[cisco-voip] Calls not forwarding over SIP trunk to Cue
Buchanan, James
jbuchanan at presidio.com
Sat Aug 13 23:48:47 EDT 2011
I assume 1.1.1.2 is your loopback. Have you tried the same interface to which your SIP is bound?
James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions
12 Cadillac Dr Ste 130 Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at presidio.com>
D: 615-866-5729 | F:615-866-5781 www.presidio.com<http://www.presidio.com/>
From: Jay Stants [mailto:jaystants at rogers.com]
Sent: Saturday, August 13, 2011 10:24 PM
To: Buchanan, James; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue
debug ccsip messages output
xx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Contact: <sip:5856784306 at 74.63.41.218>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Date: Sun, 14 Aug 2011 03:21:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 2851 2851 IN IP4 74.63.41.218
s=session
c=IN IP4 74.63.41.218
t=0 0
m=audio 14492 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Date: Sun, 14 Aug 2011 03:21:35 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Aug 14 03:21:35.836: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Date: Sun, 14 Aug 2011 03:21:35 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:5856786019 at 74.74.255.254:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Aug 14 03:21:47.324: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 CANCEL
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Content-Length: 0
Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>
Date: Sun, 14 Aug 2011 03:21:47 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 CANCEL
Content-Length: 0
Aug 14 03:21:47.356: //103434/59A64CAF97A5/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Date: Sun, 14 Aug 2011 03:21:47 GMT
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
Aug 14 03:21:47.400: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5856786019 at 74.74.255.254:56665 SIP/2.0
Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK5c9fe01a;rport
From: "+15856784306" <sip:5856784306 at 74.63.41.218>;tag=as29f9104d
To: <sip:5856786019 at 74.74.255.254:56665>;tag=64143B74-13D4
Contact: <sip:5856784306 at 74.63.41.218>
Call-ID: 13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218<mailto:13b1851f78dee7904fd538ef4265d5a7 at 74.63.41.218>
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "+15856784306" <sip:5856784306 at 74.63.41.218>;privacy=off;screen=no
Content-Length: 0
Regards,
Jay Stants
jaystants at rogers.com<mailto:jaystants at rogers.com>
From: "Buchanan, James" <jbuchanan at presidio.com>
To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Sent: Saturday, August 13, 2011 11:11:50 PM
Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
Can you send a debug ccsip messages?
James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions
12 Cadillac Dr Ste 130 Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at presidio.com>
D: 615-866-5729 | F:615-866-5781 www.presidio.com<http://www.presidio.com/>
From: Jay Stants [mailto:jaystants at rogers.com]
Sent: Saturday, August 13, 2011 10:02 PM
To: Buchanan, James; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calls not forwarding over SIP trunk to Cue
no dice - still just rings infinantly ..
Regards,
Jay Stants
jaystants at rogers.com<mailto:jaystants at rogers.com>
From: "Buchanan, James" <jbuchanan at presidio.com>
To: Jay Stants <jaystants at rogers.com>; "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Sent: Saturday, August 13, 2011 10:55:22 PM
Subject: RE: [cisco-voip] Calls not forwarding over SIP trunk to Cue
Try adding b2bua onto your voicemail dial peer.
James Buchanan| UC Technology Manager | Presidio South | Presidio Networked Solutions
12 Cadillac Dr Ste 130 Brentwood, TN 37027 | jbuchanan at presidio.com<mailto:jbuchanan at presidio.com>
D: 615-866-5729 | F:615-866-5781 www.presidio.com<http://www.presidio.com/>
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jay Stants
Sent: Saturday, August 13, 2011 9:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Calls not forwarding over SIP trunk to Cue
Having an issue where calls coming in from ITSP ring and never fwd to voicemail. Internally voicemail works. either by dialing the vm number or by pushing the messages button which essentially just speed dials the 4000 extension used for voicemail. Can someone take a look at the included config and maybe point out what i'm missing to allow for calls to be fwd'ed to Cue after the specified time.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
registrar server expires max 600 min 60
!
!
!
!
voice translation-rule 1
rule 1 /^9/ //
!
voice translation-rule 3
rule 1 /4.../ /5856786019/
!
!
voice translation-profile voip.ms
translate calling 3
translate called 1
dial-peer voice 1 voip
description **SIP Trunk to newyork.voip.ms**
translation-profile outgoing voip.ms
destination-pattern 9[2-9].[2-9].......
session protocol sipv2
session target dns:newyork.voip.ms
dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 voip
description **Incoming SIP Trunk - Voip.ms**
translation-profile incoming voip.ms
session protocol sipv2
session target ipv4:10.50.1.2
incoming called-number 5856786019
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 20 pots
destination-pattern 5.T
direct-inward-dial
no sip-register
!
dial-peer voice 3 voip
description ** Voicemail **
destination-pattern 4000
session protocol sipv2
session target ipv4:1.1.1.2
dtmf-relay sip-notify
codec g711ulaw
no vad
ephone-dn-template 1
call-forward busy 4000
call-forward noan 4000 timeout 18
ephone-dn 1 dual-line
number 4005 secondary 5856786019 no-reg
label 4005
name Jay Stants
ephone-dn-template 1
Regards,
Jay Stants
jaystants at rogers.com<mailto:jaystants at rogers.com>
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