29 Jun
2004
29 Jun
'04
1:12 a.m.
What version of IOS are you using? I have a similar issue with a
Verizon DMS100 switch right now. It turns out that some 12.2
versions of IOS have an "incompatibility" with certain switches.
Cisco does not consider this a bug. They say that it is a difference
in the interpretation of the signaling standard.
In either case upgrading to 12.3 releases is suppose to fix the
problem according to Cisco. Instead I found that it improves but
doesn't necessarily fix the problem. Here is my story.
I am trying to use the CC-Diversion header so that when an
inbound call (to an IP phone) is not answered the call is redirected
out through the gateway to the DMS100 which then has an SMDI
link into our Octel 350 VM ystem.
The SER part has been working. Initially the redirected call
just hung, dead air, until I hung up the phone. You could see this
in the debug messages on the Cisco.
When I upgraded to 12.3.9 main line release I got the general
voice mail greeting regardless of which phone initiated the call.
Being the difficult person that I am I downgraded to a 12.3 T train
release to see what happened. Now if I call from my Centrex phone
on my desk I get the greeting associated with the Calling Party ID,
my Centrex phone. If I call from a non-Penn number however I get
the general voice mail greeting. Cisco has yet to explain what is happening
but they continue to claim the problem is resolved.
The one difference is that none of the releases I upgraded to are
listed on the feature report that Cisco published, however, I have not yet
been able to get one of the identified releases. The case is still open.
Then again as soon as we brought this to Verizon's attention they
tested the PRIs and said Oh, you aren't paying for voice mail service
on that trunk. We are waiting for them to determine if we have to
pay an additional fee for this feature.
I don't have the Cisco case in front of me but what you should look
into is how the called number is mapped to the Redirect Information
Element field. If you get stuck drop me another note and I'll see if
I can dig up the specific cae.
Good luck,
Steve
CM Rahman wrote:
Actually I have a Lucent Excel switch which is
connected to the cisco
as5400 via T1 Pri. Anybody here using Excel switch with a cisco ?
Right now, when ever I do debug q931 I get this below and it hangs until
my messenger times out and it disconnects. It should answer and give me
voice prompt. Anybody have deal with same scenario as mine?
*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: Applying typeplan for
sw-type 0xD is 0x2 0x1, Called num 5122200090
*Feb 18 15:40:20.142: ISDN Se7/0:3:23 Q931: TX -> SETUP pd = 8 callref
= 0x005F
Bearer Capability i = 0x8090A2
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Called Party Number i = 0xA1, '5122200090'
Plan:ISDN, Type:National
*Feb 18 15:40:20.158: ISDN Se7/0:3:23 Q931: RX <- CALL_PROC pd = 8
callref = 0x805F
Channel ID i = 0xA98381
Exclusive, Channel 1
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CTO
CCNP, MCSE Security "Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13704 Research Blvd. Building O-Suite 4
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
-----Original Message-----
From: Stephen Kingham [mailto:Stephen.Kingham@aarnet.edu.au]
Sent: Monday, June 28, 2004 5:50 AM
To: CM Rahman
Cc: Richard; serusers(a)lists.iptel.org
Subject: Re: [Serusers] as5400 and ser
CM Rahman wrote:
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworks
h
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I am sorry, I didn't show how put the pot in
my last email, here it is,
dial-peer voice 150 voip
description CCSi voip phone
destination-pattern 9T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:216.236.160.11
codec g723r53
Answer to your question, without putting "isdn protocol-emulate
network"
I wasn't able to get PRI Layer 2 up.
Yes. ISDN has a network side and a user side so that the layer 2
protocol Q921/lapb will work.
Most PABX want to be the user side.
Any other suggestion?
yes you have to have a pots dialpeer, the Cisco VoIP gateway requires at
least one, I think maybe one for each E1 port.
Take a look at the template I have posted here:
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
op/uas/ciscoVoIPGateways/as5300-12.3-6b-sip.txt
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CTO
CCNP, MCSE Security "Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13704 Research Blvd. Building O-Suite 4
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
-----Original Message-----
From: Richard [mailto:mypop3mail@yahoo.com]
Sent: Friday, June 25, 2004 4:11 PM
To: CM Rahman; serusers(a)lists.iptel.org
Subject: RE: [Serusers] as5400 and ser
Don't know why you have the following two lines,
isdn protocol-emulate network
isdn incoming-voice modem
Also you probably need a pots dial-peer...
Cisco web site has some configuration samples.
--- CM Rahman <cmrahman(a)ccsi.com> wrote:
g
Once I send a call via messenger, I don't
hear
anything other side. But
after a while it disconnect.
Here are the cisco config
******************************
controller T1 7/0:3
framing esf
pri-group timeslots 1-24
description Prism Test
***************************************
interface Serial7/0:3:23
no ip address
isdn switch-type primary-ni
isdn protocol-emulate network
isdn incoming-voice modem
isdn T310 180000
no cdp enable
!***************************************
dial-peer voice 150 voip
description CCSi voip phone
destination-pattern 9T
session protocol sipv2
session target ipv4:216.236.160.11
codec g723r53
*****************************************
*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: Applying
typeplan for
sw-type 0xD is 0x2 0x1, Called num 5122200090
*Feb 15 16:18:09.720: ISDN Se7/0:3:23 Q931: TX ->
SETUP pd = 8 callref
= 0x002E
Bearer Capability i = 0x8090A2
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98381
Exclusive, Channel 1
Called Party Number i = 0xA1, '5122200090'
Plan:ISDN, Type:National
*Feb 15 16:18:09.732: ISDN Se7/0:3:23 Q931: RX <-
CALL_PROC pd = 8
callref = 0x802E
Channel ID i = 0xA98381
Exclusive, Channel 1
*Feb 15 16:20:17.967: ISDN Se7/0:3:23 Q931: TX ->
DISCONNECT pd = 8
callref = 0x002E
Cause i = 0x8290 - Normal call clearing
*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: RX <-
RELEASE pd = 8
callref = 0x802E
*Feb 15 16:20:17.995: ISDN Se7/0:3:23 Q931: TX ->
RELEASE_COMP pd = 8
callref = 0x002E
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CCNP, MCSE Security "Secure your self by securing
your System"
CompTI Security Plus Certified
CCS Internet
http://www.ccsi.com
13704 Research Blvd. Building O-Suite 4
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Richard
Sent: Friday, June 25, 2004 3:27 AM
To: serusers(a)lists.iptel.org
Subject: RE: [Serusers] as5400 and ser
If you check this page,
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_
uide_chapter09186a00800eadfa.html
PSTN error "63 Service or option unavailable" is
mapped to sip error "503 Service or option
unavailable" which is in the header of the message.
Also the page shows why IP phone or PSTN generates
this and how proxy is supposed to do with it. Quote,
"The SIP gateway generates this response if it is
unable to process the request due to an overload or
maintenance problem. Upon receiving this response,
the
gateway initiates a graceful call disconnect and
clears the call. "
Look like a pstn config issue. Use "debug isdn
q931",
"debug isdn q921" and "term mon" for further
debuging.
Cheers,
Richard
--- CM Rahman <cmrahman(a)ccsi.com> wrote:
<sip:pappusip@backup.ccsi.com>;tag=c270cb2a9ab14343b72218adb808612
Looking through your cisco config file, I am
guessing your E1 are not
Pri. Ami I correct? I am dealing with a
channelized
DS3 with T1 Pri. I
will also share my config file after I can get the
call routed.
Currently I am getting this below. My
understanding
is there is
something wrong in the call going from cisco to
Pri
trunk. Anybody can
give me some clue, that will be great.
146.82.136.218:5060 -> 216.236.160.11:5060
SIP/2.0 503 Service Unavailable..Via:
SIP/2.0/UDP
>216.236.160.11;branch=z9h
> G4bKc513.1c338976.0,SIP/2.0/UDP
>65.70.207.66:8675..From:
>"pappusip(a)backup.c
> csi.com"
>
>
>
>
>
4;epid=c91b05026b..To:
<sip:915125656553@backup.ccsi.com>;tag=E8186070-487.
> .Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID:
>9fef06800312431fbaa33d389f7d
> 3ac7@192.168.1.101..Server:
>Cisco-SIPGateway/IOS-12.x..CSeq: 1
>INVITE..Allo
> w-Events: telephone-event..Content-Length: 0....
>
>
>
>
>
>
>
>
>
>
C.M.
Rahman Jr.
CTO
CCNP, MCSE Security "Secure your self by
securing
>your System"
>CompTI Security Plus Certified
>CCS Internet
>http://www.ccsi.com
>13704 Research Blvd. Building O-Suite 4
>Austin, TX 78750
>Tel: 512-257-2274 Ex: 115
>
>
>-----Original Message-----
>From: Stephen Kingham
>[mailto:Stephen.Kingham@aarnet.edu.au]
>Sent: Thursday, June 24, 2004 11:56 PM
>To: CM Rahman
>Cc: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] as5400 and ser
>
>Hi
>
>Along with several other we are putting together a
>SER implementation
>Tutorial for the R&E sector.
>
>We have a page up the the AS5300 and it may help
>you, also if anyone is
>interested in reviewing it?
>
>
>
>
>
>
op/uas/ciscoas5300.html
Regards
Stephen
CM Rahman wrote:
>Anybody here using cisco as5400 for PSTN
>
>
>
>
termination? I am having some
>problem with call routing. If there are such
>
>
>
>
person
>will to help,
>please
>
>
>
>