Transfer call from fixed line to BB Z10
Hello
When you transfer calls from fixed line to BB Z10 the device only gives a split second notification of transfer call and then you do not see it any more.
So it is difficult to see the difference between direct calls and my transfered call from my fixed line.
Is it possible to keep the notification of transfered call "on screen" when it is dialing?
With friendly greets
Karel
The Z10 gives a split second a notification that the incomming phonecall is a transferred call and then shows the phone number of the original caller. Afterwords there is no way to see the differance between the straight calls and the transferred.
With my old Nokia N8 there was an onscreen "green arrow" notification.
After searching a little on the web I found out that only the old symbian phones (and some other) had that function.
https://supportforums.blackberry.com/t5/BlackBerry-Bold/Bold-Diverted-Calls-Symbol-or-Icon/td-p/9232...
Its a pitty it doesn't work on a BB Z10. I guess it will not be changed as the post above is from 2008 and now we are 2014.
With friendly greets
Karel
Similar Messages
-
Transfer call from federated failed
Hi
I am testing transfer call from federated client , I am transferring it to another lync user or to PSTN and both giving transfer not complete with error
can not complete transfer error ID 404 , source ID 239
the nice thing that I am not seeing any error with red flag in the S4 & SIP Stack event of the lync edge and FE , and when I search for the contact of the transferred user I can't find his name too in logs !
any suggestion will be helpful
by the way the internal transfer for incoming/outgoing working fine , I tried to disable/enable refer but same result , anyway I think the transfer to internal LYNC client doesn't need refer changes as it is for gateway only
I am testing the call from my Skype account
ThanksI would fix OCSLogger on your edge server and post that log:
Issues with OCSLogger and the Outbound routing messages in Snooper on Lync 2013.
Try the following:
Close Snooper and OCSLogger
Rename the default.tmx file in C:\Program Files\Microsoft Lync Server 2013\Debugging Tools directory to default.tmx.old
Copy the default.tmx file from c:\Program Files\Common Files\Microsoft Lync Server 2013\Tracing To C:\Program Files\Microsoft Lync Server 2013 Debugging Tools.
Start OCSLogger again and run your scenario. -
Transfer Upgrade from one line to another on the same plan.
Is it possible to transfer an upgrade from one line to a second line, and then have the phone that was used on the second line, be switched to the first line? If so, how would I go about doing this online?
Yes. The account owner should be able to log in to MyVerizon and click on "Transfer this upgrade" when they view all of the upgrades available for their account. If they don't have that option, another way is to just order the phone normally, then activate the new phone on the line it was ordered for briefly to accept the Terms & Conditions of the new contract. Then you can activate the old phone from that line again, which will free the new phone up to be activated on the second line.
If you're getting a 4G phone, you'll want to call in to Customer Service before you start activating though, because the 4G SIM cards can complicate the activation process. -
Incorrect Caller ID on calls from outside line via FXO port.
Have a public phone line connected to my CUCME 2801 router VIC2-2FXO card. All inbound calls are passed to DN-5001 (group number). Can receive and send calls without a problem, but incoming calls all show "911" for caller ID. Think this is simply an issue with the out bound dial-peer, of which the lowest numbered out bound dial-peer is for 911 services. Not sure how to correct this so inbound calls show the proper caller ID?
Below is a copy of my CUCME show run output from the FXO port config thru all the dial-peers. Any pointers is greatly appreciated.
Thanks.
Kirk E.
voice-port 0/0/0
connection plar opx immediate 5001
voice-port 0/0/1
voice-port 0/2/0
station-id name POTS
station-id number 7000
voice-port 0/2/1
ccm-manager config
dial-peer voice 7000 pots
destination-pattern 5006
port 0/2/0
dial-peer voice 90 pots
description Emergency Services
destination-pattern 911
port 0/0/0
forward-digits 3
dial-peer voice 91 pots
description 10 Digit local dialing
destination-pattern [234].........
port 0/0/0
forward-digits 10
dial-peer voice 92 pots
description 11 Digit local/long distance dialing
destination-pattern 1[2348].........
port 0/0/0
forward-digits 11
dial-peer voice 93 pots
description Long Distance
destination-pattern 011T
port 0/0/0
prefix 011
dial-peer voice 94 pots
description Backup bench POTS phone
destination-pattern 7000
port 0/2/0
dial-peer voice 2 voip
destination-pattern 51..
session protocol sipv2
session target ipv4:172.16.2.155
dtmf-relay sip-notify
codec g711ulaw
no vadHi
Can you find the below:-
Hi
1- Please find the below table as the following link http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
Caller ID Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
under voice-port
caller-id enable
2-If above configure and still have no caller id , please add the below commannds to the voice-port
caller-id alerting line-reversal
cptone ? "based on your"
caller-id alerting ring 2 "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
4-Do debug to make sure all ok
"debug vpm signal "
[0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
Thank you
please rate all useful information -
Why Siri cannot call out (Fixed Line or Mobile No.) in Malaysia?
Just bought a iPhone 4S 64GB with local mobile service provider (Digi) with 24 months contract (on 28 Dec 2011), but same problem happen in Malaysia, which Siri cannot call out to Fixed Line or Mobile No. (Siri will reply that: Sorry, I can't call Home's home number.)
Really hope that Apple can provide a solution to solve it ASAP.According to this forum: http://forum.lowyat.net/topic/2175165/all, it seems that Siri is expecting a 7-8 digit phone numbers excluding the country code and area code, so adding an extra "0" digit or two at the back of the phone number may be able to fix your problem. The phone number should still connect correctly. Try that, it works for me in Indonesia where some phone numbers are still 6 digits.
Hindra -
Transfer call from 78xx phone series
we are experiencing a problem on phone 78xx series when transfer a call.
When the 7861 telephone transfers a call and the called party does not answer, can not resume the call on hold.
I configure my sip profile in any particular way?We recently bought some 78xx IP phone and we found the same problem: when I try to transfer a call to another phone (either an internal or external phone) I cannot cancel the transfer, neither by pressing the line button nor the cancel softkey.
Once the cancel button is pressed the call is automatically diverted from the originating caller to the third phone, and is terminated on 78xx phone.
We already installed the latest CUCM version (9.1.2.11900) and phone firmware (sip78xx.10-1-1-9) but the issue is still present.
Hope it will be solved in next firmware update. -
PS3 droplet called from command line in XP reads in files but does not execute action
Hello.
I am a researcher interested in anxiety disorders in young children. As part of a functional magnetic resonace brain imaging study we need to compare children's responses to familiar and unfamiliar faces. The latter are no problem, but for the former we need to take pictures of the children's mothers and process them "on the fly" so that they can be incorporated into stimulus sets presented in the MRI machine (otherwise known as the "magnet").
For presentation in the magnet, the images have to be in a particular format. I have created an action that produces the appropriate format, and a PS3 droplet that behaves appropriately (outputs correctly modified files to the stated address) when a Windows XP (SP3) folder is dropped on it.
However, I need to automate the procedure further because it will be executed by individuals with little or no understanding of PS etc.
It occurred to me that I could call my droplet from the XP command line with the folder containing the relevant files as an argument (and then I would be able to incorporate this function into an overall control program).
However, I have found that this approach loads the relevant files into CS3, but that the actions don't run.
I would very much appreciate any help with this problem.
Thank you.
Adrian Angold.A command line script might work, but would lack a user interface. Error
processing and logging capabilities would also be limited.
I would consider writing a small application in a language such as Visual
Basic or C# that uses Photoshop's automation interface. The automation SDK
is provided on the Photoshop DvD. -
Transfer Calls from dect on FXS port (STCAPP)
Dear All,
I have a Dect phone that is connected to an FXS card in a Cisco Voicegateway 2900 series, i have configured it under stcapp.
Now the phone does work but it can't transfer to other internal phones. My question is how do i configure this, i have looked into stcapp supplementary services.
Voiceport
voice-port 0/3/1
cptone BE
timeouts interdigit 5
description === DECT 15360 ===
caller-id enable
Dial-peer
dial-peer voice 700 pots
service stcapp
port 0/3/0
dial-peer voice 701 pots
service stcapp
port 0/3/1
stcapp supplementary-services
port 0/2/1
fallback-dn 98307
port 0/3/0
fallback-dn 15360
port 0/3/1
fallback-dn 15360This is how to do transfer for a sccp controlled endpoint..
"During an active call, user presses hookflash and receives dial tone. User dials number for transfer and either stays online to announce (consultive transfer) or hangs up (blind transfer). When user hangs up, the call is transferred"
Please refer to the sccp supplemetary services feature guide
http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxssccpsplmft.html
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay" -
CAD - TRANSFER CALLS FROM THE APPLICATION
In CAD, when the agent press the "Transfer" button, the customer is still connected to the agent. I mean, you need first to dial a number, and then the call is really transferred.
This is confusing our agents. Is there any way to change this? Thanks in advance!Hi, I am afraid it cannot be changed. When you transfer a call using CAD, it first takes the numbers and then it opens a second channel (initiates a second call).
When transferring a call using the phone, it needs to open a second channel first before it can take a number so this is why the frist call must be held first.
G. -
How do I transfer calls from CM 7.x to ICM script in CVP environment with no SIP.
So I have a customer who currently send all internal help desk calls to PSTN, then they come to VXML Gateway that goes to CVP finally landing on ICM script to be processed and handled by an internal agent.
Ideally we want to keep this call with in the platform. No SIP here its H323 gateways and gatekeepers, dialpeers etc.
current setup is you dial 5000 that gets translation route to dial out PSTN outdial 866 123 4567 lands back on VoiceGateways with DNIS 4567. Eventually based on DNIS that get sent to ICM where its processed based on Dial script associated with DNIS and call type.
Seems something real simple in IPIVR but CVP makes it CVP-complex visually perplexing.
Thanks,
Baseer.CUCM originated calls can be processed in CVP using H.323 - look at the sections in the Guide for warm transfers. The mechanics are similar to SIP.
Regards,
Geoff -
Calling this simple servlet from command line -- ERRORS!
Below is my servlet. I call from command line via:
java BatchServlet
and I get:
Exception in thread "main" java.lang.NoClassDefFoundError: BatchServlet
IS there a reason for this
import java.io.IOException;
import javax.servlet.*;
import javax.servlet.http.*;
public class BatchServlet extends HttpServlet implements Runnable{
static Thread t = null;
public void init(ServletConfig c) throws ServletException{
super.init(c);
if (t==null){
t = new Thread(this);
t.start();
public void run(){
while (true){
try{
Thread.sleep(5000);
}catch (InterruptedException ie){
ie.printStackTrace();
System.out.println("Wake up");Same error with this little prog.....
Notice main method
package wch.util;
import java.io.IOException;
import javax.servlet.*;
import javax.servlet.http.*;
public class test {
public void main(){
System.out.println("test"); -
Transfer files from SAP R/3(unix server) to windows
Hi TechGurus,
We have a requirement where in , we are extracting data from SAP table into a text file using OPEN DATASET in TEXT MODE.
Then using a third party tool (Connect-direct), we are transferring text files to windows server using unix script, in binary mode.
The problem is that that the data is being transferred in blocks,each time beginning transfer from a new line.That is, after say transferring 25KB data it transfers next 25KB data from a new line. What happens as a result is that records
after particular amount of data is broken and remaining data transfer begins from new line.
The file size is around 25 MB.There is no such problem while transferring file of size 10 KB.
We are uploading the data from text file on windows server into MSAccess.
And error occurs during this process.
Assured points for helpful answers
Thanks in advance!!Hi,
If you want to transfer file between different SAP servers then you can use the function module EPS_FTP_PUT. You need to have a RFC destination (with sufficient authorizations for the RFC user). The meaning of the importing parameters of the function module are quite obvious. Of course, SAP offers us more than this function module:
- EPS_FTP_GET
- EPS_FTP_MPUT
- EPS_FTP_MGET
Kishi. -
Transfering calls from a POTS phone
Hi,
I have 2 analogue phones plugged into FXS ports on a Cisco 2821 router. The router is connected to a converged network using Callmanager 4.1(3) and running H.323.
I can make and receive calls between the analogue phones, between them and IP phones and from them over the PSTN. However I cannot transfer calls from them either to other analogue phones or IP phones.
I also get the same problem using other types of routers.
I would be very grateful for any help received.
Regards,
Gerry Seal.I don't think this is possible with H323, but this is available in SIP.
Configure the gateway to use SIP and add this gateway as a SIP Trunk in Callmanager.
Check this link for more details.
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a00801541d3.html
Regards,
Anup -
Calling from FXO ???
Hello,
I have a question, just to confirm something, here is the scenario:
A customer has a connection between a 3640A router and a Definity PBX using Analog signalling: FXO ports in the Router, and Station lines in the PBX.
This 3640A router is connected to many 1751-V routers in a Hub&Spoke connection. In these remote locations there are FXS ports connecting directly to analog phones.
For I have done before, I know that the behavior of this network should be as follows:
1) The remote sites could call to the central site to an extension or to the PSTN.
2) The remote sites could call to the others remote sites as well (IP routing is being used - Full IP connectivity)
3) The Hub site users CANNOT dial to the remote sites because the central ports are FXO and those ports do not provide a dial tone. I think the only way the users form the central site can dial to remote sites is using static PLAR connections from each FXO port to a remote site.
For points 1) and 2) I am pretty sure it works, BUT the customer just told me that they can dial to the Spokes using FXO ports with other routers...
If this can be done ... do I have to configure something in the cisco voice-port OR in the PBX ???
TIA
Regards!!!When the Hub calls to the extension where the FXO port of 3460 is connected this port opens automatically, giving the opportunity to mark the remote voip dial peer. Once the caller port (a telephon set connected to extension port or a call from CO line) is closed, the FXO port will close too, and so will remote the FXS port. You wont have any problems.
However, by definition you can use the PLAR connection to only one (and from only one) remote destination. Finally, you may experiment ring back problems. -
How can i transfer a call from SIP 9971 to PBX system on CME router
hello everybody,
I have a critical problem about interaction of transfering feature between CME router and pbx panasonic system in some status. let me explain more detail about this issue..i have a SIP 9971(CP-9971) registered on CME at the one site and a voice gateway that is connect with PBX system through a E1 pri trunk connection at the other site. totally the integration between CME and PBX is ok and there is no problem in two direction, i mean i can call pbx system from cp-9971 and vise versa but when i call from a phone which is registered on PBX site to SIP 9971 which is registered on cisco CME call is connected,then when i try to transfer that call to another phone at PBX site, the session is open between two panasonic phones but no audio transmited in two direction. in addition every thing works fine about SCCP phones(transfer feature works fine). here is my configuration file. i hope someone could help me because i've searched a lot but no result help help help plz....
cme router 3845 configuration
VOIP-3845#show running-config
Building configuration...
Current configuration : 12657 bytes
! Last configuration change at 11:44:01 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname VOIP-3845
boot-start-marker
boot-end-marker
no aaa new-model
clock calendar-valid
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
registrar server
voice register global
mode cme
source-address 192.168.2.1 port 5060
max-dn 720
max-pool 262
load 9971 sip9971.9-1-1SR1.loads
authenticate register
authenticate realm cisco.com
tftp-path flash:
file text
create profile sync 0063544528862458
camera
video
voice register dn 1
number 500
voice register dn 2
number 600
voice register dn 3
number 700
name test
voice register template 1
softkeys idle Newcall Redial Cfwdall
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
id mac B8BE.BF23.5242
type 9971
number 1 dn 1
template 1
username test password test
camera
video
blf-speed-dial 4 600 label "test"
voice register pool 2
id mac B8BE.BF9C.5476
type 9971
number 1 dn 2
template 1
username bank password bank
camera
video
voice register pool 3
id mac B8BE.BF9C.51D4
type 9971
number 1 dn 3
template 1
username test1 password test1
camera
video
voice register pool 4
id mac B8BE.BF9C.4FA2
number 1 dn 1
camera
video
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1576175886
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1576175886
revocation-check none
rsakeypair TP-self-signed-1576175886
crypto pki certificate chain TP-self-signed-1576175886
certificate self-signed 01
30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31353736 31373538 3836301E 170D3131 31303038 30393034
34365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 35373631
37353838 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100D6EC 47BCDC3C 82F43FF3 23522678 2616868D 9910DCD2 E36016B3 D7B40DA7
53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
A3055C06 AE860756 425E5C50 E62B3287 631B1E87 9BAC2E39 2810E120 DA3BF823
947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
00038181 000E82F6 5FBB847C 49226955 6F7DECE7 0B093513 D57C35D5 4CD22FA7
8144A080 B0D56C8D 86AF8156 0152443A A3FBE59F B1AEFFBC BEB43E09 35757BAD
4C06FC4A 0F3695E0 B00FBD30 4E8F36CE 7748F39C F9602650 7A1D2D48 DBC31237
AE3D63CE 593D31F5 62E4916F D20E30E8 30DC55C0 120FBD26 D2768DBC A67DDC34
5BDB66B1 E3
quit
license udi pid CISCO3845-MB sn FOC14421Q1Y
archive
log config
hidekeys
username admin privilege 15 secret 5 $1$Zf7j$P93opukmmEBIioVpjmHB3.
redundancy
interface Loopback10
ip address 192.168.2.1 255.255.255.0
interface Tunnel1
ip address 172.25.10.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 10
tunnel source GigabitEthernet0/1.1
tunnel mode gre multipoint
tunnel key 100
interface Tunnel2
ip address 172.25.11.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 20
tunnel source GigabitEthernet0/1.2
tunnel mode gre multipoint
interface Tunnel14
ip address 192.168.13.129 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.2.68.25
interface Tunnel18
ip address 192.168.13.137 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.9.160.236
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1.1
encapsulation dot1Q 10
ip address 10.9.160.25 255.255.255.0
interface GigabitEthernet0/1.2
encapsulation dot1Q 50
ip address 10.10.9.25 255.255.255.0
router eigrp 202
network 172.25.11.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
ip forward-protocol nd
ip http server
ip http secure-server
ip http path flash:/gui
ip route 10.2.68.0 255.255.255.0 10.9.160.1
ip route 10.10.0.0 255.255.0.0 10.10.9.1
ip route 10.64.164.30 255.255.255.255 10.9.160.1
ip route 192.168.14.0 255.255.255.0 192.168.13.130
ip route 192.168.17.0 255.255.255.0 Tunnel18
ip access-list standard REDIS1
permit 192.168.14.0
permit 192.168.17.0
route-map MYMAP1 permit 10
match ip address REDIS1
snmp-server community test RO
tftp-server flash:term11.default.loads
tftp-server flash:dkern9971.100609R2-9-0-3.sebn
tftp-server flash:kern9971.9-0-3.sebn
tftp-server flash:rootfs9971.9-0-3.sebn
tftp-server flash:sboot9971.111909R1-9-0-3.sebn
tftp-server flash:sip9971.9-0-3.loads
tftp-server flash:skern9971.022809R2-9-0-3.sebn
tftp-server flash:sccp11.9-0-2sr1s
tftp-server flash:SCCP11.9-1-1SR1S.loads
tftp-server flash:apps11.9-1-1TH1-16.sbn
tftp-server flash:cnu11.9-1-1TH1-16.sbn
tftp-server flash:cvm11sccp.9-1-1TH1-16.sbn
tftp-server flash:dsp11.9-1-1TH1-16.sbn
tftp-server flash:jar11sccp.9-1-1TH1-16.sbn
tftp-server flash:term06.default.loads
tftp-server flash:sip9971.9-1-1SR1.loads
tftp-server system:cme/sipphone
tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
tftp-server flash:Desktops/320x212x12/TN-Fountain.png
tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/Fountain.png
tftp-server flash:Desktops/320x212x12/CiscoLogo.png
tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:gui/admin_user.html
tftp-server flash:gui/admin_user.js
tftp-server flash:gui/CiscoLogo.gif
tftp-server flash:gui/Delete.gif
tftp-server flash:gui/dom.js
tftp-server flash:gui/downarrow.gif
tftp-server flash:gui/ephone_admin.html
tftp-server flash:gui/logohome.gif
tftp-server flash:gui/normal_user.html
tftp-server flash:gui/normal_user.js
tftp-server flash:gui/Plus.gif
tftp-server flash:gui/sxiconad.gif
tftp-server flash:gui/Tab.gif
tftp-server flash:gui/telephony_service.html
tftp-server flash:gui/uparrow.gif
tftp-server flash:gui/xml-test.html
tftp-server flash:gui/xml.template
tftp-server flash:ringtones/Analog1.raw
tftp-server flash:ringtones/Analog2.raw
tftp-server flash:ringtones/AreYouThere.raw
tftp-server flash:ringtones/AreYouThereF.raw
tftp-server flash:ringtones/Bass.raw
tftp-server flash:ringtones/CallBack.raw
tftp-server flash:ringtones/Chime.raw
tftp-server flash:ringtones/Classic1.raw
tftp-server flash:ringtones/Classic2.raw
tftp-server flash:ringtones/ClockShop.raw
tftp-server flash:ringtones/DistinctiveRingList.xml
tftp-server flash:ringtones/Drums1.raw
tftp-server flash:ringtones/Drums2.raw
tftp-server flash:ringtones/FilmScore.raw
tftp-server flash:ringtones/HarpSynth.raw
tftp-server flash:ringtones/Jamaica.raw
tftp-server flash:ringtones/KotoEffect.raw
tftp-server flash:ringtones/MusicBox.raw
tftp-server flash:ringtones/Piano1.raw
tftp-server flash:ringtones/Piano2.raw
tftp-server flash:ringtones/Pop.raw
tftp-server flash:ringtones/Pulse1.raw
tftp-server flash:ringtones/Ring1.raw
tftp-server flash:ringtones/Ring2.raw
tftp-server flash:ringtones/Ring3.raw
tftp-server flash:ringtones/Ring4.raw
tftp-server flash:ringtones/Ring5.raw
tftp-server flash:ringtones/Ring6.raw
tftp-server flash:ringtones/Ring7.raw
tftp-server flash:ringtones/RingList.xml
tftp-server flash:ringtones/Sax1.raw
tftp-server flash:ringtones/Sax2.raw
tftp-server flash:ringtones/Vibe.raw
tftp-server flash:APPS-1.2.1.SBN
tftp-server flash:SYS-1.2.1.SBN
tftp-server flash:GUI-1.2.1.SBN
tftp-server flash:CP7921G-1.2.1.LOADS
tftp-server flash:TNUX-1.2.1.SBN
tftp-server flash:TNUXR-1.2.1.SBN
tftp-server flash:WLAN-1.2.1.SBN
tftp-server flash:apps37sccp.1-2-1-0.bin
tftp-server flash:APPSH-1.3.1.SBN
tftp-server flash:GUIH-1.3.1.SBN
tftp-server flash:CP7925G-1.3.1.LOADS
tftp-server flash:SYSH-1.3.1.SBN
tftp-server flash:TNUXH-1.3.1.SBN
tftp-server flash:WLANH-1.3.1.SBN
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:Desktops/320x212x12/CampusNight.png
tftp-server flash:Desktops/320x212x12/CiscoFountain.png
tftp-server flash:Desktops/320x212x12/MorroRock.png
tftp-server flash:skern9971.022809R2-9-2-1.sebn
tftp-server flash:sip9971.9-2-1.loads
tftp-server flash:sboot9971.031610R1-9-2-1.sebn
tftp-server flash:rootfs9971.9-2-1.sebn
tftp-server flash:dkern9971.100609R2-9-2-1.sebn
tftp-server flash:kern9971.9-2-1.sebn
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
control-plane
mgcp profile default
dial-peer voice 1 voip
description connection-trough-PBX
destination-pattern 0....
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 voip
description K
destination-pattern 9T
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 5 voip
shutdown
destination-pattern *3709
session protocol sipv2
session target ipv4:192.168.13.130
session transport tcp
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
incoming called-number .
dial-peer voice 10 voip
gatekeeper
shutdown
telephony-service
em logout 0:0 0:0 0:0
max-ephones 262
max-dn 400
ip source-address 192.168.2.1 port 2000
load 7911 SCCP11.9-2-1S
max-conferences 12 gain -6
web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
ephone-template 1
softkeys connected Confrn Endcall Trnsfer Hold
keep-conference endcall
ephone-dn 1 dual-line
number 200
label test
name test
ephone-dn 2 dual-line
number 300
label Sepahbod
name Sepahbod
ephone-dn 4 dual-line
number 666
ephone-dn 5 dual-line
number 660
ephone-dn 6 dual-line
number 670
ephone-dn 7 dual-line
number 770
ephone-dn 8 dual-line
number 770
ephone-dn 9 dual-line
number 999
ephone 1
device-security-mode none
mac-address 18EF.639F.BCB0
keep-conference endcall
button 1:1
ephone 2
device-security-mode none
mac-address 0025.8418.B017
ephone-template 1
keep-conference endcall
button 1:2
ephone 3
device-security-mode none
mac-address F04D.A243.3154
keep-conference endcall
button 1:4
ephone 4
device-security-mode none
mac-address 6CF0.496A.69E9
button 1:4
ephone 5
device-security-mode none
mac-address 0015.E987.345F
keep-conference endcall
button 1:5
ephone 6
device-security-mode none
mac-address 0024.1DEA.614A
keep-conference endcall
button 1:6
ephone 9
device-security-mode none
mac-address 001D.7D4D.4DCB
button 1:9
line con 0
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end
and Voice Gateway connected two PBX system configuration
Current configuration : 3486 bytes
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Voice-GW
boot-start-marker
boot-end-marker
card type e1 0 2
no aaa new-model
network-clock-participate wic 2
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FHK1352F0E9
username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
redundancy
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-31
controller E1 0/2/1
interface Tunnel14
ip address 192.168.13.130 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface Tunnel17
ip address 192.168.13.134 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface FastEthernet0/0
ip address 192.168.14.252 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
ip address 10.2.68.25 255.255.255.0
duplex auto
speed auto
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.14.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 10.9.160.0 255.255.255.0 10.2.68.1
ip route 10.128.0.69 255.255.255.255 Tunnel14
ip route 192.168.2.1 255.255.255.255 192.168.13.129
ip route 192.168.17.0 255.255.255.0 Tunnel14
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
tftp-server flash:dsp11.9-2-1TH1-13.sbn
tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
tftp-server flash:cnu11.9-2-1TH1-13.sbn
tftp-server flash:apps11.9-2-1TH1-13.sbn
control-plane
voice-port 0/0/0
caller-id enable
voice-port 0/0/1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
dial-type pulse
disc_pi_off
output attenuation 1
echo-cancel coverage 32
timeouts call-disconnect 5
timeouts wait-release 1
timing hookflash-out 50
timing sup-disconnect 50
connection plar 600
caller-id enable
voice-port 0/0/3
caller-id enable
voice-port 0/2/0:15
mgcp profile default
dial-peer voice 1 pots
description connection-to-PBX
destination-pattern 0....
direct-inward-dial
port 0/2/0:15
forward-digits 4
dial-peer voice 10 voip
destination-pattern ...
session target ipv4:192.168.13.129
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 20 pots
description FXO-K
destination-pattern 9T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
direct-inward-dial
port 0/0/2
prefix 9
dial-peer voice 30 pots
description FXO-K2
destination-pattern 9T
direct-inward-dial
port 0/0/1
prefix 9
telephony-service
max-ephones 20
max-dn 100
ip source-address 192.168.14.252 port 2000
cnf-file location flash:
load 7911 term11.default.loads
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 770
line con 0
line aux 0
line 1/0 1/15
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
endHaving looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
I think you may be able to work around the problem by adding
" supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
reference
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
worth a try
Adam
Maybe you are looking for
-
Does Siebel license limits the Max task of Object Manager?
Hi, Gurus, We installed a Siebel environment on a Linux VM (Linux version el6.x86_64, Read Hat 4.4.6-4, the VM has 64g ram). The database (Oracle 11.2.0.0.1) and Oracle Http Server (ofm_webtier_linux_11.1.1.6.0_32) were also installed in the same VM.
-
Hello, I am creating JCO's and UME Data Source is ABAP. While Creating JCO's for SAP_R3_HumanResources as a Application data - in Security when i choose "useDefinedUser" and give the user name/passwd as SAPJSF jco is creating successfully and i am ab
-
BPC Database Recovery Instructions
Hi Experts, We are using BPC 7.5 Microsoft version with multiserver environment having BPC Application Server on one machine and using Shared Database server located on other machine, we have configured a database Instance for BPC databases on shared
-
XI R3 PIK for WebSEAL 6.0
The Supported Platforms document lists, under Other Tools support for the a Reverse Proxy Server "WebSEAL 6.0 (from IBM Tivoli Access Manager)". Can any of the techs provide more insight to this declaration? For instance, will a WebSEAL junction es
-
HT4623 software updates and there is no option to update in settings menu
My phone doesn't seem to be receiving software updates and doesn't give me the option in the settings menu.