[Serusers] as5400 and ser

Sat Jun 26 07:23:22 CEST 2004

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 at yahoo.com] 
Sent: Friday, June 25, 2004 4:11 PM
To: CM Rahman; serusers at 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 at ccsi.com> wrote:
> 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 at lists.iptel.org
> [mailto:serusers-bounces at lists.iptel.org] On
> Behalf Of Richard
> Sent: Friday, June 25, 2004 3:27 AM
> To: serusers at 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 at ccsi.com> wrote:
> > 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 at backup.c
> >   csi.com"
> >
>
<sip:pappusip at backup.ccsi.com>;tag=c270cb2a9ab14343b72218adb808612
> >   4;epid=c91b05026b..To:
> >
> <sip:915125656553 at backup.ccsi.com>;tag=E8186070-487.
> >   .Date: Tue, 15 Feb 2000 01:38:28 GMT..Call-ID:
> > 9fef06800312431fbaa33d389f7d
> >   3ac7 at 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 at aarnet.edu.au] 
> > Sent: Thursday, June 24, 2004 11:56 PM
> > To: CM Rahman
> > Cc: serusers at 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?
> > 
> >
>
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworksh
> > 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
> 
=== message truncated ===

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