Dial URI Delay

Is there a way to code a delay into the dial string of the Dial URI? I need to add a 1-2 second delay in two places in the dial string. I have tried commas in the example below (old school), but that doesn't seem to have any effect.
<ExecuteItem Priority="0" URL="Dial:5120,,101#,,042#"/>
My other option here is to use TAPI, but I'd rather not have to do that if I can avoid it.
Also, can somebody clarify the difference between Dial and EditDial?
Thanks,
Andy

don't blame you wanting to avoid tapi. how about using the KEY: URI? that way, you can dial the number a key at a time, inserting pauses where you want to. of course, every key will be a separate CiscoIPPhoneExecute, but at least you'll be able to get the delay you need (and probably more!)
Sending a DIAL: to a phone causes it to dial the number sent. EDITDIAL pops up a dialog on the phone that allows the user to edit the number before dialing it.

Similar Messages

Secondary dial tone delay

I have a site that is having issues with secondary dial tone. They press 6 to receive access to make outside calls. When they press 6, there is a secondary dial tone but if you wait 5 seconds you will hear a flash and then another secondary dial tone.
The problem is during the 5 seconds, they receive incoming calls on the line before they can dial out. Any suggestions would be appreciated.
This is a Cisco UC520 that has been setup through CLI.

Thanks for the reply.
I do have the secondary-dialtone 6 command under telephony-service.
I have two dial peers with a pattern beginning with 6:
destination-pattern 6T
destination-pattern 6911
The interdigit timeout is set to 3.

EX60 delays from 10 to 30 sec when calling another device

This Terminal EX60 Works OK, but the only thing is that when calling another device takes too long for the other device to receive the call. This EX60 is registered to CUCM 10.5.1 and its firmware is tc7.2.0
ANY HINT?
Thanks so much for any help.

Are you dialing URI or extensions? If its extension, do you see some delay when dialing from a SIP phone on CUCM? You "might" be experiencing some post dial-delay. Try assiging an extension a URI and call this URI directly and see if you have a delay.

PC Dialer for CME

Hi,
We ha ve internaly developped an application in JTAPI to control phone connected to Call Unified Manager.
We'd like to port this application to CME but it seems JTAPI is not supported on CME.
One solution is to use TAPI but library are not for Java.
Is there another way on CME to control a phone via PC ?
The purpose is to transmit/receive call and lookup the corporate/customer directory for name association.
Thanks for your answer

If you'd just look for a way to dial.. there's always CiscoIPPhoneExecute with a Dial uri. However, for events I'm afraid you have no choice but to use the CME TSP.
Perhaps ATAPI.NET will help.. writing something in C# when you know Java is considerably easier than having to resort to a native language. I have ATAPI.NET working for basic tasks on a CCM but I've never tried with a CME (and I'd do anything in my power to push for a CCM replacement rather than deal with the CME).

Losing GPRS connection on HWIC-3G-GSM

Hi,
I have configured Cisco 2801 with HWIC-3G-GSM to work as a GPRS router.
The configuration is:
chat-script gsm "" "atdt*99***1#" TIMEOUT 60 "CONNECT"
chat-script pin "" "at+cpin=xxxx"
interface Cellular0/1/0
ip address negotiated
ip virtual-reassembly
encapsulation ppp
dialer in-band
dialer pool-member 1
dialer-group 1
async mode interactive
ppp chap hostname ""
ppp chap password 0 ""
interface Dialer1
ip address negotiated
encapsulation ppp
dialer pool 1
dialer idle-timeout 0
dialer string gsm
dialer persistent delay initial 20
dialer-group 1
access-list 1 permit any
dialer-list 1 protocol ip list 1
line 0/1/0
script startup pin
script dialer gsm
no exec
rxspeed 236800
txspeed 118000
Normally it works fine, but when the signal level is very low it stops to respond. So this is normal, but when the signal is back again, the connection is still lost, or more specific, in my eyes the router (dialer?) doesn't know that the connection is lost.
The test scenario looks like this:
- ping 172.16.1.1 repeat 500
- ping response is received
- I'm disconnecting an antenna
- pings stops working
- connecting back the antenna
- pings are still not working, and all I can do is just to reboot router
I think that it might be a problem with interoperability with a specific GSM carrier.
I have checked what is going on with modem during this test and it looks like this:
- before test
Data Connection Information
===========================
Data Transmitted = 145108 bytes, Received = 139940 bytes
Profile 1, Packet Session Status = ACTIVE
        IP address = 172.25.188.5
Network Information
===================
Current Service Status = Normal, Service Error = None
Current Service = Combined
Packet Service = EDGE (Attached)
Packet Session Status = Active
Current Roaming Status = Home
Network Selection Mode = Automatic
Country = POL, Network = PLUS
Mobile Country Code (MCC) = 260
Mobile Network Code (MNC) = 1
Location Area Code (LAC) = 31942
Routing Area Code (RAC) = 4
Cell ID = 33761
Primary Scrambling Code = 0
PLMN Selection = Automatic
Registered PLMN = Plus GSM , Abbreviated = PLUS
Service Provider =
Radio Information
=================
Current Band = GSM 900, Channel Number = 1
Current RSSI = -56 dBm
Band Selected = GSM all band
Modem Security Information
==========================
Card Holder Verification (CHV1) = Enabled
SIM Status = OK
SIM User Operation Required = None
Number of Retries remaining = 3
- after the antenna has been disconnected
Data Connection Information
===========================
Data Transmitted = 359592 bytes, Received = 346840 bytes
Profile 1, Packet Session Status = ACTIVE
        IP address = 172.25.188.5
Network Information
===================
Current Service Status = Emergency Only, Service Error = None
Current Service = Combined
Packet Service = None
Packet Session Status = Active
Current Roaming Status = Roaming
Network Selection Mode = Automatic
Country = POL, Network = Orange
Mobile Country Code (MCC) = 260
Mobile Network Code (MNC) = 3
Location Area Code (LAC) = 40100
Routing Area Code (RAC) = 110
Cell ID = 40085
Primary Scrambling Code = 0
PLMN Selection = Automatic
Radio Information
=================
Current Band = GSM 900, Channel Number = 99
Current RSSI = -95 dBm
Band Selected = GSM all band
Modem Security Information
==========================
Card Holder Verification (CHV1) = Enabled
SIM Status = OK
SIM User Operation Required = None
Number of Retries remaining = 3
- when I plug back the antena
Data Connection Information
===========================
Data Transmitted = 359592 bytes, Received = 346840 bytes
Profile 1, Packet Session Status = ACTIVE
        IP address = 172.25.188.5
Network Information
===================
Current Service Status = Normal, Service Error = None
Current Service = Combined
Packet Service = EDGE (Attached)
Packet Session Status = Active
Current Roaming Status = Home
Network Selection Mode = Automatic
Country = POL, Network = PLUS
Mobile Country Code (MCC) = 260
Mobile Network Code (MNC) = 1
Location Area Code (LAC) = 31942
Routing Area Code (RAC) = 4
Cell ID = 33761
Primary Scrambling Code = 0
PLMN Selection = Automatic
Registered PLMN = Plus GSM , Abbreviated = PLUS
Service Provider =
Radio Information
=================
Current Band = GSM 900, Channel Number = 1
Current RSSI = -58 dBm
Band Selected = GSM all band
Modem Security Information
==========================
Card Holder Verification (CHV1) = Enabled
SIM Status = OK
SIM User Operation Required = None
Number of Retries remaining = 3
I don't really know where is the problem, maybe someone had a similar problem?

Hi,
I want to sent some more info;
I have 2 g.shdls conneciton and one 3g ..dialer0 and dilaer1 works for g.shdsl and dialer 3 for 3G conneciton.
IZMIR_BURO# sh dialer
Ce0/2/0 - dialer type = IN-BAND ASYNC NO-PARITY
Dialer pool 5, priority 0
Idle timer (never), Fast idle timer (20 secs)
Wait for carrier (30 secs), Re-enable (15 secs)
Dialer state is physical layer up
Dial reason: Persistent Dialer Profile
Interface bound to profile Di3
Time until disconnect never
Current call connected 00:00:04
Connected to internet
Di0 - dialer type = DIALER PROFILE
Idle timer (120 secs), Fast idle timer (20 secs)
Wait for carrier (30 secs), Re-enable (15 secs)
Dialer state is data link layer up
Number of active calls = 0
Dial String      Successes   Failures    Last DNIS   Last status
Di1 - dialer type = DIALER PROFILE
Idle timer (120 secs), Fast idle timer (20 secs)
Wait for carrier (30 secs), Re-enable (15 secs)
Dialer state is data link layer up
Number of active calls = 0
Dial String      Successes   Failures    Last DNIS   Last status
Di3 - dialer type = DIALER PROFILE
Idle timer (never), Fast idle timer (20 secs)
Wait for carrier (30 secs), Re-enable (15 secs)
Dialer state is physical layer up
Number of active calls = 1
Dial String      Successes   Failures    Last DNIS   Last status
internet             18479          2    00:00:04       successful   Default
IZMIR_BURO#
IZMIR_BURO#sh int dialer 3
Dialer3 is up, line protocol is up (spoofing)
Hardware is Unknown
Internet address will be negotiated using IPCP
MTU 1500 bytes, BW 56 Kbit/sec, DLY 20000 usec,
     reliability 255/255, txload 1/255, rxload 1/255
Encapsulation PPP, loopback not set
Keepalive set (10 sec)
DTR is pulsed for 1 seconds on reset
Interface is bound to Ce0/2/0
Last input never, output never, output hang never
Last clearing of "show interface" counters 2w0d
Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 6339
Queueing strategy: weighted fair
Output queue: 0/1000/64/6339 (size/max total/threshold/drops)
     Conversations 0/1/16 (active/max active/max total)
     Reserved Conversations 0/0 (allocated/max allocated)
     Available Bandwidth 42 kilobits/sec
5 minute input rate 0 bits/sec, 0 packets/sec
5 minute output rate 0 bits/sec, 0 packets/sec
     169997 packets input, 201362608 bytes
     94633 packets output, 17498566 bytes
Bound to:
Cellular0/2/0 is up, line protocol is down
Hardware is HSPA/UMTS/EDGE/GPRS-850/900/1800/1900/2100MHz
MTU 1500 bytes, BW 2000 Kbit/sec, DLY 20000 usec,
     reliability 255/255, txload 1/255, rxload 1/255
Encapsulation PPP, LCP TERMsent, loopback not set
Keepalive not supported
Interface is bound to Di3 (Encapsulation PPP)
Last input 00:00:03, output 00:00:01, output hang never
Last clearing of "show interface" counters never
Input queue: 1/75/0/0 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: weighted fair
Output queue: 0/1000/64/0 (size/max total/threshold/drops)
     Conversations 0/2/16 (active/max active/max total)
     Reserved Conversations 0/0 (allocated/max allocated)
     Available Bandwidth 1500 kilobits/sec
1 minute input rate 0 bits/sec, 0 packets/sec
1 minute output rate 0 bits/sec, 0 packets/sec
     336436 packets input, 205597072 bytes, 0 no buffer
     Received 0 broadcasts, 0 runts, 0 giants, 0 throttles
     0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort
     298008 packets output, 20549654 bytes, 0 underruns
     0 output errors, 0 collisions, 18486 interface resets
     0 unknown protocol drops
     0 output buffer failures, 0 output buffers swapped out
     0 carrier transitions
     DCD=up DSR=up DTR=up RTS=up CTS=up
IZMIR_BURO#sh caller time
Line           User               Limit     Remaining Timer Type
vty 514        dpcmerkez          00:03:00 00:02:59 Idle Exec
Vi1            \
Vi2            \
IZMIR_BURO#sh ip route
Codes: C - connected, S - static, R - RIP, M - mobile, B - BGP
       D - EIGRP, EX - EIGRP external, O - OSPF, IA - OSPF inter area
       N1 - OSPF NSSA external type 1, N2 - OSPF NSSA external type 2
       E1 - OSPF external type 1, E2 - OSPF external type 2
       i - IS-IS, su - IS-IS summary, L1 - IS-IS level-1, L2 - IS-IS level-2
       ia - IS-IS inter area, * - candidate default, U - per-user static route
       o - ODR, P - periodic downloaded static route
Gateway of last resort is 0.0.0.0 to network 0.0.0.0
     83.0.0.0/32 is subnetted, 3 subnets
S       83.66.63.5 is directly connected, Dialer0
                   is directly connected, Dialer1
C       83.66.163.121 is directly connected, Dialer1
C       83.66.163.109 is directly connected, Dialer0
     172.25.0.0/24 is subnetted, 1 subnets
C       172.25.130.0 is directly connected, FastEthernet0/0
     10.0.0.0/32 is subnetted, 1 subnets
C       10.150.10.3 is directly connected, Dialer0
                    is directly connected, Dialer1
S*   0.0.0.0/0 is directly connected, Dialer0
               is directly connected, Dialer1
IZMIR_BURO#sh cell
IZMIR_BURO#sh cellular 0/2/0 all
Hardware Information
====================
Modem Firmware Version = K1_0_2_18AP C:/WS/F
Modem Firmware built = 12/29/08
Hardware Version = 1.0
International Mobile Subscriber Identity (IMSI) = 286027450382740
International Mobile Equipment Identity (IMEI) = 011955000180192
Factory Serial Number (FSN) = D66294005801004
Modem Status = Online
Current Modem Temperature = 26 deg C, State = Normal
Profile Information
====================
Profile 1 = INACTIVE*
PDP Type = IPv4
Access Point Name (APN) = internet
Authentication = None
Username: , Password:
* - Default profile
Data Connection Information
===========================
Data Transmitted = 17403856 bytes, Received = 201192534 bytes
Profile 1, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 2, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 3, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 4, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 5, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 6, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 7, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 8, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 9, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 10, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 11, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 12, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 13, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 14, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 15, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Profile 16, Packet Session Status = INACTIVE
        Inactivity Reason = Normal inactivate state
Network Information
===================
Current Service Status = Emergency Only, Service Error = Illegal ME
Current Service = Invalid
Packet Service = None
Packet Session Status = Inactive
Current Roaming Status = Home
Network Selection Mode = Automatic
Country = TUR, Network = VODAFONE
Mobile Country Code (MCC) = 286
Mobile Network Code (MNC) = 2
Location Area Code (LAC) = 10151
Routing Area Code (RAC) = 11
Cell ID = 21806
Primary Scrambling Code = 124
PLMN Selection = Automatic
Radio Information
=================
Current Band = WCDMA 2100, Channel Number = 10663
Current RSSI = -77 dBm
Band Selected = Auto
Number of nearby cells = 1
Cell 1
        Primary Scrambling Code = 0x7C
        RSCP = -82 dBm, ECIO = -7 dBm
Modem Security Information
==========================
Card Holder Verification (CHV1) = Disabled
SIM Status = OK
SIM User Operation Required = None
Number of Retries remaining = 3
IZMIR_BURO#

URI dialing changed Outlook contact card

Hello I have provided all the versioning info etc at the bottom of this post.
The issue we are experiences is as follows. Before enabling SIP URI dialing within our Jabber Client the contact details for a User would display the following fields
BusinessPhone = person's AD "telephoneNumber" field and the
Other = persons AD "ipPhone" field
This was accomplished by way of editing the following values in the jabber-config.xml file
<Directory>
<OtherPhone>ipPhone</OtherPhone>
<BDIOtherPhone>ipPhone</BDIOtherPhone>
</Directory>
After Updating the jabber-config.xml file to the one listed below, the following display problem is now occurring in Outlook contact Cards.
The person's Telephone number is being replaced by the SIP URI (See Screenshot)
Is there a way to have SIP URI dialing turned on but continue to show the proper phone numbers in the Outlook contact card? If so how is this accomplished?
Jabber for Windows = 10.5.0
Windows Version 7
UCM = 9.1.2.12900
IMP version = 9.1.1.419000
Our Jabber XML file looks like this
</Client>
<config version="1.0">
<Options>
<StartCallWithVideo>false</StartCallWithVideo>
<AllowUserCustomTabs>true</AllowUserCustomTabs>
</Options>
<Policies>
<InitialPhoneSelection>deskphone</InitialPhoneSelection>
<Disallowed_File_Transfer_Types>.MSI;.MP3;.AVI;.MP4;.386;.ACM;.ADE;.ADP;.ARJ;.ASP;.AVB;.BAS;.BAT;.BIN;.CGI;.CHM;.CLA;.CLASS;.CMD;.CNV;.COM;.CPL;.CRT;.DLL;.DRV;.EXE;.GMS;.GZ;.HLP;.HTA;.HTT;.INF;.INI;.INS;.ISP;.LNK;.MPD;.MSC;.MSI;.MSP;.MST;.OCX;.OPO;.PCD;.PHP;.PIF;.PL;.PRC;.RAR;.REG;.SCR;.SCT;.SH;.SHB;.SHS;.SYS;.TAR;.TLB;.TSP;.URL;.VB;.VBE;.VBS;.VXD;.WBS;.WBT;.WIZ;.WSC;.WSF;.WSH;.ZIP</Disallowed_File_Transfer_Types>
<EnableSIPURIDialling>true</EnableSIPURIDialling>
</Policies>
<Directory>
<OtherPhone>ipPhone</OtherPhone>
<BDIOtherPhone>ipPhone</BDIOtherPhone>
</Directory>
</config>

Thanks Christopher for the update on 2013
could you comment if there is any fix to how jabber itself displays contacts differently after enabling URI dialing. I find the users are confused by seeing a uri form and not selecting it even though it's a valid way to ring an extension. Should be nice to have both the uri and ipPhone listed.

Is there a way to put a delay on a dial peer in CME?

My users... Keep on dialing 9911, then hanging up because it was an accident. Then emergency services call back and get main automated attendant, then they dispatch out to the office pissed off. 3 times this month now...
Can I put a 1 second delay on the 911 dial peer so that if someone dials 9911, and realizes their mistake, they can hang up and no harm/no foul. In the off chance that it is a real emergency, the 1 second delay is not going to hurt anyone, but it will give the user that second to change his/her mind and hang up without further irritating emergency services.

Hello Clark,
refer to this
http://www.cisco.com/c/en/us/td/docs/ios/voice/dialpeer/configuration/guide/12_4t/vd_12_4t_book/vd_dp_feat_cfg.html
example
"9, 911." The comma in this example means that the router will pause for 1 second between dialing the "9" and dialing the "9" to allow for a secondary dial tone.
(,). comma indicates a 1-second pause in dialing
Br,
Nadeem

How to create a SIP URI speed dial

Hi - I would like to set up speed dials to SIP URI's (as opposed to a phone number).
Does anyone have an example of how the URI is specified in the web config screen and whether I need to then include a dial plan to handle the [email protected] URI. Thanks

IP dialling is enabled.
This is an example SIP URI (the Lenny URI :D)
sip:[email protected]
So what would the syntax for a speed dial be ?
Extended Functions: fnc=sd;sub=sip:[email protected];ext=sip:[email protected]

Outbound Dialer Delays when connecting calls

I have a particular dialer campaign that has a 5 seconds delay from when the callee picks up the phone and the IVR greeting starts playing; it goes something like this: ring, ring, hello (silence for +- 5 secs), IVR greeting starts.
The campaign is:
Transfer to IVR Campaign
     Enabled IP AMD
     Transfer to IVR Route Point
          Terminate tone detect
Dialing Mode:
Predictive_Blended
ICM 6 with IP IVR Queue Manager 3.5
We have 7 other campaigns with the same config and those ARE NOT experiencing the delay problem.
any help is greatly appreciated.
thanks

Hi Voiceops,
Do you have any updates for your case?
I also have same issue.
Thuc

Z30 Phone Dialing Delay

When using the dial pad on the phone, there is quite a delay between when I select a number and when it registers on the phone. That creates problems when I'm dialing because I can't tell if it's just being slow or I missed the dial pad for some reason. It wasn't always like this, so I am wondering if this is related to a software update. Does anyone else have this issue? Brad

Hello,
It could be related to a software update...did you do one? Give us details please.
Let us know!

My samsung stratosphere is very slow to place a call. I input the number and push CALL. Nothing happens. I try it again and nothing happens. Then some time later (2-5 mins) the phone goes ahead and dials. What is causing this delay in dialing?

my samsung stratosphere is very slow to place a call. I input the number and push CALL. Nothing happens. I try it again and nothing happens. Then some time later (2-5 mins) the phone goes ahead and dials. What is causing this delay in dialing.

wadams5641,
Let's add some speed to your service and ensure you can make calls without the
delay. When was the last time you turned off your phone? Is the software up to
date?
KarenC_VZW

A symbol to delay dial

In traditional fax/modem ,we can use ',' to delay dial action to waiting for greeting or fax signal,or remote respond.
For example:
when we dial 11111111,,,801,dial action will send 11111111 and wait 3 seconds ,then send 801.
Is there a same function symbol to added in prefix or mask in CUCM or CME?
Sent from Cisco Technical Support iPad App

An additional resource about this to add to Jaime's post, https://supportforums.cisco.com/docs/DOC-29944.

Long post dial delay.

I am trying to reduce end to end post dial delay. The numbers are international and domestic so 1+ and 011+ now I am using the 1T and 011T, which hunting method is the fastest and is there any trunking parameters that will add a # to the end of a number dialed out?
Thanks

Jason,
Here are a couple of things that can be adjusted to reduce the post dial delay.
1. Under the voice-port, reduce the interdigit timer by typing "timeouts interdigit x" where x is the number of seconds to wait in between each digit. The extension of this is that it will wait x seconds after the last digit.
2. You can use a translation rule to append a # to the end of the dialed number. This will be a little difficult for the international calls because the exact number of digits required to dial an international number is variable.
Create a translation rule similar to the following
translation rule 1
rule 1 1.......... 1..........#
This will append a # to all numbers beginning with a 1followed by 10 digits
translation rule 2
rule 1 011................... 011...................# (international number with 19 digits)
rule 7 011............ 011............# (international number plus 12 digits)
rule 8 011........... 011...........# (international number plus 11 digits)
rule 9 011.......... 011..........# (international number plus10 digits)
For the international translation rule. The longest match must be placed at the top of the rules for this to work as rules are matched in order.
Hope that helps.
Ademola

Cucm upgraded from 8.6 to 9.1 and now want to add uri dialing

Environment:
Upgraded 8.6.2 to 9.1
AD Sync users and authentication
Upgrade went well
I thought that with the upgrade the existing AD integration would bring over the Directory URI or give me the option to choose the Directory URI choices from the existing AD integration. I went to the existing LDAP Directory integration and I can't choose to sync Directory URI with MSRTCP... or email. I am forced to create a new LDAP directory so I can have the option to choose. Fair enough I said... I will create another one pointing to the same AD. I did that and then I was going to delete the old LDAP directory. It gives me a warning that all users that were created with this LDAP integration will be deleted. Yikes!!!
I'm stuck. Should I sync the new LDAP directory with the URI (which is pointed to the same AD as the old one) then delete the old LDAP Directory? Will my users stay intact?

I found a note in the documentation that expkained that MSRTCSIP... URI needs to be populated manualy or if the user was/is a Lync user it is populated automatically by Lync. Easy fix was add all users as Lync users then delete users as Lync users and then the AD attribute stays populated.

Cisco vcs "ghost" or fake URIs dialing attempts

A customer reported that there was many call attempts from non-registered or "fake" URIs in their VCS Control and VCS expressway, the consecuence is the saturation of the customer internet bandwidth due the numerous requests.

This has been discussed before in these forums. Have a look at some of the following threads:
nuisance-h323-calls-sx20
ghost-calls-tandberg-e20
sourceh323idcisco-incomingcalls
vcs-sip-abuse-reported
Wayne
Please remember to rate responses and to mark your question as answered if appriopriate.

Dial URI Delay

Similar Messages

Maybe you are looking for