[Serusers] as5400 and ser

[CM Rahman]

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 at aarnet.edu.au] 
Sent: Monday, June 28, 2004 5:50 AM
To: CM Rahman
Cc: Richard; serusers at lists.iptel.org
Subject: Re: [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/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 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/sipworks
h
>  
>
>>>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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>

-- 
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer 
mailto:Stephen.Kingham at 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