Diversion of calls MX 922

The printer used for receipt of the fax this morning all phone calls are faxes, I need some advice, tried several things without success.

Returned by the procedures for setting up fax and now the problem is gone. It seems that somewhere along the line I have not properly put in place the portion of the printer fax.

Tags: Canon Printer

Similar Questions

After authentication redirecting before the user logged
Version of apex 4.1, custom authentication based on users of the Oracle database.

My goal is to allow a user to connect, and if their password is entered in the grace period (7 days before the due date, for example) they will be taken directly to a page to change their password, rather than the home page.

I have a page on page 101 (login) process to check if a password is about to expire (triggers a warning 28002). If this is the case, I store the warning message in a part of the application, "AI_LOGIN_WARNING".

In my custom authentication module, I have a post_authentication procedure: -.

------------------------------------------------
PROCEDURE post_authentication
IS

BEGIN

IF INSTR (: AI_LOGIN_WARNING, ' 28002') > 0
THEN
htmldb_application.g_unrecoverable_error: = TRUE;
owa_util. REDIRECT_URL ('f? p =' | v ('APP_ID') |) ': MY_ACCOUNT:' | v ('APP_SESSION'));
END IF;

END post_authentication;
-----------------------------------------------

where 'MY_ACCOUNT' is the alias of page for a page to allow the user to change his password.

There is a process on the MY_ACCOUNT page to change a password by calling a database procedure, passing in parameters p_lid = >: APP_USER, p_password =: P103_NEW_PASSWORD:

--------------------------------------------------------------------
PROCEDURE change_own_password
(p_lid IN VARCHAR2,
p_password IN VARCHAR2)
IS

e_not_current_user EXCEPTION;

BEGIN

IF p_lid! = V ('APP_USER')
THEN
RAISE e_not_current_user;
END IF;

EXECUTE IMMEDIATE 'modify user ' | V ('APP_USER') | ' identified by ' | p_password;

EXCEPTION

WHEN e_not_current_user
THEN
RAISE_APPLICATION_ERROR (-20010, ' you can't edit another user "password is).

END change_own_password;
---------------------------------------------------------------------

The first problem is that the app only redirects to the MY_ACCOUNT page if it is set to the Public.
If it is set to the Public, when I try to change my password I get the following error: ORA-01918: there is no such thing as 'NOBODY' user.
If I put APP_USER programmatically I get ORA-20010 ' you can not edit another user "password is error.

I seems as if I am redirected to the MY_ACCOUNT page before I was connected, even if the diversion is called in the later part of the AUTHENTICATION of my authentication scheme.

How can I make sure that Apex connect me before re - direct?

Thank you.
Hello

* Remove the message authentication process
* Creation of a branch in Page 0 or your homepage (before header) that redirects users 'Change password page' and set the condition so it runs only when it meets your requirement.

That should do the trick.

Kind regards
Hari

V - Portal of Vonage and Linksys RTP300 screw up the call diversion

I have a situation where a Linksys RTP300 legacy and vestigial is unconnected to all phones, but only to the LAN seems to screw up the call between Vonage numbers forwarding phones attached to a single Vonage V-portal which can be found on the same LAN. Incoming and outgoing calls are not affected. Remove the Linksys RTP300 seems to fix the problem. Can someone tell me why this would happen because the RTP300 is not connected either for all phones? And why the call forwarding is only affected (tent to transfer calls sends just calls to your voicemail, even if the other line is not picked up or not in use). The network is flat where the V-Portal of Vonage Linksys RTP300 and one small switch 8 ports are linked directly to the five ports integrated switch a firewall router, with the WAN port on the firewall router attached to a T1 CSU/DSU. I'm assuming (perhaps wrongly), if no phone is connected to the RTP300, there would be no SIP traffic either. Or would the RTP300 send SIP traffic even with no attached phone?

Hello.

These products are processed by the Cisco Small Business Support Community.

* If my post answered your question, please mark it as "acceptable Solution".

Thank you!

Patch presets - various method to call

Hello

at first sorry for my English, I will try to explain as simply as possible.

You know not how to appeal/load presets patches but without auxiliary send channels?

There are two reasons, I need this function:

(1) a lot of sends that is created by tasks presets are just reflections. I prefer another reverb send and plugins, and I don't want to use sends it created by logic built-in presets. But I like the preset at Channel Strip, so why I can't just call them only without sending.

(2) I have a lot of auxiliary entries nominated by myself, but not always use all of them. Then when I call some preset patches it creates send and aux usage first of all not used. And then it's very annoying when I in the channel of sending example reverb with entry called "parallel drum comp. So I need to go to the mixer, look for these new channels to send created, their diversion or delete if I don't want them at all.

It is very annoying, especially when I scroll down through library, and looking for interesting sounds, then I did not control what happens in my mixer, what's new is created, where routed... It makes my messy logical environment and - more importantly - my messy brain.

Can you help me somehow, how to solve?

Thanks in advance and regards to the best.

I forgot to mention patches presets to create auxiliary channel, which is adding up the track, and it's OK for me, I like this kind of summary of the channels. I have only one problem and dislikes chains self-creation of sends it. Adding titles are superb

Start the asynchronous call brutal Typedef Bug

There is a nasty bug which I think is the cause of many anomalies weird I see with the events of the user, like where some get fired and if I probe the refnum of the event on a VI that was launched using the asynchronous call node start I get some weird value for reference as 8450 or 5500 instead of some great typical integer. It is not also match the value that I get when I initialize the reference. This happens only intermittently, but I can reproduce the bug that I see on a smaller scale to a certain extent. This is not exactly the same as what I see in my current project, but I guarantee you both are related. Also, I'm pretty confident that this has to do with the help of LVlibs as well.

So... to reproduce some questions:

Unzip the attached code and open the project

Open Main.vi. It is hard to see because it's pink, but notice the point of constraint on the node to call asynchronous start. This is provided at this point because I have a cluster of non-typedef in the connector pane, but a TD cluster plugged into it.

Now open AsyncCall.vi

Drag the eventcluster.ctl of the project on the façade of the asynccall.vi

CTRL + x on the typedef cluster that has placed you on the front panel

Select the non-typedef cluster by clicking it

CTRL + v to replace the TD not cluster with the cluster of TD and save

Return to main.vi, you will notice that the point of constraint does not go far.

Open context-sensitive help and notice that the ctrl types match, but it's as if LV does not recognize it on the beginning of the node of the asynchronous call.

Remove the node from asynchronous start call, then replace it. The cluster to the top wire. Voila, no point of constraint.

Second question - same result but different method to get there.

Now that you have components of connector typedef stress points and no more because you've taken the first steps of this 'exercise', remove the EventCluster.ctl from the library and record.

WOAH, look the points of back strain, because node call asynchronous start still referencing the typedef cluster that he thinks that should be in the library. This can be seen by removing the cluster on main.vi and then right-click on the node to call asynchronous start on the side of the connector and creating a new constant of cluster

It is creates a greyed out of control! Why? Well, we will reopen the context-sensitive help. Whadda you know, it's always looking for the control in Bug.lvlib that no longer exists.

Now, the question that I'll have in my complete project that I can't post and can not reproduce on a smaller scale updates the typedef causes the dot of coercion. Otherwise I can't update my typedef cluster that contains all my events without going and replacing EVERY SINGLE launch async call node EVERY time I have add a new event.

Major problem.

Please let me know if these steps to reproduce were not clear or you have difficulties to reproduce the problem. I use LV2013 SP1. I opened the project in 2014 to see if it has been fixed in a later version, but I saw the same thing.

I can repro with measurements of @GregFreeman and also confirm that I saw this same issue at least since the LV2012, but they have not reported it having not been able to provide a minimum test (thanks, @GregFreeman!) scenario

For the record, it seems that the bug here, it is the spread of type sometimes makes an incorrect assumption / optimization as to if the conpane of the start the asynchronous call node must be updated when the source changes.

A more obvious change - say, add/remove an entry, reverse order, or change data types altogether - always seems to spread properly.

Incorrect optimization seems to be a terminal retains the same type of database, but transforms the type definitions - or, if the type definition is re-related or related outside a library owner.

@GregFreeman watch the bug goes from non-typedef typedef, but it's actually worse in the other direction - when a link to a missing file is maintained.

Call the asynchronous starting node seems to maintain a list of links that is distinct from that of the VI, and this list of links separated, this is what seems to not be properly invalidated. For example, in the screenshot, I illustrated example of Greg that the node generates no error in the compiler even after parenthood and rename the Typedef...

... even when we "Create Constant" on this terminal incriminated with list obsolete links, we get a compilation error. Since then, the grayed out type highlighted in the contextual help cannot be found, because 'Bug.lvlib:EventCluster.ctl' no longer exists, but the list of links separated from this node was not notified:

It is worth noting that "Bug.lvlib:EventCluster.ctl" does not appear in the list of links of the VI at this stage.

Often, no compiler error is generated after this failure occurs and as Greg reports, you could end up with undefined behavior (e.g. suspicious Refnums and events that seem to not fire not) (and I'll add it to this list a hearty portion of DAborts with diversion total number of messages).

In addition, you * could * receive errors of cryptic linker for generations, but maybe not (the above screenshot, you'll notice I added two builds, neither of which seems to have a problem of building). (It seems that the broken link is travel with the distribution of the source, even if 'Disconnect the definitions of Type' is selected during the build process. That is why I believe anecdotally that node maintains a list of link separately the list of VI, and it's maybe part of the problem).

It is noted that during this refactor (de-parent and rename) all screws and control remained open and in memory and all files have been saved. No funny business where LabVIEW would be unable to update links in a file that was not in memory.

Another note - in the original example, all source files have been unifiles, and I can add anecdotal report this bug is much more insidious when separate compiled Code is active on the source files. In this case, the source may appear to be perfect - no point of constraint, no link expired - but the code that is currently running can be broken. In other words, what you see is not what you get, which makes debugging impossible. (This bug in particular is one of the few who makes "Cache of compiled clear objects" become a normal procedure controlled throughout the application development)

Anyway, I wanted to draw attention to this issue, given that this thread is not yet associated with a CAR and it's a serious bug that generates a behavior undefined performance caused by a fairly normal refactor now has a well-characterized small repro case.

Unconditional call forward on SPA9000

Hi all, I am trying to set with CFD for all calls on SPA9000 SPA922 telephone. I have one-way voice problem.
I did extensive debugging with traces SIP and I think I found a problem.

Here is a part of SIP trace, step 2: (I deleted the IP and all the unnecessary parts)

16:19:50.67257 - C5CM - xx.xx.xx.xx - S = 5060 - D = 5060 - TOS = 0 x 68.
INVITE sip:[email protected] SIP/2.0

Referred-by:; REF is 103 Diversion:;. reason = unconditional
User-Agent: Linksys/SPA9000-6.1.5

v = 0
o = Clarent 162066 162067 IN IP4 zz.zz.zz.zz
s = Clarent C5CM
c = IN IP4 222.173.190.239
t = 0 0
m = audio 48879 RTP/AVP 0 8 101 100 102 103 104 105 t38
a = rtpmap:0 PCMU/8000
a = rtpmap:8 PCMA/8000
a rtpmap:101 telephone-event/8000 =
a = fmtp:101 0-15
a = rtpmap:100 AC100/8000
a = rtpmap:102 AC102/8000
a = rtpmap:103 AC103/8000
a = rtpmap:104 AC104/8000
a = rtpmap:105 AC105/8000

As I see it, the problem is in this line:
c = IN IP4 222.173.190.239
Here must be IP address: zz.zz.zz.zz

Whence came this IP address?
It is not present in the config.

I tried with firmware version 5.2.5 and 6.1.5 SPA9000. They both show the same behavior. I would go to the old versions and test them because I know that it worked in the past, but it seems that SPA9000 is not possible to downgrade.

Kind regards
Miro

Transfer calls to SIP phones problem

Hello.

When a user with a Cisco 3905 tries to transfer a call to another IP phone, the call successfully diversion; However, the new beneficiary do not hear anything. If he speaks, the original caller can listen to it, but the recipient cannot.

Is there anything us missing in our setup in the router of the CMF? All I could do to fix this?

I attach the run-config and a message from ccsip debugging I extracted by a call that had been transferred.

Whatever that either would be appreciated. Thank you!.

Just to give more information on why this solves the problem, some devices/carriers do not manage the call for sendonly to switching sendrcv after a hold event and continue to stay in rcvonly. This workaround lets the other side think that the call is in sendrcv mode all the time.

CISCO SPA122 busy for incoming calls

Hello

I have a problem with some SPA122.

They are registered, they can dial a number, but they can't receive calls.

In our cause to hang up asterisk SoftSwitch are Busy (0) or Failed (20).

But ATAs are saved.

I've noticed 2 things:
- ATA, registered with a private IP address don't have questions while some ata registered with a public IP address have problems. All the ata, with the address private or public IP had the same configuration.
- In these clients with ata busy (0) we reset the data and we have the same configuration. In this way, they still run.
Our SoftSwitch Asterisk is MOR by Kolmisoft. They said "it happens when ATA does not respond to requests to qualify. He is redirecting request type. Please check ATA configuration if there is no road or forward together on this. »

We have reproduced the call and checked the trace of the SIP.

Something is not ok with ATA itself. It returns the following:

302 moved temporarily SIP/2.0
To: <> [email protected] / * /: 5060 >; tag = d299d2248086f464i0
From: "37052058393" <> [email protected] / * />; tag = as088ffdb4
Call ID: [email protected]/ * /: 5060
CSeq: INVITE 102
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060; branch = z9hG4bK027fee05
Contact: <> [email protected] / * />
Diversion: <> [email protected] / * />; reason = unconditional
Server: Cisco/SPA122-1.1.0(011)
Content-Length: 0

He is redirecting request type. Please check ATA configuration if there is no road or forward together on this.

We use firmware 1.3.5 and attached our config file.

Please, can you help me with this problem?
```
 00xxxxxxxxx748
```
```
 00xxxxxxxxx621
```
All calls are transferred elsewhere as shown in the response to INVITE captured on top of thread. I see neither occupied nor error response sent to SPA122. So, who the BUSY signal / ERROR you mentioned? It seems that it's Asterisk and SPA122 is guilty here...

Note that the configuration of the joint disclose passwords SNMP to the public, so it's time to change.

Rate of useful advice. It will help others to find solutions.

Call Cisco UBE is not received endpoint

Hello

I am tryng to install what UBE between two GUY, the idea is that R1 will send the call to the CUBE that will send the call to R2 (following a pattern). Earlier today, when the call is sent to R1 call is received for CUBE but resemble the call is not to send to R2, even when leaving to R2 dial-peer is matched. I installed a phone in CUBE and I can can appeal to R2 using the same outgoing dial peer match R2, so I don't know why CUBE isn't a R1 to R2 call diversion

Schema:

H323 H323

R1 ----------------> CUBE -----------------> R2

DN = 2... DN = 3...

Debugging

======

Call is received of R2 and outgoing dial-Exchange R2 files is put in correspondence, but call for R2

GK #debug voip dialpeer

VoIP dialpeer default debug is on

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Number = 2003, called number = 3001, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1999

* Jun 29 12:56:00.927: //-1/C7785C1D8183/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Number = 2003, called number = 3001, Voice-Interface = 0 x 0.

Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

Peer Type Info = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpAssociateIncomingPeerCore:

Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1999

* Jun 29 12:56:00.931: //-1/C7785C1D8183/DPM/dpMatchSafModulePlugin:

dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

* Jun 29 12:56:00.931: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Number =, called number = 3001, Peer Info Type = DIALPEER_INFO_SPEECH

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Associate the rule of = DP_MATCH_DEST; Called number = 3001

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result = Success (0) after DP_MATCH_DEST

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

dialstring = 3001, saf_enabled = 0, saf_dndb_lookup = 1, dp_result = 0

* Jun 29 12:56:00.935: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result = Success (0)

The outgoing dial matched host list:

1: Tag dial-peer = 2000

CUBE see the race

============

GK #show run
Building configuration...

Current configuration: 2628 bytes
!
! Last modification of the configuration at 12:31:19 UTC Friday, June 29, 2012
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
no password encryption service
!
hostname GK
!
boot-start-marker
start the flash c2800nm-adventerprisek9_ivs_li - mz.151 - 4.M3.bin system
boot-end-marker
!
!
enable secret 5 $1$ szeU$ enWyM69bTodk0Aiwyxw5R.
!
No aaa new-model
!
!
dot11 syslog
IP source-route
!
!
IP cef
!
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
!
!
!
voip phone service
h323 connections allow h323
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
H323
SIP
!
!
!
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1225F3KT
!
redundancy
!
!
controller LAN 0/0/0
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
192.168.4.1 IP address 255.255.255.0
automatic duplex
automatic speed
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/2/0
no ip address
Shutdown
no fair queue
2000000 clock frequency
!
interface Serial0/2/1
no ip address
Shutdown
2000000 clock frequency
!
!
Router eigrp 1
192.168.4.0 network
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
NLS RESP-timeout 1
CPD cr id 1
!
!
!
!
!
!
control plan
!
!
Voice-port 1/0/0
!
Voice-port 1/0/1
!
!
!
profile MGCP default
!
SCCP local FastEthernet0/0
SCCP ccm 192.168.4.1 ID 1 version4.0
SCCP
!
SCCP ccm Group 1
associate the ccm 1 priority 1
the associated profile 1 registry GK-VOUGEOT
!
transcode dspfarm profile 1
Codec g729r8
Codec g729br8
Codec g729ar8
Codec g729abr8
Codec g711alaw
Codec g711ulaw
maximum sessions 3
associate the PCRS application
!
Dial-peer voice 2000 voip
destination-model 3...
session target ipv4:192.168.3.2
Transparent codec
!
voice pots Dial-peer 5001
destination-model 5001
port 1/0/0
!
Dial-peer voice voip 1999
session target ipv4:192.168.3.1
incoming called-number 3...
!
!
!
!
access controller
area local zone1 micasa.com 192.168.4.1 intrazone enable
area local micasa.com enable-intrazone zone2
zone prefix zone1 2... GW-priority 10 CME - Lab
zone prefix zone2 3... GW-priority 10 CME-Legacy
GW-type-prefix 1 #* by default-technology
no downtime
!
!
phone service
5 units of sdspfarm
sdspfarm tag 1 GK-VOUGEOT
Max-joined 4
Max - dn 4
IP source-address 192.168.4.1 port 2000
MAX conferences 8-6 win
transfer full-consult system
!
!
!
Line con 0
password leoleo
opening of session
line to 0
line vty 0 4
password leoleo
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
end

GK #.

R1 show race

=========

CME - Lab #show run
Building configuration...

Current configuration: 7806 bytes
!
! Last modification of the configuration at 08:23:06 LAPAZ Friday, June 29, 2012
! NVRAM config update at 13:13:31 LAPAZ Thursday, June 28, 2012
!
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
encryption password service
!
CME - Lab host name
!
boot-start-marker
start the flash c2800nm-adventerprisek9 - mz.151 - 3.T3.bin system
boot-end-marker
!
!
type 0 2 t1 card
enable secret 5 $1$jJ.7$b/Yaq0M36fgZSrYpCK5LD/
!
No aaa new-model
!
clock timezone LAPAZ-4 0
network-clock-participate wic 2
!
dot11 syslog
IP source-route
!
!
IP cef
!
DHCP excluded-address 192.168.1.1 IP 192.168.1.20
DHCP excluded-address IP 192.168.2.1 192.168.2.20
DHCP excluded-address IP 192.168.3.1 192.168.3.20
DHCP excluded-address IP 192.168.6.1 192.168.6.20
!
IP dhcp DATA pool
network 192.168.1.0 255.255.255.0
option 150 ip 192.168.2.1
default router 192.168.1.1
!
pool IP dhcp VOZ - CME
network 192.168.2.0 255.255.255.0
option 150 ip 192.168.2.1
default router 192.168.2.1
!
VOZ-CUCM dhcp IP pool
network 192.168.6.0 255.255.255.0
option 150 ip 192.168.6.10
router by default - 192.168.6.1
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
primary-qsig ISDN switch type
!
!
!
voip phone service
list of approved IP addresses
IPv4 192.168.3.2
SIP
!
voice class codec 1
g711ulaw codec preference 1
codec preference 2 g729r8
g729br8 preferably 3 codec
g711alaw preferably 4 codec
!
!
!
!
translation of the voice-rule 1
rule 1 3... / / 1809544\0 /.
!
voice translation-rule 100
rule 1 /2/ /1002/
!
voice translation-rule 101
rule 1 /1002/ /2/
!
!
voice translation-profile SITE-CODE-CallerID
definition of 100 calls
!
SITE-CODE-DNIS voice translation-profile
translate called 101
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1225F3GX
Archives
The config log
hidekeys
!
redundancy
!
!
controller T1 2/0/0
the clock source internal
long CableLength 0dB
time intervals PRI - Group 1-2: 24
!
!
class-map correspondence-any WEB browsing
http protocol game
secure http protocol game
class-map matching VOICE
match Protocol rtp
!
!
OUTBOUND-POLICY policy-map
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
no ip address
automatic duplex
automatic speed
!
interface FastEthernet0/0.10
encapsulation dot1Q 10
IP 192.168.2.1 255.255.255.0
H323-gateway voip interface
H323-gateway voip id zone1 ipaddr 192.168.4.1 1719
H323-gateway voip h323-id CME - Lab
port of link voip H323-gateway 192.168.2.1
!
interface FastEthernet0/0.11
encapsulation dot1Q 11
192.168.6.1 IP address 255.255.255.0
!
interface FastEthernet0/0.50
encapsulation dot1Q 50
IP 192.168.1.1 255.255.255.0
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/0/0
address 192.168.3.1 IP 255.255.255.0
encapsulation ppp
64000 clock frequency
!
interface Serial0/0/1
no ip address
Shutdown
2000000 clock frequency
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
primary-qsig ISDN switch type
ISDN timer T310 120000
ISDN-emulate network protocol
ISDN incoming-voice
No cdp enable
!
!
Router eigrp 1
network 192.168.1.0
network 192.168.2.0
192.168.3.0 network
network 192.168.6.0
Auto-resume
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
exploitation forest esm config
!
!
!
!
!
flash:/PHONE/6921_6941_6961/Sccp/cmterm-69xx-sccp.9-2-1-0.tar cmterm-69xx-sccp.9-2-1-0.tar alias TFTP server
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/BOOT69xx.0-0-0-14.zz.sgn BOOT69xx.0 - 0-0 - 14.zz.sgn
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/SP69xx.0-0-0-8.zz.sgn SP69xx.0 - 0-0 - 8.zz.sgn
alias of flash:/PHONE/6921_6941_6961/Sccp/SCCP69xx.9-2-1-0.loads SCCP69xx.9 - 2-1 TFTP server - 0.loads
alias of flash:/PHONE/6921_6941_6961/Sccp/SCCP69xx.9-2-1-0.zz.sgn SCCP69xx.9 - 2-1 TFTP server - 0.zz.sgn
Server TFTP flash:/PHONE/7942_7962/Sccp/apps42.9-2-1TH1-13.sbn alias apps42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/cnu42.9-2-1TH1-13.sbn alias cnu42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/cvm42sccp.9-2-1TH1-13.sbn alias cvm42sccp.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/dsp42.9-2-1TH1-13.sbn alias dsp42.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7942_7962/Sccp/jar42sccp.9-2-1TH1-13.sbn alias jar42sccp.9 - 2-1TH1 - 13.sbn
alias server TFTP flash:/PHONE/7942_7962/Sccp/SCCP42.9-2-1S.loads SCCP42.9 - 2 - 1 S .loads
flash:/PHONE/7942_7962/Sccp/term42.default.loads alias term42.default.loads TFTP server
flash:/PHONE/7942_7962/Sccp/term62.default.loads alias term62.default.loads TFTP server
Server TFTP flash:/PHONE/7945_7965/Sccp/cmterm-7945_7965-sccp.9-2-1.tar alias cmterm-7945_7965 - sccp.9 - 2 - 1.tar
Server TFTP flash:/PHONE/7945_7965/Sccp/apps45.9-2-1TH1-13.sbn alias apps45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/cnu45.9-2-1TH1-13.sbn alias cnu45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/cvm45sccp.9-2-1TH1-13.sbn alias cvm45sccp.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/dsp45.9-2-1TH1-13.sbn alias dsp45.9 - 2-1TH1 - 13.sbn
Server TFTP flash:/PHONE/7945_7965/Sccp/jar45sccp.9-2-1TH1-13.sbn alias jar45sccp.9 - 2-1TH1 - 13.sbn
alias server TFTP flash:/PHONE/7945_7965/Sccp/SCCP45.9-2-1S.loads SCCP45.9 - 2 - 1 S .loads
flash:/PHONE/7945_7965/Sccp/term45.default.loads alias term45.default.loads TFTP server
flash:/PHONE/7945_7965/Sccp/term65.default.loads alias term65.default.loads TFTP server
alias of server TFTP flash:/PHONE/6921_6941_6961/Sccp/DSP69xx.0-0-0-8.zz.sgn DSP69xx.0 - 0-0 - 8.zz.sgn
!
control plan
!
!
voice-port 0/2/0:23
!
Voice-port 1/0/0
signal groundStart
name of station id Leo Salcie
activation of the caller ID
!
Voice-port 1/0/1
!
voice-port 3/0/0
!
voice-port 1/0/3
!
voice-port 0/3/2
!
voice-port 0/3/3
!
CCM-Manager mgcp
CCM-Manager config server 192.168.6.10
!
MGCP
type of service mgcp MGCP call-agent 192.168.6.10 version 0.1
MGCP ecm t38 fax
!
profile MGCP default
!
SCCP local FastEthernet0/0.10
SCCP ccm 192.168.2.1 ID 1 priority 1 version 4.1
SCCP
!
SCCP ccm Group 1
associate the ccm 1 priority 1
the associated profile 1 save CME-VOUGEOT
!
transcode dspfarm profile 1
Codec g711ulaw
Codec g711alaw
Codec g729ar8
Codec g729abr8
Codec g729br8
Codec g723r53
maximum sessions 4
associate the PCRS application
!
Dial-peer cor custom
name 911call
name localcall
name ldcall
!
!
Dial-peer cor list LD
Member 911call
Member localcall
Member ldcall
!
Dial-peer cor list localcall
Member localcall
!
Dial-peer cor list ldcall
Member ldcall
!
!
Dial-peer voice 1000 voip
destination-model 3...
session target ipv4:192.168.4.1
!
!
!
!
phone service
5 units of sdspfarm
sessions to transcode sdspfarm 4
sdspfarm tag 1 GUY-VOUGEOT
Logout em 0:0 0:0 0:0
Max-joined 12
Max - dn 12
Source-address 192.168.2.1 IP port 2000
Auto assign 1 to 6
load SCCP45.9 - 2-1 7965 s
MAX conferences 8-6 win
transfer full-consult system
create the file cnf version-stamp 7960 21 June 2012 09:37:02
!
!
ePhone-dn 1
number 2001
!
!
ePhone-dn 2
number 2002
!
!
ePhone-dn 3
2003 Edition
!
!
ePhone-dn 4
issue 2004
!
!
ePhone-dn 5
issue 2005
!
!
ePhone-dn 6
Number of 2006
!
!
ePhone-dn 9 double line
number 9999
!
!
ePhone 1
security-mode device no
503D Mac address. E5E9.2001
Max-calls-by button-2
type of 6921
button 1:1
!
!
!
ePhone 2
security-mode device no
A 64-0 Mac address. E715. F5B2
type of 7965
key 1:2
!
!
!
ePhone 5
security-mode device no
503D Mac address. E52F. C3E4
Max-calls-by button-2
key 1:3
!
!
!
ePhone 9
security-mode device no
00F0.1D00.0081 Mac address
!
!
!
ePhone 99
security-mode device no
address Mac AABB. CC11.2233
key 1:9
!
!
!
!
Line con 0
line to 0
line vty 0 4
password 7 151E0E03082F24
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
NTP master 3
Server NTP 192.168.2.1
end

R2, see the race

==========

CME-Legacy #show run
Building configuration...

Current configuration: 3189 bytes
!
! Last modification of the configuration at 22:44:24 UTC Thursday, June 28, 2012
!
version 15.1
horodateurs service debug datetime msec
Log service timestamps datetime msec
encryption password service
!
CME-Legacy host name
!
boot-start-marker
boot-end-marker
!
!
type 0 3 t1 card
enable secret 5 $1$ $0iFd Um7r8JhKS6fMrEpFH64rB0
!
No aaa new-model
!
network-clock-participate wic 3
!
dot11 syslog
IP source-route
!
!
IP cef
!
!
!
no ip domain search
No ipv6 cef
!
Authenticated MultiLink bundle-name Panel
!
!
!
!
primary-qsig ISDN switch type
!
!
!
voip phone service
list of approved IP addresses
IPv4 192.168.2.1
IPv4 192.168.3.1
SIP
!
voice class codec 1
g711ulaw codec preference 1
codec preference 2 g729r8
g729br8 preferably 3 codec
g711alaw preferably 4 codec
!
!
!
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
Crypto pki token removal timeout default 0
!
!
!
!
license udi pid CISCO2811 sn FHK1246F3KE
Archives
The config log
hidekeys
!
redundancy
!
!
controller LAN 0/0/0
!
controller T1 3/0/0
the clock source internal
long CableLength 0dB
time intervals PRI - Group 1-2: 24
!
!
!
!
!
!
!
!
!
interface Loopback0
IP 192.168.100.1 address 255.255.255.0
!
interface FastEthernet0/0
IP 192.168.4.2 255.255.255.0
automatic duplex
automatic speed
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface Serial0/2/0
IP 192.168.3.2 255.255.255.0
encapsulation ppp
!
interface Serial0/2/1
no ip address
Shutdown
2000000 clock frequency
!
interface Serial0/3/0:23
no ip address
encapsulation hdlc
primary-qsig ISDN switch type
ISDN incoming-voice
No cdp enable
!
!
Router eigrp 1
192.168.3.0 network
192.168.4.0 network
network 192.168.100.0
Auto-resume
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
exploitation forest esm config
!
!
!
!
!
!
control plan
!
!
voice-port 3/0/0:23
!
Voice-port 1/0/0
!
Voice-port 1/0/1
!
CCM-Manager mgcp
CCM-Manager config server 192.168.6.10
!
MGCP
type of service mgcp MGCP call-agent 192.168.6.10 version 0.1
MGCP ecm t38 fax
!
profile MGCP default
!
SCCP local Loopback0
SCCP ccm 192.168.100.1 ID 1 priority 1 version 7.0
SCCP
!
SCCP ccm Group 1
bind the interface Loopback0
associate the ccm 1 priority 1
the associated profile 1 save CME-VOUGEOT
!
transcode dspfarm profile 1
Codec g711ulaw
Codec g711alaw
Codec g729ar8
Codec g729abr8
Codec g729r8
Codec g723r53
maximum sessions 2
associate the PCRS application
!
voice pots Dial-peer 3001
destination-model 3001
port 1/0/0
!
Dial-peer voice 1000 voip
destination-model 2...
session target ipv4:192.168.3.1
!
voice pots Dial-peer 3002
destination-model 3002
port 0/1/1
!
Dial-peer voice voip 5000
session target ipv4:192.168.4.1
incoming called-number 3...
!
!
!
!
phone service
5 units of sdspfarm
sessions to transcode sdspfarm 2
sdspfarm tag 1 GUY-VOUGEOT
Logout em 0:0 0:0 0:0
Max-joined 4
Max - dn 4
IP source-address 192.168.100.1 port 2000
MAX conferences 8-6 win
transfer full-consult system
create a cnf-files version-stamp 1 January 2002 00:00:00
!
!
!
Line con 0
password 7 09404B 06150018
opening of session
line to 0
line vty 0 4
password 7 11051C0A1B1704
opening of session
transport of entry all
!
Scheduler allocate 20000 1000
end

CME-Legacy #.

Any help would be appreciated

Concerning

Technically, this isn't a CUBE configuration. The calls come in via a VOIP dial-peer and they are sent to a pots dial-peer.

Can you get debug voip ccapi inout of R2. We have to see what happens when the call comes in on this router and why the call disconnects.

If you can put the debugging in a text file and join here it will be better to post directly here

Please note the useful messages

"For the love of God is broader than the extent of the human spirit and the heart of the eternal is most wonderfully kind."

Dell AIO 922 printer installed, windows 7 will not print to it

connected my Dell AIO 922 printer via the USB port. Windows 7 found installed the rpinter and said that he installed the drivers. It appears in the Printers window and as installed devices. When I try to print it says "the system cannot find the file specified."

I have the same problem and when I tried to download a driver for my Dell AIO 922 printer it says that this printer is not compatible with Windows 7. Thus, it seems that we are on a printer. This is the Web site from the list of compatible printers for Windows 7 Dell. The AIO 922 figure on the list is not compatible. Here is the website http://support.dell.com/support/topics/global.aspx/support/dsn/document?c=us&cs=19&docid=721171C9910C8085E040AE0AB7E1537B&l=en&s=dhs

Also, I called Dell and they said that from now on, anyone who upgrades to Windows 7 with a printer Dell AIO 922 is out of luck and there is nothing they can do because they do not have a printer driver for Windows 7.

I have, however, found on another site... I have not yet tried or even looked at in depth, but maybe this will help. Look at the last comment.
http://social.technet.Microsoft.com/forums/en/w7itprohardware/thread/aad4c406-7f10-4879-9fca-ac5c7fdc9e1b

Start a video call is grayed out in Live messenger

With the help of Live Messenger 2011 (build 15.4.3502.922). To start a video call is greyed out, but the camera is seen if I go through the audio and video Setup. I can also receive a video call by someone else and the camera works fine. Using the Microsoft Lifecam Cinema with the current drivers. Imagine what we could be... If we could imagine. Daniel A. Galant

Hi Daniel,.

The question you have posted is related to Windows Live Messenger and would be better suited to the Windows Live community. Please visit the link below to find a community that will provide the support you want.

http://windowslivehelp.com/solutions.aspx?CategoryID=07042916-d29c-4cac-8657-e700a4866c47

It will be useful.

Thank you and best regards,

Srinivas

Microsoft Answers Support Engineer

Visit our Microsoft answers feedback Forum and let us know what you think

Fake phone call from 424-253-4445

Hello James,.

I would really appreciate your insight to a phone call that I received concerning malicious information, I was told, on my computer.

Naturally, I am very curious when I receive a call on my phone number at home of an unknown number. The woman I spoke to said she is a representative for windows and asked me to go on my computer to erase the harmful activity. I explained my concern about my concern, thinking it might be a scam of some kind.

She gave me this number 424-253-4445, reassure me is it windows technical support. I noticed that I have more 2 500 kinds of activity which is harmful for my system.

Can you please tell me if you know something of this type of service through windows team support?

Thank you

Beth

Separated from: https://answers.Microsoft.com/en-us/Windows/Forum/windows_8-windows_install/Windows-81-Preview-questions/ace82704-1032-4539-95f2-7d1d1f25a43c

It's absolutely a scam! No company like Microsoft, Apple, Dell or anyone else call you for any reason unless you call them.

There's a certain phone scams going on right now and if you have the caller ID, just look at the display and if you see "Microsoft" or if it is "Private" and that the appellant represents a computer company, ignore the call.

There are web sites that publish the fake pop-up messages indicating that your computer has problems and a scanner. NEVER, NEVER, NEVER, cela!

Windows 8/8.1 already has an anti-virus/anti-spyware program (Windows Defender) If you are not using anything else and it will cover you. I tested Microsoft Security Essentials (now called Windows Defender in Windows 8) for about three years and sometimes, if you get a web page infected, the web page is displayed as an empty and your web browser automatically updates your home page with a warning displayed. It does not work.

If you get a pop-up message that stays on the screen as mentioned previously, DO NOT click on it. Press CTRL + ALT + DEL and bring up the Task Manager and close all programs that appear as running. You can see the web browser nearby. Then you can close the Task Manager and open IE and if he asks me to restore your last session, say NO and go to the home page. You'll be fine right there. The pop ups are called "diversions" and locks your system.

Windows 8/8.1 will clean and optimize itself if you let on and intact for about 5 minutes. You will see the 'Flag' icon change a flag with a clock which means that it runs Windows updates, disk cleanup, Disk Defrag, and update/analysis of Windows Defender itself. You don't have to do anything.

New iPhone than 7 to 8 hours time to call to life

I got a new iPhone 7 more, ordered through Verizon, a few days ago. Yesterday, I noticed that she had 8 hours and 51 minutes of life call time, which is impossible.

This phone has replaced my iPhone 6

I have not configured this phone via an iPhone backup 6. I plug it into iTunes once for synchronization of music and iTunes did ask if I want to set it up as a new phone. I said yes.

I use the same iTunes/iCloud account that I used on my old phone, however.

My use in line with Verizon for this month is only 113 minutes, so this number is clearly wrong.

Still, I have the old phone and looked at the number of hours of calls from life. It's much more than 24 hours, the new phone has not inherit the old phone number.

DIDS, I get a phone used or defective?

Very strange... no one here can say what has happened. I go back to Verizon and ask them. If it was me, I insist on getting a new phone... that one has been used somewhere.

Barry

IPhone 7 people cannot hear me very well on voice calls

People can't hear me very well when talking on IPhone 7. I can hear them perfectly but they say that my volume is very low

Hi there johnty53!

Welcome to the communities of Apple Support and thank you for bringing your question about the quality of the audio calls on your iPhone 7. Being able to be heard on the calls is certainly of the highest importance so I'm happy to help you today.

I understand that you have a brand new iPhone, but I have some things that we can test first to make sure that nothing hinders the microphones. Also, you'll want to make sure that 10 iOS is updated, updates may have software patches to improve the functionality of your iPhone.

Update your iPhone:
1. Open the settings app.
2. Settings > general > software update.
Please also ensure that your carrier settings are also updated:
1. Make sure that your device is connected to a Wi-Fi or cellular network.
2. Tap Settings > general > topic. If an update is available, you will see an option to update your carrier settings.
Get help with microphones on your iPhone, iPad and iPod touch

Take care!

updated Mac and now I am unable to answer the phone calls of the Mac

Yesterday I updated my Mac to the new operating system. Since then, I could not pick up calls from the Mac as I used to.

Can someone tell me how to solve this problem?

Hi jlfrisella!

I see that you are unable to use the calls feature cell phones iPhone on your Mac since the update to Mac OS Sierra. I can understand wanting this function runs on the right, and I'm happy to help you. Please see requirements and instructions in the following articles.

Continuity of use to connect your Mac, iPhone, iPad, iPod touch and Apple Watch

Thank you for using communities of Apple Support.

See you soon!

Diversion of calls MX 922

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