So to communicate successfully with CUE through either direct calls
or forwarded calls, you'll have to use either SIP with G711 codec or
H323 with either G711 or G729 codec for the inbound call leg from
the WAN?
Wouldn't the local transcoder transcode the call legs between the
G729 SIP inbound from the WAN and the G711 SIP outbound to CUE?
JD
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Date: Thu, 21 Feb 2008 18:04:26 -0800
Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Problem
I see that you are using SIP inbound using g729 and using SIP
outbound using g711u. This scenario is not supported.
You have to use g711 on both legs or change the ipipgw -> cme leg of
the call to use h323.
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-
On-Demand and Audio Certification Training Tools for the Cisco CCIE
R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
From: Devildoc [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 21, 2008 4:40 PM
To: [EMAIL PROTECTED]; 'Jose Linero Welcker'; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
No, i already checked that and it was there along with transfer-
pattern. Here is my telephony service configuration.
telephony-service
load 7910 P00403020214
load 7960-7940 P00307020400
max-ephones 5
max-dn 15
ip source-address 172.25.102.1 port 2000
timeouts interdigit 5
system message BR2-Site
sdspfarm units 1
sdspfarm transcode sessions 4
sdspfarm tag 1 XCODER2
create cnf-files version-stamp Jan 01 2002 00:00:00
voicemail 3500
max-conferences 8 gain -6
call-forward pattern .T
transfer-system full-consult
transfer-pattern .T
Do you have any other idea? I am stumped at this point. Is there
any other debug that i can get to find out where the problems lie?
Thank you for any suggestion.
JD
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Date: Thu, 21 Feb 2008 15:43:40 -0800
If your direct call to CUE worked from CCM then the transcoder must
be working correctly and the allow-connections command must be
configured correctly.
I am guessing you are missing "call-forward pattern .T" inside
telephony-service.
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-
On-Demand and Audio Certification Training Tools for the Cisco CCIE
R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
From: Devildoc [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 21, 2008 1:39 PM
To: [EMAIL PROTECTED]; 'Jose Linero Welcker'; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Vik,
Sorry to revisit this post again. I don't know if this issue has
been resolved, but i ran into the same problem that Jose ran into
with forwarding calls to CUE from IPIPGW.
I have configured IPIPGW and tested calls to CUE before without any
problem. This is the first time i ran into this problem with POD
25. I have never used POD 25 before so i don't know if it's the
hardware issue with the DSP resources or my configuration.
When the router booted up, i could see the MTP device on the local
router registered successfully. However, after a few minutes, the
MTP device unregistered abnormally. I had to go into the dspfarm
profile to shut it down and bring it back up online again. After
that, it seemed to stay up but the calls to CUE never worked.
When i called to BR2 at 3003 from HQ 1003 and got forwarded to CUE
due to noan, I just got a busy signal. However, when i called CUE
pilot number directly, i got connected to CUE but DTMF didn not work.
Below is the debug for the direct call to CUE along with my
dialpeers configuration. Can you spot any misconfiguration?
Thanks for you help.
JD
dial-peer voice 1200 voip
description <<Outbound To IPIPGW for G729 H323 Calls>>
destination-pattern [12]...$
session target ipv4:172.25.100.1
dtmf-relay h245-alphanumeric
!
dial-peer voice 3000 voip
description <<Inbound From IPIPGW for G729 SIP Calls>>
session protocol sipv2
incoming called-number 3...$
dtmf-relay rtp-nte digit-drop
!
dial-peer voice 3500 voip
description <<Inbound and Outbound Calls for VM-AA-TM>>
destination-pattern 3[567]00
session protocol sipv2
session target ipv4:10.25.202.2
incoming called-number 399[89]....
dtmf-relay sip-notify
codec g711ulaw
no vad
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 2
Multicast call-legs: 0
Total call-legs: 4
3A1D : 4 265330ms.1 +70 pid:3000 Answer 2003 active
dur 00:00:14 tx:732/14640 rx:725/13940
IP 162.25.102.1:18928 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:
0/0/0ms g729r8
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
3A1D : 5 265340ms.1 +50 pid:3500 Originate 3500 active
dur 00:00:14 tx:731/116960 rx:462/73443
IP 10.25.202.2:16908 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:
0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
0 : 7 265480ms.1 +0 pid:0 Originate connecting
dur 00:00:14 tx:732/14640 rx:725/13940
IP 10.25.202.1:2000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:
0/0/0ms g729r8
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
0 : 9 265480ms.2 +0 pid:0 Originate connecting
dur 00:00:14 tx:731/116960 rx:447/71043
IP 10.25.202.1:2000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:
0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 2
Multicast call-legs: 0
Total call-legs: 4
Profile ID = 1, Service = TRANSCODING, Resource ID = 1
Profile Description :
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP Status : ASSOCIATED
Resource Provider : FLEX_DSPRM Status : UP
Number of Resource Configured : 4
Number of Resource Available : 4
Codec Configuration
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
Codec : gsmfr, Maximum Packetization Period : 20
Codec : g729r8, Maximum Packetization Period : 60
Codec : g729br8, Maximum Packetization Period : 60
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID
PKTS_TXED PKTS_RXED
0 1 4.4.20 UP 1 USED xcode 1 2
1823 1808
0 1 4.4.20 UP 1 USED xcode 1 3
1823 1028
0 1 4.4.20 UP N/A FREE xcode 1 -
- -
0 1 4.4.20 UP N/A FREE xcode 1 -
- -
0 1 4.4.20 UP N/A FREE xcode 1 -
- -
Total number of DSPFARM DSP channel(s) 4
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Date: Wed, 20 Feb 2008 16:02:17 -0800
Subject: Re: [OSL | CCIE_Voice] CUE FNA FB Problem
when you use the "service session" can you get an active call and
enter the command "sh call act voice brief" and find out what codecs
and what protocols are being used. We'll have a better idea of what
is going on when you post the answer to this question.
The service session command invokes the default session- a built-in
tcl script. I've posted the description of this tcl script below:
#sh call applicatio voice
Script Name : session
URL : builtin:app_session_script.tcl
Type : Service
State: Registered
Life : Builtin
Exec Instances: 0
Parameters registered under session namespace:
name type default value description
uid-len I 10 the number of digits in
UID
warning-time I 30 the time (in secs)
within which a user is warned before the calling time expires (call
terminates)
pin-len I 4 the number of digits in
PIN
retry-count I 3 the number of attempts
to reenter PIN
redirect-number S the telephone number
where a call is redirected to
Script Code Begin:
--------------------------------
TCL Script version 2.0 - 2.1
# app_session.tcl
#----------------------------------
# August 1999, Saravanan Shanmugham
#
# Copyright (c) 1998, 1999, 2000, 2001, 2002, 2003 by cisco Systems,
Inc.
# All rights reserved.
#------------------
#
# This tcl script mimics the default SESSION app
#
# If DID is configured, just place the call to the dnis
# Otherwise, output dial-tone and collect digits from the
# caller against the dial-plan.
#
# Then place the call. If successful, connect it up, otherwise
# the caller should hear a busy or congested signal.
# The main routine just establishes the statemachine and then exits.
# From then on the system drives the statemachine depending on the
# events it recieves and calls the appropriate tcl procedure
--------------------------------------------------------------------------------------------------------------
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-
On-Demand and Audio Certification Training Tools for the Cisco CCIE
R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
From: Jose Linero Welcker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 20, 2008 2:21 PM
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Hi Vik:
Thanks for your answer, to make the things clear, if I have
different codecs one of the legs shoud be H323, wright?, however if
I use G711 in both legs, then I can use sip-to-sip, wright?.
The other thing is, with the configuration I sent you adding the
command "service session" it works, the the dial-peer configuration
is:
dial-peer voice 4 voip
service session
session protocol sipv2
incoming called-number 3...
dtmf-relay rtp-nte
It is another way to fix it without changing the codecs or protocol
of the legs, my question is, could we use it in the lab exam?
Thanks,
Jose
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Date: Wed, 20 Feb 2008 14:13:27 -0800
you cannot sip - sip. One leg must be H323. You must g711 on the
inbound and outbound sip call legs to make this work.
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-
On-Demand and Audio Certification Training Tools for the Cisco CCIE
R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
From: Jose Linero Welcker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 20, 2008 2:01 PM
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Hi Vik:
Yes it is just for the calls coming from the WAN, What I am seeing
is the CME sends a SIP 302 message to the IPIPGW, and the IPIPGW
shoud send a re-invite to the number of the CUE, however the IPIPGW
sends an ACK and I heard a busy tone, attached are the
configurations of the IPIPGW and the CME, could you please take a
look at them.
Regards,
Jose
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; ccie_voice@onlinestudylist.com
Subject: RE: [OSL | CCIE_Voice] CUE FNA FB Problem
Date: Wed, 20 Feb 2008 13:53:02 -0800
Does the call-forward work from the PSTN?
If it is specifically the calls from over the WAN that do not fwd
then I go along with DevilDoc and Jason- transcoder or incoming dial-
peer not matching correctly.
If it doesn't work for g711u end to end, then that potentially rules
out the xcoder as being the cause.
Please confirm that it is only calls from over the WAN that cannot
be fwded and also post the output of debug voip dialpeer.
Vik Malhi - CCIE #13890, CCSI #31584
Sr Technical Instructor - IPexpert, Inc.
A Cisco Learning Partner - We Accept Learning Credits!
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Mailto: [EMAIL PROTECTED]
IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-
On-Demand and Audio Certification Training Tools for the Cisco CCIE
R&S Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice
Lab and CCIE Storage Lab Certifications.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf OfJose Linero Welcker
Sent: Wednesday, February 20, 2008 1:17 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] CUE FNA FB Problem
Hi all:
I have this configuration:
CM --- H323 Trunk --- IPIPGW --- SIP Trunk --- CME
The H323 is using G711 and the the SIP Trunk is using G729, the
calls between the phones in CCM and CME are working without
problems, however when I am going to forward the call to CUE due
CFNA or CFB I received a reorder tone and I am seeing this SIP
Message:
Sent:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 162.1.103.1:5060;branch=z9hG4bK92655
From: 'BR1-Phone2' <sip:[EMAIL PROTECTED]>;tag=8EC914-101D
To: <sip:[EMAIL PROTECTED]>;tag=1CEC58-818
Date: Wed, 20 Feb 2008 21:06:32 GMT
Call-ID: [EMAIL PROTECTED]
Timestamp: 1203541592
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Contact: <sip:[EMAIL PROTECTED]>
Diversion: <sip:[EMAIL PROTECTED]:5060>;reason=no-answer
Content-Length: 0
and the IPIPGW answer is an ACK not a reinvite, in the CME I have
configured this for CUE:
telephony-service
voicemail 3600
call-forward pattern .T
web admin system name Admin password cisco
dial-peer voice 6 voip
destination-pattern 3600
session protocol sipv2
session target ipv4:10.1.202.2
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
ephone-dn 1 dual-line
number 3001
call-forward busy 3600
call-forward noan 3600 timeout 10
!
Does anybody see this problem, I am trying to reach the voicemail,
but I can't.
Regards,
Jose
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