Dial up earthlink - incoming calls wake up computer
Earthlink dial up user - incoming calls wake up my computer - why is this happening - can it be fixed?
Also modem dials, says connecting then I only receive a long tone and disconnects. Message says no carrier detected check phone lines - phone company says ok - calls can be placed out and received in. Earthlink says no problem there. Any ideas? This happens sometimes when trying to connect and I have tried several access numbers (no busy signal) Please help!
Firefox 4 versions and later do not run on a PowerPC Mac. '''Firefox 4 and later require at least OS X 10.5''' and an Intel Mac.
http://www.mozilla.org/en-US/firefox/8.0/system-requirements/
Firefox(supported) 3.6.24 (Mac OS X 10.4 and later with PowerPC's) can be found here(but you have it and you are supported):
http://www.mozilla.com/en-US/firefox/all-older.html
thank you
Please mark "Solved" the answer that really solve the problem, to help others with a similar problem.
Similar Messages
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Using answer-address in dial-peer when it has incoming called number
Hello,
We would like to use an inbound dial peer that will match by the Calling Number using answer-address.
In our current configuration we already have inbound dial peers with "incoming called number" for Fax2Mail services.
We want to use the calling number for matching to a different dial peer, without the "fax detect" service.
As we notice, as soon as he matching the incoming called number, even if have identical dial peer with the same incoming called number, he stop the matching processes and the gateway ignore the answer-address.
Is there any way to match the dial peer by Calling Number even if he have the incoming called number field?
ThanksHello,
in the Inbound Dial-peer matching process, the 'incoming called-number' has the highest priority over asnwer- address & destination pattern. So if you want to match a particular inbound dial-peer based on calling number, add only asnwer-address and don't configure incoming-called number on the same dial-peer.
FYI: Understanding the dial-peer matching process
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html?referring_site=smartnavRD#topic3
//Suresh
Please rate all the useful posts. -
How to limit incoming call on B-ACD CME?
I have configure B-ACD on CME 4 and its working. But I have a problem with incoming call from outside (PSTN). I set queue-len for 30 queue. But on busy hour, some call hear only ring tone. No greeting tone or busy tone. When I check call session, its only 10 or 15 session.
How to limit incoming call, let say, for 10 calls only? The 11th call will receive busy tone.
Here are my config:
application
service queue flash:app-b-acd-2.1.2.2.tcl
param queue-len 30
param queue-manager-debugs 1
param number-of-hunt-grps 1
param aa-hunt2 0
service aa flash:app-b-acd-aa-2.1.2.2.tcl
paramspace english index 1
param number-of-hunt-grps 1
param handoff-string aa
param dial-by-extension-option 1
paramspace english language en
param max-time-vm-retry 1
param max-extension-length 7
param aa-pilot 2999
paramspace english location flash:
param second-greeting-time 30
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param max-time-call-retry 60
param voice-mail 0
param service-name queue
FYI: I have no voice mail. 'param voice-mail 0' is to send unanswered call to my operator.
Regards,
IwanHi Iwan,
To restrict the number of calls to the application you need to configure only the appropriate number (10 in your example) of dial-peers to point to the AA Pilot number. Have a look;
Set Up Incoming Dial Peers for AA Pilot Numbers
In this task, you associate dial peers for incoming calls with the AA service that you want them to use.
Cisco Unified CME B-ACD is available for outside calls through voice ports and trunks, for which dial peers must be set up. When you set up a dial peer, you use the service command to associate it with the name of the Cisco Unified CME B-ACD AA service that you want callers to that dial peer to reach.
Note You must configure a dial peer for each incoming DID voice port.
To determine how many ports or trunks you must have for your Cisco Unified CME B-ACD service, consider the following:
Total number of phones across all ephone hunt groups
Total number of slots in the queues across all queues
Total number of PSTN ports feeding into the queues
The number of simultaneous calls that Cisco Unified CME B-ACD can handle is limited by the number of PSTN ports, but these ports may not always be in use. For example, you could have three queues with ten slots per queue, but configure only 10 ports instead of 30 because you do not expect the three queues to ever be full at one time.
From this good doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00805f2305.html#wp1003323
Hope this helps!
Rob -
My ipad will not wake on incoming calls on face time
If I let my ipad go to sleep it will not wake up on incoming calls. My calls other devices are not received.
Try putting it into recovery mode.
Connect iPad to your computer via USB while it's turned on.
Open iTunes
Hold the iPad's Sleep/Wake switch (power button on the top of the iPad) down until the "power off" slider appears.
Power off the iPad completely (wait for the progress spinner to disappear).
Hold the Sleep/Wake switch AND the Home button down for a full 10 seconds.
After 10 seconds (the second time you'll see the Apple logo on the screen) release the Sleep/Wake switch while still holding the Home button.
Continue to hold the Home button for another 10 seconds until the USB connection image appears on the iPad's screen, then release the Home button.
A pop-up message will appear on iTunes saying it has detected an iPad in recovery mode, you must restore...
Click on the "Restore" button and the process will begin. -
After I dropped my Iphone 4s my back glass broke and now the screen won't turn on neither respond to touch but i can hear incoming calls and notifications. I tried pluggin to computer/other power sources and also tried pushing the home and power button together for about 20 seconds
It's broken due to neglect, namely being dropped. Replace it.
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Need help Creating a translation pattern that adds dial out digits to incoming calls
I came across an article yesterday and it showed the steps how to fix Missed Call/Received Call numbers so that you can dial them from the menu correctly (auto-add a 9, etc.)?
I tried it this morning and came up with this translation pattern:
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
This translation pattern rule 1 adds the dial out character 8 and strips 201 for local calls. Rule 2 adds dial out character 8 and adds 1 for long distance. The purpose of this translation rule is when the ephone receives the phone call the characters 8 and 1 are added so when you quickly need to redial you do not have to edit the number and add 8 for each call.
I tested the translation-rule:
ROUTER-2911#test voice translation-rule 6 9082121231
Matched with rule 2
Original number: 9082121231 Translated number: 819082121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#test voice translation-rule 6 2019121231
Matched with rule 1
Original number: 2019121231 Translated number: 89121231
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
ROUTER-2911#
Issue is I am not sure with my inbound call leg if it can even work. We dial out through the SIP Trunk and the incoming is translated to the AutoAttendant on Cisco Unity Express.
voice translation-rule 1
rule 1 /2015552100/ /2003/
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
dial-peer voice 9 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
call-block translation-profile incoming BLOCKED-INCOMING
call-block disconnect-cause incoming call-reject
session protocol sipv2
session target dns:nd01-04.fs.SIPPROVIDER.net
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
Can what I am trying to do be done with my current setup?Hi patldmart012,
The dial-peer 9 that you have attached will not be affected by following config
voice translation-rule 6
rule 1 /^201\(.*\)/ /8\1/
rule 2 /\(..........\)/ /81\1/
voice translation-profile filter_Incoming
translate calling 6
Because you have not applied the translation profile "filter_incoming" on the dial-peer.
Could you please provide the exact call flow?
Along with that, If you are facing issue with calls on SIP Trunk, please collect following debugs in a logging buffer and attach the file. I will analyse it and will get back to you.
debug voip ccapi inout
debug ccsip message
debug voice translation
Debug h225 asn1 (If H323 involved)
Debug h245 asn1 (If H323 involved)
Debug MGCP Packets (If MGCP involved)
Also provide the running config of the GW.
These are verbose debugs, so please collect them in the following manner:
Router(config)# service sequence
Router(config)# service timestamps debug datetime msec
Router(config)# logging buffered 30000000 7
Router(config)# no logging con
Router(config)# no logging mon
Router# Clear log
Router# term no mon
<Enable debugs, then wait for issue to occur.>
Router# term len 0
<Enable session capture to txt file in terminal program.>
Router# Undebug all
Router# sh log
Once i have the logs, i will analyse it and will get back to you.
Regards,
Mudit Mathur -
Hi All!
I have a problem with the SPA122 telephony adapter, uncorrectly process the subscriber signaling at the end of the call.
1) Outbound call from FXS port SPA122 . When a remote caller hangs up first , the subscriber SPA122 Reorder Tone played with a delay specified in the Reorder Delay. This circuit is working properly.
2 ) Incoming call from VoIP to the SPA122. When a remote caller hangs up first , subscriber on the FXS port of the SPA122 hears silence ~ 3-4 seconds , then SPA122 plays Dial Tone, as if he had just picked up the phone and he 's going to call . No signal lights out (Busy Tone or Reorder Tone) will not play .
Config is attached.
Model: SPA122, LAN, 2 FXS
Hardware Version: 1.0.0 Boot Version: 1.0.1 (Oct 6 2011 - 20:04:00)
Firmware Version: 1.3.2-XU (014) Jul 2 2013
Recovery Firmware: 1.0.2 (001)
WAN MAC Address: 6C:20:56:55:3A:B6
Host Name: SPA122
Domain Name: (none)
Serial Number: CCQ16450LG3
However, other VoIP terminals registered to Huawei, including older versions of the Linksys SPA2102 work in these scenarios correctly.
Where to kick it?[2] is misconfiguration on your's side. You have CPC turned on, but no CPC capable device. Set CPC Duration to zero to turn off CPC.
By the way, wrong forum for your question. You should consider to move it to space. -
Show dial pad when answering incoming call
I need to be able to press '1' after I answer an incoming call (Google Voice call screening). The dial pad doesn't appear on the screen. If anyone has solved this problem, I'd be grateful to hear how you did.
format the phone. make sure all users from contact list have set their default number. this helped in my case.
Message Edited by damiri on 30-Aug-2009 10:04 PM -
Incoming calls adjust computer volume unwarranted
A few times in the past I've tried to use "supported devices" with my Mac and newest version of Lync via USB. This didn't work since you still needed to register the devices independently through the device itself (Polycom CX600, Snom 821, etc.)
instead of through Lync like you can with the PC version... (that's another sad story of Microsoft's software claims!)
To get to the root of this issue, now that I no longer use any USB connected devices with Lync, I cannot get the software to STOP adjusting my system's audio levels upon incoming calls. I currently use a standalone phone device for phone calls. The software
still shows a "Custom Device" attached and the only selection under "Audio Device Settings" and there's no way to de-select it or turn it off.
Any ideas?Are you running Skype by chance? It has a similar behavior.
The following links my point you in the right direction:
http://www.mac-forums.com/forums/os-x-operating-system/236819-macbook-air-how-disable-osxs-auto-switching-volume-when-using-headphone-jack.html
https://www.bartbusschots.ie/s/2007/12/01/stop-skype-messing-with-your-volume-behind-your-back/
Andrew Morpeth
Lync Server Specialist - Auckland, NZ
Blog - http://www.lync.geek.nz
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question, please click "Mark As Answer" -
Iphone 4S; Phone & audio piece; Incoming calls are NOT routed automatically??
When I dial out; no problem the calls automatically goes through ear pice.
When I get incomming calls the call goes to phone, not even to the speaker in in the phone. So I have to
1.Slide the bar (then first realizing (again) dam: I hear nothing)
2. Choose the box for speaker options
3. Choose my blue ant piece..
By then some callers think I am not picking up or there is somthing wrong..!
My 3 colleaguers have same phone and no such problems. We are at a loss and I could not find reply in community groups. There was note about Iphone 5 and not correcting to Car speaker automatically. This was noted as a softwre issue Apple might address in next version. But since my colleagues do not have problem with their Iphone it must be either my settings or a weird fault with my phone.
One of my collegues have same ear pice as me, so the issueis not that either.
Is there some weird setting I need to fix to correct the problem?
Thank youAre you able to make calls? I would try a couple different things -
1. Try doing a double hard reset - Press and hold home button and the sleep wake button at the same time until the screen displays the slide to shut down red button. Continue pressing and holding both the buttons until the screen goes black and then the apple logo flashes up. Continue pressing and holding both the home and sleep wake button beyond this point until the screen goes black again. At this point, let go of the home button and press once the sleep wake button as you would to start your phone. See if this double hard reset helps clear out any cache.
2. If this fails, I would try restoring the phone as a new phone in iTunes. Backup your phone before this so that you dont lose your contacts, etc. You can do a couple test calls once the firmware is installed before setting up the phone, data, apps, etc. If it looks good, try restoring your phone from an earlier backup.
If everything fails, call AppleCare or see a Genius in the store. -
Incoming calls are going directly to Voicemail, Iphone 5S
I switched phones from an Iphone 4 to this Iphone 5S and my incoming calls are going directly to voice-mail and not ringing the phone. The only app I downloaded that is different from my Iphone 4 is the GroupMe app, but I don't know if that is necessarily the cause. Anyone have a suggestion? thanks
Hi, I don't know if this info will help many, but for a few it may. I had this problem over the weekend. Called Tech Support 2x, they tried everything, resetting,, *73, *228, *900 , there were no "do not disturb" settings, etc. They prepared a work ticket for the next level of help.
Meanwhile, I couldn't figure out what changed all of a sudden. I knew I had no calls coming in after accessing my voicemail. I looked at my recent calls and noticed i had dialed a *83 prior to the voicemail *86. Fat thumbs. I remember years ago, having the features of *83 for do not disturb and *84 to stop the feature. Just for giggles, i dialed *84, tried from another phone calling my number and WHOOLA, it came thru.
If anyone has these issues, try the *73 first. But just for the heck of it, do *84 as well. Takes half a second. If those dont' work, then call Tech Support. I wanted to share to help at least some of you as I understand the frustration of not having incoming calls. I even called Tech Support and shared the info. As this was not an option given to me from Tech Support, but then again, i don't believe *83 or *84 is offered anymore. I just dated myself, but hope it helps others ! -
Switched over to MGCP from H323, no incoming calls fast busy
Hello, I'm on the network side crossing over to the Voice side. We replaced a 3825 Voice Router at a branch office with a 2921. The 3825 was setup with a T1 and had a PRI connected to the FXO ports. The 2921 is now connected via fiber and TAC helped get the router registered to the Call Manager. I'm trying to match up the old dial peers on the new router. I can't make out going or receive incoming calls, I get a This Call Can't be completed at this time.
When it was on the T1, the branch office was using H323. Now that's connects to the same CUCM, it's on MGCP. Shouldn't the old Dial Peers work on the new router?I had to configure the FXO ports for the DN to route the incoming calls. I learned just because the Call Manager configured the Voice Gateway as a client, you still must configure the FXO ports to route the main DID to a DN on the LAN.
Great advice, learning alot about telephony and VoIP.
Wish I had more experience troubleshooting the CUCM and DID portion, also learned the phone compnay doesn't turn up their ISDN switch until you configure your PBX. And a PRI testing isn't the same as the data T1, it's about having the Signaling channel turned up and or configured.
Lessons learned, wished I would have crossed over to VoIP earlier. -
Fast busy signal with incoming calls and unable to connect to voice mail
We are using CISCO 2821. The system started to have problem connecting to voice mailbox (when dialing internally to the VM extension) and it used to take a reboot to bring it back. Now it will not connect to voice mail (gets fast busy signal), plus incoming calls get a fast busy signal after the first ring tone. To make it worse, the Call Manager Express web page does not load. Does anyone have any idea how to solve the problem? Thanks!
Called ATT...they verified the problem and said it was an intermittent signal problem....
only issue is I have to keep the phone off to make sure I received voicemails....
Definitely NOT an iPhone problem, but an ATT problem. -
Unity Express - Incoming calls wont get voice mail
CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.
However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.
I searched for another post which suggested the following commands:
telephony-service
call-forward pattern .T
voice service voip
allow connections from h323 to sip
I've double checked them and there's still something wrong.
Here's my current configuration:
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
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000301
max-ephones 24
max-dn 24
ip source-address 192.168.20.1 port 2000
auto assign 1 to 24
system message Comtek
voicemail 3000
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
time-webedit
transfer-system full-consult
transfer-pattern 2...
transfer-pattern 3...
directory last-name-first
directory entry 2 2001 name Phone Two 7912
directory entry 3 2000 name Phone One 7970
ephone-dn 1 dual-line
number 2000 secondary 441833000000
call-forward busy 3000
call-forward noan 3000 timeout 10
no huntstop
ephone 1
no multicast-moh
device-security-mode none
mac-address 0017.0EF0.3642
type 7970
button 1:1
So pros, any suggestions?
ThanksI made a new dial-peer to handle incoming calls as follows.
dial-peer voice 1000 voip
description Incoming SIP
translation-profile incoming SIPin
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay rtp-nte
no vad
The translation-profile puts the call through to my 2000 extension.
This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.
To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting
dur 00:00:00 tx:0/0 rx:0/0
Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
This is the "show call active voice brief" for an external incoming call when the call is established.
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active
dur 00:00:02 tx:105/16800 rx:104/16640
IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active
dur 00:00:02 tx:0/0 rx:105/16800
Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
Not too sure where to go from here. -
Bought a new phone and now we can not receive incoming calls
This week I bought a new Panasonic cordless handset with no answering machine to replace our AT&T desk phone. I set it all up and waited the 7 hours for it to charge. Checked for dial tone, ringer, and volume. Called the house # from my cell phone and the screen on the Panasonic never showed an incoming call - kicked into voice mail (provided by Verizon) after the preset 5 rings so I left a test message. Panasonic screen showed a "1 new voice mail message". I was able to dial the voice mail access phone # and listen to my test message. Disconnected the new handset and reconnected our old phone - worked just fine. Returned the Panasonic to the store where purchased and exchanged for another (same model). Sales assoc. said there may be a code needed from Verizon. Spent hours on the phone attempting to speak to a real person at the customer service # and decided none work there anymore - everything was automated! After "4 digit pin" frustrations I gave up!
So if you've read this far and can help I'd sure appreciate it! Line works fine, old phone works fine, checked ringer/volume on new phone. Only working phone jack in the house (used to have 2 but when we contracted with V last year they only changed over one - another frustration). And how can I find out when our contract is up?!?Your mostly talking to peers here. Better off using the
http://www22.verizon.com/support/residential/contact-us/index.htm
option which is on all the Verizon pages. Should always lead you to phone numbers (1-800-verizon) and often an online chat. Sometimes will show you and email option.
As far as peers helping we would need to know what type phone service and general area of the country. Obviously varies as I have had no trouble with panasonic, uniden and other phones at my two locations (NY and FL).
Maybe you are looking for
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