en')
-- Executing [30105555@from-internal:5] Wait("SIP/CISCO-00000112", "1") in new stack
-- Executing [30105555@from-internal:6] Congestion("SIP/CISCO-00000112", "20") in new stack
trixbox1*CLI>

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3da2562a2dc55;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28013152
To: ;tag=as2659a90e
Call-ID: 929cce00-d77153f5-13d20-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Content-Length: 0


== Spawn extension (from-internal, 30105555, 6) exited non-zero on 'SIP/CISCO-00000112'
-- Executing [h@from-internal:1] Macro("SIP/CISCO-00000112", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/CISCO-00000112", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/CISCO-00000112", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/CISCO-00000112", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/CISCO-00000112", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/CISCO-00000112' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/CISCO-00000112'
trixbox1*CLI>
UDP://10.48.12.5:5060 --->
ACK sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Wed, 09 Mar 2011 10:18:29 GMT
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28013152
Allow-Events: presence
Content-Length: 0
To: ;tag=as2659a90e
Call-ID: 929cce00-d77153f5-13d20-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3da2562a2dc55
CSeq: 101 ACK
Max-Forwards: 70


--- (10 headers 0 lines) ---
Really destroying SIP dialog '929cce00-d77153f5-13d20-50c300a@10.48.12.5' Method: ACK
Reliably Transmitting (no NAT) to 10.48.12.5:5060:
OPTIONS sip:10.48.12.5 SIP/2.0
Via: SIP/2.0/UDP 10.48.90.31:5060;branch=z9hG4bK58302523;rport
Max-Forwards: 70
From: "Unknown" ;tag=as63fd19a1
To:
Contact:
Call-ID: 6e562cc01c9b362703edcd6d18015bd2@10.48.90.31
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Wed, 09 Mar 2011 10:08:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
trixbox1*CLI>
UDP://10.48.12.5:5060 --->
SIP/2.0 200 OK
Date: Wed, 09 Mar 2011 10:18:55 GMT
Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS
From: "Unknown" ;tag=as63fd19a1
Content-Length: 0
To: ;tag=980713642
Call-ID: 6e562cc01c9b362703edcd6d18015bd2@10.48.90.31
Via: SIP/2.0/UDP 10.48.90.31:5060;branch=z9hG4bK58302523;rport
CSeq: 102 OPTIONS


--- (9 headers 0 lines) ---
Really destroying SIP dialog '6e562cc01c9b362703edcd6d18015bd2@10.48.90.31' Method: OPTIONS
trixbox1*CLI>



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
You can't run NAT with
Wed, 03/09/2011 - 8:32am

You can't run NAT with CUCM

Make sure you have the CUCM network listed in Asterisk localnet so it is excluded from NAT

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
Where should i make those
Wed, 03/09/2011 - 10:07am

I added the below into SIP_NAT.conf

[General]

localnet=10.48.12.0/255.255.254.0

and its still the same but there is more delay on getting the ringing tone



SkykingOH
Posts: 9678
 Member Since:
2007-12-17
You can't run NAT with CUCM
Wed, 03/09/2011 - 9:28am

You can't run NAT with CUCM

Make sure you have CUCM network list in Asterisk localnet to exclude from NAT.

--

Scott

aka "Skyking"



jon0881
Posts: 28
 Member Since:
2011-03-01
im getting these debugs all
Wed, 03/09/2011 - 10:00am

im getting these debugs all the time too

trixbox1*CLI>
Reliably Transmitting (no NAT) to 10.48.12.5:5060:
OPTIONS sip:10.48.12.5 SIP/2.0
Via: SIP/2.0/UDP 10.48.90.31:5060;branch=z9hG4bK4897b1b2;rport
Max-Forwards: 70
From: "Unknown" ;tag=as45926d76
To:
Contact:
Call-ID: 00f547212f31c70f3960ab0e67d6f85d@10.48.90.31
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Wed, 09 Mar 2011 16:43:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---
trixbox1*CLI>
UDP://10.48.12.5:5060 --->
SIP/2.0 200 OK
Date: Wed, 09 Mar 2011 16:54:19 GMT
Allow: INVITE, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, OPTIONS
From: "Unknown" ;tag=as45926d76
Content-Length: 0
To: ;tag=1961152116
Call-ID: 00f547212f31c70f3960ab0e67d6f85d@10.48.90.31
Via: SIP/2.0/UDP 10.48.90.31:5060;branch=z9hG4bK4897b1b2;rport
CSeq: 102 OPTIONS


--- (9 headers 0 lines) ---
Really destroying SIP dialog '00f547212f31c70f3960ab0e67d6f85d@10.48.90.31' Method: OPTIONS
trixbox1*CLI>
trixbox1*CLI>



jon0881
Posts: 28
 Member Since:
2011-03-01
Amost got it!!! Ive added a
Thu, 03/10/2011 - 11:45am

Amost got it!!!

Ive added a few lines in extensions_custom.conf. Now with 3 out of the 4 configs below i can hear comedien mail but it asks for the mailbox and password. Its not passing the mailbox ID


[from-internal-custom]

; TRY 1
;exten => 30105555,1,GotoIf($[${RDNIS}]?2:400)
;exten => 30105555,2,MailboxExists(${RDNIS}@default)
;exten => 30105555,3,Congestion
;exten => 30105555,103,Voicemail(su${RDNIS})
;exten => 30105555,104,Playback(vm-goodbye)
;exten => 30105555,105,Hangup
;exten => 30105555,400,VoicemailMain

; TRY 2
;exten => 30105555,1,GotoIf($"${CALLERID(rdnis)}" = ""?400)
;exten => 30105555,2,MailboxExists(${CALLERID(rdnis)}@default)
;exten => 30105555,3,Congestion
;exten => 30105555,103,103,Voicemail(su${CALLERID(rdnis)})
;exten => 30105555,104,Playback(vm-goodbye)
;exten => 30105555,105,Hangup
;exten => 30105555,400,VoicemailMain

; TRY 3

exten=30105555,1,GotoIf($["${RDNIS}"=""]?400)
exten=30105555,2,MailboxExists(${RDNIS}@default)
exten=30105555,3,Congestion
exten=30105555,103,Voicemail(su${RDNIS})
exten=30105555,104,Playback(vm-goodbye)
exten=30105555,105,Hangup
exten=30105555,400,VoicemailMain

;TRY 4

;exten => 30105555,1,GotoIf($["${CALLERID(rdnis)}" = ""]?400)
;exten => 30105555,2,MailboxExists(${CALLERID(rdnis)}@default)
;exten => 30105555,3,Congestion
;exten => 30105555,103,103,Voicemail(su${CALLERID(rdnis)})
;exten => 30105555,104,Playback(vm-goodbye)
;exten => 30105555,105,Hangup
;exten => 30105555,400,VoicemailMain

Here are the debugs


trixbox1*CLI>
UDP://10.48.12.5:5060 --->
INVITE sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Thu, 10 Mar 2011 18:42:29 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28050208
Allow-Events: presence
Supported: timer,replaces
Min-SE: 1800
Diversion: "Jon " ;reason=unconditional;privacy=off;screen=yes
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To:
Contact:
Expires: 180
Call-ID: 25785e00-d7911b95-147c2-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3e7772f5e97d4
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70


--- (19 headers 0 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Sending to 10.48.12.5 : 5060 (no NAT)
Using INVITE request as basis request - 25785e00-d7911b95-147c2-50c300a@10.48.12.5
No user '9***PSTN***' in SIP users list
Found peer 'CISCO' for '9***PSTN***' from 10.48.12.5:5060
Looking for 30105555 in from-internal (domain 10.48.90.31)
RDNIS for this call is is 30105085 (reason unconditional;privacy=off;screen=yes)
list_route: hop:
trixbox1*CLI>

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3e7772f5e97d4;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28050208
To:
Call-ID: 25785e00-d7911b95-147c2-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
ession-Expires: 1800;refresher=uas
Contact:
Content-Length: 0


-- Executing [30105555@from-internal:1] GotoIf("SIP/CISCO-00000008", "1?400") in new stack
-- Goto (from-internal,30105555,400)
-- Executing [30105555@from-internal:400] VoiceMailMain("SIP/CISCO-00000008", "") in new stack
Audio is at 10.48.90.31 port 18256
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3e7772f5e97d4;received=10.48.12.5
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28050208
To: ;tag=as58dd0581
Call-ID: 25785e00-d7911b95-147c2-50c300a@10.48.12.5
CSeq: 101 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Content-Length: 296

v=0
o=root 300770086 300770086 IN IP4 10.48.90.31
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 10.48.90.31
t=0 0
m=audio 18256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


trixbox1*CLI>
UDP://10.48.12.5:5060 --->
ACK sip:30105555@10.48.90.31:5060 SIP/2.0
Date: Thu, 10 Mar 2011 18:42:29 GMT
From: ;tag=8e8fe018-1ac7-4bf9-a0e4-3e2a61db5c5b-28050208
Allow-Events: presence
Content-Length: 208
To: ;tag=as58dd0581
Content-Type: application/sdp
Call-ID: 25785e00-d7911b95-147c2-50c300a@10.48.12.5
Via: SIP/2.0/UDP 10.48.12.5:5060;branch=z9hG4bK3e77817ddb0d7
CSeq: 101 ACK
Max-Forwards: 70

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 10.48.12.5
s=SIP Call
c=IN IP4 10.48.12.5
t=0 0
m=audio 25900 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.48.12.5:25900
-- Playing 'vm-login.gsm' (language 'en')



jon0881
Posts: 28
 Member Since:
2011-03-01
If anyone who is in a
Fri, 03/11/2011 - 9:28am

If anyone who is in a simmilar situation to me i can seriously reccomend the O'Riley book on asterisk. Im about 30% of the way through it but one particular chapter on contexts and the dial plan helped simplify the whole thing and help me make the conversion from Cisco to Asterisk in my mind.

Unfortunatly im still a little stuck on this problem, the statements for the VM pilot extn are a bit beyond the book. Using one of the setups above i manages to see the right RDNIS in the script but it still went to 30105555@from-internal. Ive also tried adding in include=> default to the above.

:(



obeliks
Posts: 878
 Member Since:
2010-03-14
You are not calling the
Fri, 03/11/2011 - 11:12am

You are not calling the Voicemail app correctly. I am guessing the reason is you followed the example from voip-info which is incorrect.
You can find the proper syntax by doing core show application VoiceMail



jon0881
Posts: 28
 Member Since:
2011-03-01
Thanks for that. The output
Fri, 03/11/2011 - 11:54am

Thanks for that. The output of that comand is below.


trixbox1*CLI>
-= Info about application 'VoiceMail' =-

[Synopsis]
Leave a Voicemail message

[Description]
VoiceMail(mailbox[@context][&mailbox[@context]][...][,options]): This
application allows the calling party to leave a message for the specified
list of mailboxes. When multiple mailboxes are specified, the greeting will
be taken from the first mailbox specified. Dialplan execution will stop if the
specified mailbox does not exist.
The Voicemail application will exit if any of the following DTMF digits are
received:
0 - Jump to the 'o' extension in the current dialplan context.
* - Jump to the 'a' extension in the current dialplan context.
This application will set the following channel variable upon completion:
VMSTATUS - This indicates the status of the execution of the VoiceMail
application. The possible values are:
SUCCESS | USEREXIT | FAILED

Options:
b - Play the 'busy' greeting to the calling party.
d([c]) - Accept digits for a new extension in context c, if played during
the greeting. Context defaults to the current context.
g(#) - Use the specified amount of gain when recording the voicemail
message. The units are whole-number decibels (dB).
Only works on supported technologies, which is DAHDI only.
s - Skip the playback of instructions for leaving a message to the
calling party.
u - Play the 'unavailable' greeting.

trixbox1*CLI>



jon0881
Posts: 28
 Member Since:
2011-03-01
ext-local has the below info
Fri, 03/11/2011 - 12:10pm

ext-local has the below info for my remote mailbox

exten => 30105085,1,Macro(exten-vm,30105085,30105085)
exten => 30105085,n,Goto(vmret,1)
exten => 30105085,hint,CUSTOM/30105085&Custom:DND30105085
exten => ${VM_PREFIX}30105085,1,Macro(vm,30105085,DIRECTDIAL,${IVR_RETVM})
exten => ${VM_PREFIX}30105085,n,Goto(vmret,1)
exten => vmb30105085,1,Macro(vm,30105085,BUSY,${IVR_RETVM})
exten => vmb30105085,n,Goto(vmret,1)
exten => vmu30105085,1,Macro(vm,30105085,NOANSWER,${IVR_RETVM})
exten => vmu30105085,n,Goto(vmret,1)
exten => vms30105085,1,Macro(vm,30105085,NOMESSAGE,${IVR_RETVM})
exten => vms30105085,n,Goto(vmret,1)



jon0881
Posts: 28
 Member Since:
2011-03-01
One of those try's in
Sat, 03/12/2011 - 1:21am

One of those try's in extensions_custom.conf managed to get the right rdnis number. Do i need a set step or something to change the context to reach the custom extensions?

Ive spend so much work and home time on this i think i can justify another few days then i think i wil have to pay for some help....



jon0881
Posts: 28
 Member Since:
2011-03-01
Persistance pays
Mon, 03/14/2011 - 9:23am

Persistance pays off!

Although i feel like im talking to myself now i feel like i should tell anyone that finds this in the future that i have got it working. Well ive got the voicemail RDNIS bit working, ive not even throught about voicemail retreval and MWI yet!

Here is the config for the extension that is the pilot number that Cisco is calling;


exten => 30105555,1,Macro(exten-vm,${CALLERID(rdnis)},${CALLERID(rdnis)})
exten => 30105555,n,Goto(vmret,1)
exten => 30105555,n,Goto(from-internal,${CALLERID(rdnis)},1)
exten => 30105555,hint,SIP/${CALLERID(rdnis)}&Custom:DND${CALLERID(rdnis)}
exten => ${VM_PREFIX}${CALLERID(rdnis)},1,Macro(vm,${CALLERID(rdnis)},DIRECTDIAL,${IVR_RETVM})
exten => ${VM_PREFIX}${CALLERID(rdnis)},n,Goto(vmret,1)
exten => vmb{CALLERID(rdnis)},,1,Macro(vm,${CALLERID(rdnis)},BUSY,${IVR_RETVM})
exten => vmb${CALLERID(rdnis)},n,Goto(vmret,1)
exten => vmu${CALLERID(rdnis)},1,Macro(vm,${CALLERID(rdnis)},NOANSWER,${IVR_RETVM})
exten => vmu${CALLERID(rdnis)},n,Goto(vmret,1)
exten => vms${CALLERID(rdnis)},1,Macro(vm,${CALLERID(rdnis)},NOMESSAGE,${IVR_RETVM})
exten => vms${CALLERID(rdnis)},n,Goto(vmret,1)



Andrew_Funk
Posts: 1
 Member Since:
2011-11-09
Trixbox with cisco call manager
Wed, 11/09/2011 - 9:39am

Hello Jon,
My name is Andrew, now I am working on the integration trixbox and cisco call manager so the phones registered in the cisco call manager can use the trixbox voicemail.
Please I need your help because I've made the trunk but when I call from a phone in CCM to a trixbox extention I answer it and the call drops, and when it redirect me to the trixbox voicemail I have just the busy tone.
I will appreciate if you can help me make it work.

Thank you in advance

Andrew



Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.