Shared DN not forwarding to Call Handler

I'm sure I'm probably over looking something simple but here is my problem.
I have a shared DN that forwards to Unity but instead of the Call Handler greeting I am geeting the Auto Attendant. The Call Handler is set to record a message and send it to a public distribution list.
Anyone out that have any devine insight to my problem?

This is a direct call on and off network to the shared DN.
The routing rule states:
On
Both
Any
Any
2947
Any
Always
Send to greeting for NGAL J6 HELPDESK VOICEMAIL
I do not have the extension listed in the Call Handler.

Similar Messages

What's the best way to transfer (not forward) a call from one iPhone to another?

What's the best way to transfer (not forward) a call from one iPhone to another? Is there an app available that does this? I'm asking about receiving a call, then transferring that caller to another iPhone on a separate number and then disconnecting while those two users are joined up in a conversation.

Ask your carrier. This would be a feature provided by them.

No Call Forward to Call Handler

Hi,
we have cucm 8.6 and cuc 8.6
inbound call from ITSP transferred to CUC call handler for opening greetings and by pressing '0' it transfers to operator.
i want to configure custom busy message when operator is busy with other calls.
i configured another call handler with number 1660 and forwarded the operator calls to the call handler if busy.
the busy message is playing when i dialed the number 1660 from any extension, but the calls not getting transferred when the operator is busy.
i tried with forward all calls to the callhandler and getting the message ' cannot be reachable'.
i configured busy message on opening greeting call handler itself, in the callhandler "greetings" section's "busy" option, it didn't work as the call already got transferred to cucm.so its not coming back to callhandler if the ext is busy.
please can anyone guide me to configure busy message for operator.

Hi Anas,
i tried as you said,
its transferring the call if extension is busy by skipping welcome greetings, it's not what i wanted.
i need if any inbound call comes from ITSP first the welcome message with IVR plays, if the caller press '0' then only it should transfer to operator, if the operator is busy then it should play "Operators Busy" message.

CUCM not forwarding CTI called number - Subscriber Sign-in

I am running into an issue in CUCM/CUC 8.6(2a) when setting up external voicemail access.
I set up a CTI RP (Dn=2300) to forward to voicemail and set up a routing rule to send the Call
to subscriber sign in. The call keeps going to the opening greeting. I did a port status monitor
and I don't see the forwarding stations directory number. Just the Unity Pilot Point number.
I just set this up in CUCM 8.6(1) and I have it working. I mirrored the config but don't get the same results.
I'm not sure if maybe a service parameter changed or something and I am missing it.
TIA

Thanks for sharing your findings Rob. Your post helped me solve an issue I was having with a SIP trunk between CUCM and CUC.
Just to recap in case this post can help anyone else, if you don't check the Redirecting Diversion Header Delivery - Outbound then Call Manager will not forward calls with a redirect number or reason code to Unity. Using RTMT Port Monitor you'll find the call will complete but as a direct call to Unity.
The problem I was facing was CFNA (internal, external, etc) from a DN in CUCM to CUC would send the caller to the Opening Greeting, instead of the voicemail box of the user, which in this case was functioning properly because Redirecting Diversion Header Delivery was not checked.
Thanks again to both Robs for your thread,
Derek

Spa3102 would not forward a voip call to pstn line

Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a simple
network device . I have done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
The common changes to make for audio problems typically would be to setup a STUN server. A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server. This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Server:
The following web page has a list of "Public STUN Servers"
http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
I see XLite also has a setting to use a STUN server on the "Topology" tab.

System-wide Call Handler

Thanks for the help in advance.
I'm trying to set up a message that will play when any outward--facing number is dialed, saying that our offices are closed. This message will be used only for a couple of weeks, so I don't want to do any major rerouting. Here's what I have tried: In Unity, I have a Call Handler with the extension 3199 set up, as well as a Forwarded Routing Rule to route any dialed number in 6000-6999 to the Call Handler and go straight to the greeting. The Call Handler is set with an alternate greeting that is enabled and it is set to . I can call 3199 from my office phone and get the greeting, but I cannot get it to work when calling any number in the 6000s. We have other Call Handlers and Forwarding Routing Rules configured for numbers in this range, so I don't want to change anything, just override it for a couple of weeks. The FRR is set to the highest order, so that should not be an issue.
I have looked into CTI Route Points, and that may be the solution, but I have not had much luck here, either. We have several CTIs that are already configured, but they were created to route calls for a single extension (i.e. calling a department), not dozens of extensions to 3199. I did create a CTI, but where I'm running into trouble is with associating DNs to it. If I add a DN that is already associated with a phone (for instance, mine), it gives me an error about shared lines. I don't want to change any lines, but I do need all the calls in the range 6000-6999 (or even 6500-6999) to be forwarded to 3199. Is there an easy way to do this? Is there a way to add dozens of active DNs to the CTI? Or, am I doing it all wrong? Thanks again.

David Rojas wrote:Hello Josh,If i understand the call flow is moreless the following:PSTN-> Main Number-> Forward All -> Call Handler Alternate greeting.You have set up:Forward Routing Rule w/condition of 6000-6999 range-> Send call to: Go directly to greetings ->-Call handler w/ extension 3199 which plays Alternate Greeting.Issue: When calling number is in the 6000s range it does not match the FRR which is the first match on the list.Probing questions:1- What condition are you using: Forwarding stationg, Calling Number, Called Number? Have you changed the Field option?2- If you use a wildcard such as star (*) do you still experience the same issue?Reading the help page of the FRR:"Note that you can add multiple Calling, Dialed, Forwarding Number conditions to a single routing rule to create more complex patterns (for example, all numbers in the range 2000-3999 plus all numbers greater than 5000). When a call matches all conditions for the rule, the call is routed as specified in Send Call To. # Note : Your integration may not support this option. "3- The rule the previous fact, have you tested with a rule that matches a single number?Regards,Dâ√!d
I do not believe the Call Flow listed is how it is configured. How would I look at the Call Flow to see exactly which number it is using as the Main Number. We have a Main Number that we advertise, but I do not know if that is the number that you are referencing. 3199 is just a DN that was created and has no purpose outside of the Call Handler. It is listed as a DN in Call Manager, but there is nothing set to forward to it in Call Manager.
To follow up on your other questions:
I am using Dialed Number. I have tried using Equals 6*, as well as In 6000-6999, with the same results. It is set to Send Call To Call Handler, Go Directly to Greetings. If I change the Forwarding Rule to match one single extension, it still does not forward and goes directly to the number I called; so, if I call 6934, it goes directly to 6934.

Calls not being directed tomy System Call Handler

I built a small auto attendant for an acquisition group of users moving to our head office, there are two toll free numbers as well as 4 DIDs, the carrier began forwarding these on move day to the pilot number of my AA.
The issue is that the toll free numbers work but the DIDs do not, they are routing to the UC default opening greeting for some reason rather than my AA call handler.
It's as though for some reason call manager is sending something different for these DIDs as opposed to the TFs and the UC doesn't know what to do with them so it sends to the default greeting.
Any ideas as to what may be happening would be greatly appreciated.
Regards,
Andrew Bardowell
Thomson Reuters

Not to worry everyone problem solved, apologies for this time waster.
The DIDs were forwarded as redirecting numbers instead of sending without as with the toll free numbers even though I specifically asked the carrier to check this on cut over night.
Consequently when the toll free numbers come in they work correctly while the DIDs do not. The TFs because the tech who worked on those programmed them to send just the called number (caller ID) and the calling number.
The tech working the DIDs sent a redirecting number as well (the original DID) which confused Unity Connection.

Call forward to unity connection call handler

have the following setup:
cucm 8.6
cuc 8.6
a cti route point (DN 1000 )with forward all to unity connection call handler.
phone users who choose to have some quiet time can forward all their calls to the cti rp.
1. i need to allow only specfic DNs the ability to forward to the cti rp.
2. at times that the allowed DNs didnt choose to forward to cti rp then normal behaviour should occur - busy or unanswerd calls should go to DNs personal voice mail.
thanks

You can assign a specific partition to cti-rp and assign it to a CSS and this CSS should be assigned to those users (at call forward settings in line) only to whom you are trying to give this facility and for other users you can
or you can create a TP for it with specific partition and set the called number as per voice mail.
Suresh

CUCME 8.6 Call not forwarding Voicemail

Hi frieds,
     In our office we are using CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
My 2951 configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server.
Dial peer we are using for voice mail:
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad.
2901 Configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
ipv4 172.16.19.82
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad
============================
Debug CCSIP Calls
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0xAF40FD8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 5000
Called Number            : 1099
Source IP Address (Sig ): 172.16.19.80
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name         :
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.16.19.80
Source IP Port    (Media): 25364
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200
For your reference I here attach a network diagram
What the command which I missed?

Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
Sent from Cisco Technical Support iPhone App

HT4994 I have an Iphone 4 and can not forward calls. every time I put the number in the call forward field it jumps back a page and the button moves to off. can anyone assist.

I can not forward calls. I go to call forwarding, turn it on, copy and paste my phone number into the correct field (including the international dialling code).
I then tap the Call Forwarding button, which takes me back one stage and the Call Forwarding button goes to off after a couple of seconds. Anyone who can help?

Yes, a blocked call will go immediately to your voice mail.
If you would rather not have that behavior, ask your cellular carrier if you can block that person through the carrier.

Problem with call forwarding. Calls can not be forwarded for incoming external calls

Hi Everybody, how are you?
I have a problem with call forwarding. Everything was fine but now is not working.
In the reception of an office, the receptionist activate the call forward option to an internal extension. If somebody, internal in the office, call to the reception, the call is forwarding to the extension configured. But if I call from the outside (in example, from my cellphone) the call is not forwarded to the extension configured and continue ringing in the reception phone. Why this behavior? Any idea?
If you know something please tell me.
Thanks. Best regards.
Andres Collazos.

I encounter a similar problem with 9.1.1.
My problem is link to this bug ID : CSCtq10477.
Mathieu

1.call information such as call waiting,call forwarding, call holding is not working when caller the call me or im wait for my call i cant see any title by iphon , and also holding is the same.2. there is not any option for call baring.

Hi. please solve my problem.
.call information such as call waiting,call forwarding, call holding is not working when caller the call me or im wait for my call i cant see any title by iphon , and also holding is the same.2. there is not any option for call baring.3.playback music is not hearing by second partner during the call.4.i cant select ringing ton from saving tons.
thank you in advance.

Hi Ersin,
Exception 1 would seem to be FORMATTING_ERROR, which suggests something in the design of the Smartform. However, that doesn't fit with the form being generated from a report, but only erroring when called from a function module. I can think of no reason why the different calling method would be a factor.
When FORMATTING_ERROR is raise it should also set a message ID and number, are you able to determine what they are?
Regards,
Nick

Sap to non sap connection ( program not registered/cpic-call

Hi all,
 Im trying to connect sap to siebel system.IN TCP/IP connections iv selected register program id & passed the parameters ...program id,gateway host and gateway service
i followed michal krawczyk steps & also checked in smgw
transaction also but program id is not registered there
 im getting error like program sapprog not registerd cpic/call
Detailed error
 Error - program sapprog is not registsred
 location - sap gateway on host saple / sapgw01
 detail - Tp sapprog is not register
 component - gateway
 counter - 387
 module -gwr3cpic
 line- 1621
 return code - 679
 subrc - 0
 please try to help me out in this issue. all inputs r rewarded.
Regards,
Lavanya

Hi
Thanks prince,iv checked entries in sm59, i checked smgw tranaction(gateway monitor) for logged on clients
its showing like this
number/luname/tpname/sys.type/hostname/hostaddres/last req
0 /sapnav/sapgw00/local_r3/sapnav.sapdomain.com/192.158.3.219
if i double click on above...
read/write socket = 4
handle = 4
time = tue jan 26 16:54:13 2006
socket = 1652
stat = ni_connected
type = stream ipv4
out = 1122 messages 64 bytes
in = 1 messages 64 bytes
local =127.0.0.1:3300
remote = 127.0.0.1:1848
options =
lu = sapnav
tp = sapgw00
lon lu = sapnav
long tp = sapgw00
ext program = 0
no conversation = 24
appc version = 6
hostaddr[0] =
hostaddr[1] = 127.0.0.1
offset in = 0
timeouts in = 0
connected = tue jan 24
last req = mon jan
request block number in= -1
connection out = -1
target out = -1
offset out = 0
restlen out = 0
timeouts out = 0
request block number out = -1
accept info = einfo
snc forced = 0
just see gateway monitor 4 sapnav/parameters & attributesProfile
parameter Value
gw/max_sleep 20
gw/conn_disconnect 900
gw/conn_pending 60
gw/gw_disconnect 1800
gw/req_stack_size 30
gw/max_conn 500
gw/max_wp 0
gw/max_conn_per_wp 10
gw/cpic_timeout 20
gw/deallocate_timeout 600
gw/reg_timeout 60
gw/reg_lb_level 1
gw/reg_lb_default 20
gw/side_info C:\usr\sap\D01\DVMGS00\data\sideinfo.DAT
gw/sec_info C:\usr\sap\D01\DVMGS00\data\secinfo.DAT
gw/startup C:\usr\sap\D01\DVMGS00\data\gwstartup.DAT
gw/stat 0
gw/tcp_security 1
gw/cpic_security 1
gw/max_sys 300
gw/max_shm_req 50
gw/max_shm_req_per_conn 10
gw/max_overflow_size 5000000
gw/max_overflow_usage 10
gw/keepalive 300
gw/reg_keepalive 300
gw/compatibility 0
gw/monitor 1
gw/netstat_once 1
gw/close_routes 120
gw/timeout 0
gw/internal_timeout 0
gw/nitrace 0
gw/accept_remote_trace_level 1
gw/resolve_timeout 0
rdisp/max_gateways 100
rdisp/max_comm_entries 500
rdisp/sna_gateway sapnav
rdisp/sna_gw_service sapgw00
rdisp/TRACE_LOGGING OFF
gw/rem_start REMOTE_SHELL
gw/remsh rsh
exe/gwrd C:\usr\sap\D01\SYS\exe\run\gwrd.EXE
snc/enable 0
snc/permit_insecure_start 0
Attributes
Release 640
Release no 6400
internal version 2
start time Tue Jan 24 16:53:47 2006
build time Aug 17 2004 23:33:51
build with Unicode TRUE
build with Threads TRUE
gateway hostname sapnav
gateway service sapgw00
req_sync_limit 24
appc_ca_blk_size 34396
gwreq_ln 34588
appcdata_ln 34316
overflow_size_limit 500000
overflow_use 0
trace level 1
trace level external programs 2
PID Gateway Reader 4952
internal version 2
Shared memory Sizes in bytes
CONN_TBL entry 1748
CONN_TBL total 874000
STATISTIC area 1392
iv changed some values for my safety.plz check it and let me know the error to register program id. Hope ull solve my issue this time.
Thanks in advance,
Lavanya.

Re: [iPlanet-JATO] Re: session timeout when not submitting to a handler

Mark--
I know what's happening here, but am curious about your approach. You said
in an earlier email that you were generating links directly to JSPs, but
from what you are describing, you are generating JATO-style links to access
JATO pages. Nothing wrong with that, but there is a signficant difference.
Actually, it just occurred to me, I'm wondering what your URLs look like.
The way the request dispatching works in JATO is it ignores anything after
an initial "." in the final part of the URL path. For example, a request
for "/myapp/module1/MyPage.jsp" doesn't actually try to hit the JSP, instead
it tries to hit the JATO page "/myapp/module1/MyPage".
The end result is that you may think you are accessing a JSP directly, but
are instead accessing a JATO page. The reason the request dispatching works
this way is because it is illegal to access JATO JSPs directly, and there is
actually a (disabled) JATO feature that piggybacks on the use of the
dot-delimited URL.
So, now I need to understand your intent. I wasn't really sure why you were
generating direct JSP/page links to begin with. This works against the Type
II architecture JATO uses, in which all JATO requests go back to the
controller servlet.
If you are trying to design something like a menu page, you may have thought
that it was burdensome to create a number of HREF children, plus implement
event handlers for each of them. This definitely would be burdensome beyond
just a handful of links, but this is why JATO provides other mechanisms for
doing what I'll call here "polymorphic HREFs".
Assuming this menu page scenario, the easiest thing to do is to simply use
one HREF child on the page, and add a value to it each time it is rendered
that distinguishes it from the other instances on the page. In your event
handler for the HREF, you simply check this value and use it to decide which
page to forward to. You can add a value to an HREF or Button by using the
"addExtraValue()" method. Or, if you are using JATO 1.2, you can add extra
query string NVPs right in the JSP document using the "queryParams"
attribute of the <jato:href> tag. Thus, your one HREFchild and event
handler become "polymorphic" because what they do depends on the context in
which they are invoked.
Now, I still don't have confirmation that this is what you were trying to
do, so until I do, let me explain the exception you're seeing. JATO assumes
that when a request comes in for a page that includes the pageAttributes
NVP, it is a request coming from a previously generated JATO page. Because
of the way JATO works, this means that the request dispatching code should
send the request back to the originally rendered page. For example, if Page
A renders an HREF, which the user then activates, JATO sends the request
back to Page A for handling. All of the HREFs and forms generated during
rendering of Page A actually refer back to Page A, regardless of where those
links or buttons actually pass the request in their event handlers/Command
objects.
So, what's happening when you include the pageAttributes in your HREFs is
that JATO is assuming that a request is being sent to the target page, with
the assumption that the target page has a mechanism in place to handle the
request. This assumption relies on the specification of the "originator" of
the request being specified in the request. For links/HREFs, the name and
value of the HREF is sent along with the request. For forms, the name and
value of the button that was pressed are sent in the request. JATO uses the
presence of these name/value pairs to decide which event handler, or which
Command object, to invoke to handle the request.
The exception you are receiving is saying that there was no object on the
target page that indicated it could handle the request. This is to be
expected, since you have not specified a query parameter that indicates
which CommandField child is responsible the request. However, this is where
I see the disconnect, because that is not what I believe you were trying to
do (as explained above).
So now, given all the information above, can you tell me what you're trying
to accomplish, and whether or not the info I've given you has helped you to
design a mechanism more in line with a JATO approach? If not, given that I
understand what you're trying to do, I can offer a more concrete solution.
Todd
----- Original Message -----
From: <Mark_Dubinsky@p...>
Sent: Monday, November 05, 2001 2:54 PM
Subject: [iPlanet-JATO] Re: session timeout when not submitting to a handler
This is the exception we get:
(And BTW, leaving a blank value for the pageAttributes doesn't help)
[05/Nov/2001 17:49:18:4] error: <portalServlet.processRequest>
javax.servlet.ServletException: The request was not be handled by the
specified handler
at java.lang.Throwable.fillInStackTrace(Native Method)
at java.lang.Throwable.fillInStackTrace(Compiled Code)
at java.lang.Throwable.<init>(Compiled Code)
at java.lang.Exception.<init>(Compiled Code)
at
javax.servlet.ServletException.<init>(ServletException.java:107)
at
com.putnaminvestments.common.jato.ApplicationServletBase.dispatchRequ
est(Compiled Code)
at
com.putnaminvestments.common.jato.ApplicationServletBase.processReque
st(Compiled Code)
at
com.putnaminvestments.bp.portal.portalServlet.processRequest(Compiled
Code)
at
com.putnaminvestments.common.jato.ApplicationServletBase.doPost(Compi
led Code)
at
com.putnaminvestments.common.jato.ApplicationServletBase.doGet(Compil
ed Code)
at javax.servlet.http.HttpServlet.service(Compiled Code)
at com.putnaminvestments.bp.bpServletBase.service(Compiled
Code)
at javax.servlet.http.HttpServlet.service(Compiled Code)
at
com.netscape.server.servlet.servletrunner.ServletInfo.service(Compile
d Code)
at
com.netscape.server.servlet.servletrunner.ServletRunner.execute(Compi
led Code)
at com.kivasoft.applogic.AppLogic.execute(Compiled Code)
at com.kivasoft.applogic.AppLogic.execute(Compiled Code)
at com.kivasoft.thread.ThreadBasic.run(Native Method)
at com.kivasoft.thread.ThreadBasic.run(Native Method)
at com.kivasoft.thread.ThreadBasic.run(Native Method)
at com.kivasoft.thread.ThreadBasic.run(Native Method)
at com.kivasoft.thread.ThreadBasic.run(Compiled Code)
at java.lang.Thread.run(Compiled Code)
--- In iPlanet-JATO@y..., "Todd Fast" <Todd.Fast@S...> wrote:
Mark--
Initially we tried to add the pageAttributes NVP as well, but that
was
causing an exception, so we stopped doing that.That's odd--what was the exception?
Our problem now is that when the SessionTimes out it does not go
to
onSessionTimeout method as in processRequestMethod of the
ApplicationServletBase it looks for pageAttributes. If it is notnull
then only onSessionTimeOut method is called.This is sadly the only technique for determining if a session hastimed out
and a new one been created, versus the initial creation of thesession.
Is there any work around for this? Maybe you can suggest how topass
the pageAttributes without causing the initial exception?Definitely--let me know what the exception was and I'll be able tosuggest
something. However, it shouldn't really be any harder thanappending a
"jato.pageAttributes=" empty NVP on the HREF.
Todd
Todd Fast
Senior Engineer
Sun/Netscape Alliance
todd.fast@s...
For more information about JATO, please visit:
http://developer.iplanet.com/tech/appserver/framework/index.jsp

OK, here's what I'm trying to do: We have, like you said, a menu
page. The pages that it goes to and the number of links are all
variable and read from the database. In NetD we were able to create
URLs in the form
pgXYZ?SPIDERSESSION=abcd
so this is what I'm trying to replicate here. So the URL that works
is
pgContactUs?GXHC_GX_jst=fc7b7e61662d6164&GXHC_gx_session_id_=cc9c6dfa5
601afa7
which I interpreted to be the equivalent of the old Netd way. Our
javascript also loads other frames of the page in the same manner.
And I believe the URL-rewritten frame sources of a frameset look like
this too.
This all worked except for the timeout problem. In theory we could
rewrite all URLs to go to a handler, but that would be...
inconvenient.

Setting up PLAR with Cisco Unity Connection Call Handler

This is a lab setup and Im doing it to learn. No customer involvement.
Setup
Analog phones - FXS Port - 2951 Router - FXS port - FXO Port - 2951 Router - CUCM 9 - UC 9
|------------PSTN Emulator-------------| |-----MGCP GW----------|
I have a CTI Route Point configured as DN 7000 and it has the default VM profile. The CTI RP is set to FWD All to VM.
The FXO port is set to PLAR to 7000.
When I dial from the PSTN analog phones through the FXO port, I hear the first ring, the FXO port answers, then I hear what sounds like the recorded message beginning to play. Immediately after, I hear the recorded message "You cannot be transferred to this number. Check the number and try again."
I dont do anything and within a second or two, I hear the recorded message for the system call handler start.
I did some more testing. I added an E1 trunk between the PSTN and the MGCP gateway. In the CUCM, I created another CTI Route Point with a DN that I could dial from my PSTN cloud. I also set that CTI Route point up to Call FWD all to VM.
When I dial using the E1 trunk, the call hits the system call handler as expected and I hear my recorded greeting (as expected).
However, calling through the FXS-FXO tie line consistently gets me the error message recording followed by my recorded greeting.
Im currently using loopstart on the tie line, though I have also tried ground start with no difference.
Any ideas?
Jeff

If I understand correctly, you want outside calls to go directly to the call handler but internal calls to ring whatever phone this extension is on.
If I am understanding correctly, then this will probably resolve it. For the purposes of the explanation I will assume that extension 1000 is the number in question:
Create a new partition, we'll just call it ToVM or something like that
Create a new CTI route point with extension 1000 and put it in the ToVM partition, forward all calls for this CTI route point to voicemail
If you don't have one already, create a calling search space for the voice gateway. Call it Gateway-CSS. This should have the same partitions that the gateway can normally call, but it should also have the ToVM partition and that partition should be HIGHER in the list than the partition that has the normal extension 1000 on it.
Apply Gateway-CSS to the gateway
Configure the normal extension 1000 (not in the ToVM partition) the way that you want it to work.
Now when external callers dial 1000 they will go to the call handler because that partition is higher in the CSS and the CTI Route point should be hit first. Internal callers will ring the phone (or whatever it is) because they only have access to the regular internal partition (and not ToVM) that 1000 is in.
This is all assuming that I understand you correctly!

Shared DN not forwarding to Call Handler

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