[Serusers] as5400 and ser
CM Rahman
cmrahman at ccsi.com
Tue Jun 29 04:51:57 CEST 2004
HI,
I am using Version 12.3(7)T1. I am doing very simple thing, routing the
call from a sip phone via ser to PSTN (cisco 5400). Right now, I have
send call via h323 from excel switch to cisco without any problem. But
when ever I am sending call to excel switch, it hangs.
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
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: Steve Blair [mailto:blairs at isc.upenn.edu]
Sent: Monday, June 28, 2004 5:13 PM
To: CM Rahman
Cc: Stephen Kingham; serusers at lists.iptel.org
Subject: Re: [Serusers] as5400 and ser
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 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/sipworks
h
>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/sipwork
s
>>
>>
>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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