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Asterisk as Voicemail for Cisco Call manger

rbblue234
Level 1
Level 1

‎06-29-2013 06:16 AM - edited ‎03-19-2019 06:55 AM

Hi All,

I'm currently getting SIP/2.0 500 Internal Server Error when trying to turn on MWI.   Currently the cisco call manager is able to forward calls into asterisk for RNA, for CFA to VM. If the user hits the messages key, they are routed over to asterisk and asked for their password. Everything appears to be okay and working except for MWI.  When asterisk tries to dial the MWI number, CUCM is providing a SIP/2.0 500 Internal Server Error response.

The VoIP phone extension is 314.  The MWI turn on number is 880.  Asterisk IP is 192.168.10.51 and CUCM IP is 192.168.10.49.  Again, All call routing works..  As in if asterisk hands a PSTN call to CUCM, it will accept the call and route to the correct VoIP phone for answer. If no one answers that phone, the call is redirected back to asterisk for Voicemail.

sip_additional.conf

[cucmIN]

disallow=all

type=friend

context=sccp

host=192.168.10.49

allow=ulaw

allow=g729

nat=no

canreinvite=yes

qualify=yes

[cucmOut]

disallow=all

host=192.168.10.49

type=friend

allow=ulaw

allow=g729

nat=no

canreinvite=yes

qualify=yes

context=from-trunk-sip-cucmOut

extensions_customer.conf

[sccp]

include => ext-local

include => outbound-allroutes

include => app-vmmain

include => ext-featurecodes

include => ext-queues

include => Cisco-Voicemail

include => ciscovmail

[Cisco-Voicemail]

exten => 88808,1,GotoIf(${MAILBOX_EXISTS(${CALLERID(num)}@ciscovmail)} = "1"?400)

exten => 88808,2,Voicemail(${CALLERID(RDNIS)}@ciscovmail,u)

exten => 88808,3,Playback(vm-goodbye)

exten => 88808,4,Hangup

exten => 88808,400,VoicemailMain(${CALLERID(num)}@ciscovmail)

[ciscovmail]

exten => _280XXX,1,SetCallerID(${EXTEN:3})

exten => _280XXX,2,Dial(SIP/881@192.168.10.49)

exten => _280XXX,3,Answer

exten => _280XXX,4,Wait,1

exten => _280XXX,5,Hangup

exten => _281XXX,1,SetCallerID(${EXTEN:3})

exten => _281XXX,2,Dial(SIP/880@192.168.10.49)

exten => _281XXX,3,Answer

exten => _281XXX,4,Wait,1

exten => _281XXX,5,Hangup

Sip Debug

[2013-06-29 08:48:57] WARNING[31860]: pbx_spool.c:297 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/mwion.call.x1GJ102r: Operation not permitted

    -- Attempting call on SIP/880@cucmOut for 314@from-sip:2 (Retry 1)

  == Using SIP RTP TOS bits 184

  == Using SIP RTP CoS mark 5

Audio is at 13798

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (no NAT) to 192.168.10.49:5060:

INVITE sip:880@192.168.10.49 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5bf64f67

Max-Forwards: 70

From: "VoiceMail" <sip:314@192.168.10.51>;tag=as66a9cb2f

To: <sip:880@192.168.10.49>

Contact: <sip:314@192.168.10.51:5060>

Call-ID: 6ec8be502fc1fc1606c6fe491a635c37@192.168.10.51:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.10.1(1.8.21.0)

Date: Sat, 29 Jun 2013 12:48:57 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 237

v=0

o=root 109490670 109490670 IN IP4 192.168.10.51

s=Asterisk PBX 1.8.21.0

c=IN IP4 192.168.10.51

t=0 0

m=audio 13798 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

---

<--- SIP read from UDP:192.168.10.49:5060 --->

SIP/2.0 100 Trying

Date: Sat, 29 Jun 2013 12:48:57 GMT

From: "VoiceMail" <sip:314@192.168.10.51>;tag=as66a9cb2f

Allow-Events: presence

Content-Length: 0

To: <sip:880@192.168.10.49>

Call-ID: 6ec8be502fc1fc1606c6fe491a635c37@192.168.10.51:5060

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5bf64f67

CSeq: 102 INVITE

<------------->

--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.49:5060 --->

SIP/2.0 500 Internal Server Error

Date: Sat, 29 Jun 2013 12:48:57 GMT

From: "VoiceMail" <sip:314@192.168.10.51>;tag=as66a9cb2f

Allow-Events: presence

Content-Length: 0

To: <sip:880@192.168.10.49>;tag=ed20608f-032f-4323-ac4c-650fe15afd77-19794265

Call-ID: 6ec8be502fc1fc1606c6fe491a635c37@192.168.10.51:5060

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5bf64f67

CSeq: 102 INVITE

<------------->

--- (9 headers 0 lines) ---

    -- Got SIP response 500 "Internal Server Error" back from 192.168.10.49:5060

Transmitting (no NAT) to 192.168.10.49:5060:

ACK sip:880@192.168.10.49 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5bf64f67

Max-Forwards: 70

From: "VoiceMail" <sip:314@192.168.10.51>;tag=as66a9cb2f

To: <sip:880@192.168.10.49>;tag=ed20608f-032f-4323-ac4c-650fe15afd77-19794265

Contact: <sip:314@192.168.10.51:5060>

Call-ID: 6ec8be502fc1fc1606c6fe491a635c37@192.168.10.51:5060

CSeq: 102 ACK

User-Agent: FPBX-2.10.1(1.8.21.0)

Content-Length: 0

---

[2013-06-29 08:48:57] NOTICE[4291]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)

[2013-06-29 08:48:57] NOTICE[4291]: pbx_spool.c:375 attempt_thread: Queued call to SIP/880@cucmOut expired without completion after 0 attempts

Really destroying SIP dialog '6ec8be502fc1fc1606c6fe491a635c37@192.168.10.51:5060' Method: INVITE

    -- SEP000821969E93: unknown: 0, active call? no

    -- SEP000821969E93: Sending phone a token rejection (sccp.conf:fallback=false), ask again in '60' seconds

Labels:
0 Helpful
3 Replies 3

Tommer Catlin
VIP Alumni
VIP Alumni

‎07-01-2013 01:03 PM

CUCM can accept unsolicated MWI requests from outside vendors.  For example, Exchange 2010 and above, you can MWI lights sent over the SIP trunk with no codes.  Unless Asterisk does not support unsolicated MWI requests?

0 Helpful

rbblue234
Level 1
Level 1
In response to Tommer Catlin

‎07-01-2013 01:35 PM

HI!

Thanks for the respone.  Asterisk is the acting as the voicemail server here.  Asterisk is trying to light the MWI light on the phone which is attached to CUCM.   When asterisk makes the call to the CUCM MWI on light, we are getting a 500 internal error code.

I sinced changed the MWI on light to be 7880 (in an effort to move the MWI to something 100% out of the normal dial plan) and recieve the same results.  I enabled CUCM traces and pulled the following.    Any idea?  Again..  This is a sip call from asterisk @ 192.168.10.51 to CUCM 7.1.5 @ 192.168.10.49.  The caller ID was rewritten to 314 which is the extension of the phone, and call is being placed to 7880 which is MWI on extension.

|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:20000>

07/01/2013 13:06:20.984 CCM|//SIP/Stack/States/0xebe3248/0xebe3248 : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE)  to (STATE_ACTIVE, SUBSTATE_NONE)|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:40000>

07/01/2013 13:06:31.708 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 814 from 192.168.10.51:[5060]:

INVITE sip:7880@192.168.10.49 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5f3ccdf7

Max-Forwards: 70

From: "314" <314>;tag=as7b4adfb3

To: <7880>

Contact: <314>

Call-ID: 79b657c759f521d62ac821d84e449368@192.168.10.51:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.10.1(1.8.21.0)

Date: Mon, 01 Jul 2013 20:06:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 259

v=0

o=root 67849568 67849568 IN IP4 192.168.10.51

s=Asterisk PBX 1.8.21.0

c=IN IP4 192.168.10.51

t=0 0

m=audio 16744 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=ptime:20

a=sendrecv

|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:20000>

07/01/2013 13:06:31.708 CCM|//SIP/Stack/States/0xebe1610/0xebe1610 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:40000>

07/01/2013 13:06:31.709 CCM|//SIP/Stack/States/0xebe1610/0xebe1610 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, SUBSTATE_NONE)|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:40000>

07/01/2013 13:06:31.709 CCM|DbMobility: can't find remdest 314 in map|<:STANDALONECLUSTER><:192.168.10.49><:ERROR><:FFFFFF>

07/01/2013 13:06:31.710 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.10.51:[5060]:

SIP/2.0 100 Trying

Date: Mon, 01 Jul 2013 20:06:31 GMT

From: "314" <314>;tag=as7b4adfb3

Allow-Events: presence

Content-Length: 0

To: <7880>

Call-ID: 79b657c759f521d62ac821d84e449368@192.168.10.51:5060

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5f3ccdf7

CSeq: 102 INVITE

|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:20000>

07/01/2013 13:06:31.710 CCM|DbMobility: can't find remdest 314 in map|<:STANDALONECLUSTER><:192.168.10.49><:ERROR><:FFFFFF>

07/01/2013 13:06:31.711 CCM|Digit Analysis: getDaRes - voiceMailCallingSearchSpace=[]|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:0800>

07/01/2013 13:06:31.711 CCM|Digit analysis: match(pi="2", fqcn="", cn="314",plv="5", pss="RRG", TodFilteredPss="RRG", dd="7880",dac="0")|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:0800>

07/01/2013 13:06:31.711 CCM|Digit analysis: analysis results|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:0800>

07/01/2013 13:06:31.712 CCM||PretransformCallingPartyNumber=314

|CallingPartyNumber=314

|DialingPartition=RRG

|DialingPattern=7880

|FullyQualifiedCalledPartyNumber=7880

|DialingPatternRegularExpression=(7880)

|DialingWhere=

|PatternType=Enterprise

|PotentialMatches=NoPotentialMatchesExist

|DialingSdlProcessId=(0,0,0)

|PretransformDigitString=7880

|PretransformTagsList=SUBSCRIBER

|PretransformPositionalMatchList=7880

|CollectedDigits=7880

|UnconsumedDigits=

|TagsList=SUBSCRIBER

|PositionalMatchList=7880

|VoiceMailbox=

|VoiceMailCallingSearchSpace=

|VoiceMailPilotNumber=

|RouteBlockFlag=BlockThisPattern

|RouteBlockCause=0

|AlertingName=

|UnicodeDisplayName=

|DisplayNameLocale=1

|InterceptPartition=RRG

|InterceptPattern=7880

|InterceptWhere=

|InterceptSdlProcessId=(0,0,0)

|InterceptSsType=16777228

|InterceptSsKey=6966

|InterceptSsNotifyType=1

|OverlapSendingFlagEnabled=0

|WithTags=

|WithValues=

|CallingPartyNumberPi=NotSelected

|ConnectedPartyNumberPi=NotSelected

|CallingPartyNamePi=NotSelected

|ConnectedPartyNamePi=NotSelected

|CallManagerDeviceType=NoDeviceType

|PatternPrecedenceLevel=Routine

|CallableEndPointName=[dd554baf-c660-64c2-fe60-03cd09799f1e]

|PatternNodeId=[dd554baf-c660-64c2-fe60-03cd09799f1e]

|AARNeighborhood=[]

|AARDestinationMask=[]

|AARKeepCallHistory=true

|AARVoiceMailEnabled=false

|NetworkLocation=OnNet

|Calling Party Number Type=Cisco Unified CallManager

|Calling Party Numbering Plan=Cisco Unified CallManager

|Called Party Number Type=Cisco Unified CallManager

|Called Party Numbering Plan=Cisco Unified CallManager

|ProvideOutsideDialtone=false

|AllowDeviceOverride=false

|AlternateMatches= Information Not Available

|TranslationPatternDetails= Information Not Available

|ResourcePriorityNamespace=|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:0800>

07/01/2013 13:06:31.712 CCM|Cdcc::fireCfInterceptInd: precLvl=5|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:0800>

07/01/2013 13:06:31.713 CCM|ConnectionManager - wait_AuDisconnectRequest ERROR:NO ENTRY FOUND IN TABLE,CI(21025965,0),dcType=1,IFCreated(0,0),PID(0-0,0-0),IFHandling(0,0),MCNode(0,0)|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:ERROR><:0800>

07/01/2013 13:06:31.713 CCM|MatrixControl:updatePartyMediaCoordinatorNodeId: party1 videoCapable=0, party 2 videocapable=0|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:ALL><:FFFF>

07/01/2013 13:06:31.714 CCM|//SIP/Stack/States/0xebe1610/0xebe1610 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)|<:STANDALONECLUSTER><:192.168.10.49><:1><:192.168.10.51><:><:STATE transition=""><:40000>

07/01/2013 13:06:31.714 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.10.51:[5060]:

SIP/2.0 500 Internal Server Error

Date: Mon, 01 Jul 2013 20:06:31 GMT

From: "314" <314>;tag=as7b4adfb3

Allow-Events: presence

Content-Length: 0

To: <7880>;tag=ed20608f-032f-4323-ac4c-650fe15afd77-21025965

Call-ID: 79b657c759f521d62ac821d84e449368@192.168.10.51:5060

Via: SIP/2.0/UDP 192.168.10.51:5060;branch=z9hG4bK5f3ccdf7

CSeq: 102 INVITE

0 Helpful

CSCO10659222
Level 1
Level 1

‎07-30-2013 07:42 AM

Refer to this site for the MWI on/off  workaround  for an  CallManager http://shaun.net/2008/05/cisco-callmanager-3-with-asterisk-vm/

0 Helpful
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