26 Jun
2004
26 Jun
'04
8:23 a.m.
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.
Any other suggestion?
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
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:
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
=== message truncated ===
__________________________________
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
http://promotions.yahoo.com/new_mail
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_g
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:
person
Anybody here using cisco as5400 for PSTN
termination? I am having some
>problem with call routing. If there are such will to help,
please
28 Jun
28 Jun
1:50 p.m.
New subject: [Serusers] as5400 and ser
CM Rahman wrote:
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/sipworkshop/…
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
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:
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
=== message truncated ===
__________________________________
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
http://promotions.yahoo.com/new_mail
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer
mailto:Stephen.Kingham@aarnet.edu.au
Telephone +61 2 6222 3575 (office)
+61 419 417 471 (mobile)
Voice and Video over IP
for The Australian Academic Research Network (AARNet) and
http://www.aarnet.edu.au
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_g
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:
person
>will to help,
>please
>
> Anybody here using cisco as5400 for PSTN
termination? I am having some
>problem with call routing. If there are such
>
>
7457
days inactive
7459
days old
1 comments
2 participants
participants (2)
-
CM Rahman -
Stephen Kingham