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