SPA3000 dial of Voip to PSTN problem
Do it with the dial plan. Set the PSTN Caller default DP a number available and configure the dial plan:
(S0<:[email protected]>) where [email protected] is the sip uri when you want to transfer the call.
If it's a PSTN number you want to transfer the call to you must configure tab PSTN line with the credentials of voip to voip account, you want to use to transfer the call.
Of course, the line 1 tab you set enable component IP: Yes, if you send to a sip uri.
Tags: VoIP Adapters
Similar Questions
-
need help on voip to pstn oubound call on spa8800
Hello everyone,
I deployed a system and gateway Cisco SPA8800 3cx VoIP system.
internal calls arrive properly. external calls are produced as well.
But impossible to make outgoing calls to the RTC from the voip network.
Please how to configure a line dial plan in sp8800 to allow the outgoing call?
Thank you for your intervention.
Duplicate of needing help on voip to pstn oubound call on spa8800
Please remove this redundant thread. Thank you.
-
SPA3102 problem: State PSTN: still rings in the State
We got a new router Linksys SPA3102 VOIP gateway, and we have some problems to operate.
The SPA3102 is properly connected to the local network, but it seems to have problems with the incoming PSTN line.
However, it does not sound when I call the number RTC. On the other hand, State PSTN always says 'Ringing '...
When I restart the SPA3102, then the extension on the pbx sipx she must call rings (without anyone calling the PSTN number).
The status page says something like:
Voltage: 14 (V) loop current: 0.0 registration of (my) State: not recorded the last record to: next record in: last call VoIP number: [email protected] last call number RTC: last calling VoIP: last PSTN Caller:, last RTPC Disconnect reason: CPC Signal RTC activity Timer: 30000 (ms) mapped Port SIP: 5061 Type of call: VoIP Gateway call VoIP State: Idle PSTN State : ringer VoIP: tone RTC:
(Note: the bridge didn't need to register for the sipx-pbx).
While it fires after a reboot of the SPA3102, nothing happens when I dial my mobile PSTN - I hear only ringing on mobile (it says busy or more).
Does anyone have advice?
Andreas
It did not break. The problem was with the cable to connect to the UK British Telecom. I haven't used the gray wire that came with it, but used a telephone cord UK I had to hang out - bad move.
When we talk about Linuxemporium, there where I bought the SPA3102, I was made aware that the cable might be the problem. So I went on the road in a small phone shop to get a US - UK adapter (packaged with the SPA3102 is still an American cable), that didn't fix it. I had another well-known adapter from a retailer of Electronics UK, and that one has solved the problem. The SPA3102 fortunately now works as my PSTN gateway.
I hope this can help others with similar problems.
-
Hi all
What am I supposed to hear when the SPA picks up the call?
My VOIP works and my PTSN works. Ive plugged the socket "phone" on my Fax (which is on the same phone number as adsl, and is set to 'reply' mode) 'line' on my SPA3102 and implemented according to the instructions of JMG (www.jmgtechnology.com.au/spa_3102_guide.pdf). The fax receives faxes. If I dial my phone number of voip phone attached to the rings SPA and everything is good. So: I dial the fax number, the phone in the SPA rings, then it stops. Here, I'm guessing that the SPA should have picked up the call (after the series of 5 seconds). But the phone I cling will tone, that it seems Ive been hung up on. I had once a voice saying: 'Please enter pin followed by the key voice portal sharp code', but that only happened once and the PIN that I had, I put in the SPA did not work. Is that what I'm supposed to hear? What am I supposed to hear when the SPA picks up the call?
Thank you
T.On the SPA3102, if you 'Ring through line 1', Yes, an incoming call from the PSTN line rings the phone on the SPA3102 to the number of seconds defined in the PSTN answer delay setting. After the gateway PSTN-to-voip the SPA3102 will get the call. If you set the PIN then the gateway which will respond to the call and give you three "beeps" as a signal for you to enter a PIN. If you have authentication set to None and the default dial plan configured to receive dialing number, you will hear a dial tone. If you have authentication set to None, and the defined default dial plan to automatically dial a voip number that you will continue to hear ringing until the answer to the voip call.
I don't know what the impact is a device of fax on the PSTN line.
-
SPA 3102 - call "RTC for VOIP" setting is not on CLI/CID phone line active.
Hi guys
Recently, I discovered if the PSTN line is having with CLI/CID then helped the PSTN to VOIP Call not established with the standard settings. But others see (VOIP to PSTN) works well OK. Basically, it works well with the PSTN without CLI/CID lines. I guess when with CLI, there different voltage levels online. ? You would let me know what are the parameters that I need to SPA in order to work with the PSTN line that allowed the CLI/CID please? This situation is covered by Sri Lanka Telecom PSTN lines. At the moment I have IT of standard parameters such as it comes with the SPA3102.
Your feedback and help here...
Syslog & debugging can save so much time...
Line with no CLI:
INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-a16a89a0 From: PradeephSL
This INVITATION is accepted by proxy.
Line with CLI:
INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-ea75ea0 From: PradeephSL
This INVITATION is rejected by proxy with "403 incorrect authentication."
But Howard has already struck and explained...
-
Connection of remote offices SPA3102
My boss just bought two SPA3102 VoIP routers in an attempt to set up a VoIP connection to a remote office, thus avoiding expensive international calls. I'm looking for advice on setting up.
This is the ideal configuration, we're after. Box, it is in the seat, and it is connected via the RTC port on a phone line (in fact the PBX). His IP address is 192.168.32.8 (we are without using the routing of each box function, then the two boxes are connected to the network via the WAN port, with the side LAN not used at all). Any incoming call on the telephone line must be connected directly on the phone connected to the box of two. There is no phone connected to the box one.
Box 2 is in remote desktop. For testing purposes, its IP address is currently 192.168.32.9. He got a telephone connected to the telephone port, and the PSTN port is connected to a standard telephone line. We want the phone to use the RTC line by default, but if the user dials 9 then, we wish the appeal directed to the PSTN line on box a (i.e. PBX.), where it can dial into a PBX it seeks.
The offices are connected by a virtual private network, so there is no NAT anywhere to worry.
I have a problem in this place, what with the instruction light manual and confusing options, so any help much appreciated!
Here are the settings I've worked. I have omitted some.
Head office
WAN configuration:
Connection type: static IP address
Static IP: 192.168.32.8
Subnet mask: 255.255.255.0
Enable WAN Web server: Yes
Line 1:
Activate the line: no
PSTN line:
Activate the line: Yes
Use outbound Proxy: no
Register: no
Call without Reg: Yes
Call for years without Reg: Yes
Numbering plan 1: (P1
[email protected] >)(causes the call to transfer to the other box after a delay of one second)
Enable VoIP to PSTN Gateway: Yes
PSTN to activate the VoIP gateway: Yes
PSTN through line 1 ring: no
PSTN response time: 0
PSTN Ring through delay: 0
Remote Desktop box:
WAN configuration:
Connection type: static IP address
Static IP: 192.168.32.9
Subnet mask: 255.255.255.0
Enable WAN Web server: Yes
Line 1:
The line activation: Yes
Register: no
Call without Reg: Yes
Appeal of ABS without Reg: Yes
Auto RTC backup: Yes
Dial plan: (0 x<:@gw0>|) <>
xx. <:@gw0>| <7:192.168.32.8:5061>)(This operation transfers the call to the seat on numbering 7 9 box gets an outside line, any number beginning with 0 is also formed through the PSTN line)
PSTN line:
Activate the line: Yes
Enable VoIP to PSTN Gateway: Yes
PSTN-to-VoIP gateway enable: no
I have a problem on the left: calls to the phone remotely (both by the PBX and telephone line remote) are very quiet. I played with the settings of Gain PSTN SPA to compensate, but I set the remote zone to its maximum of 12 and it is still too quiet, so I think that there is a problem elsewhere.
-
ATA 186 to connect to an SPA3102
I have things hung as follows:
PBX<==>SPA3102<==><==>ATA186 analog telephone
IP addresses:
SPA3102 - 10.1.1.144
ATA186 - 10.1.1.45
SPA3102 Config:
5.2.13 firmware (GW002)
Line1 is off.
Ethernet cable is connected to the Internet portOn the PSTN line tab:
SIP port: 5060
POST SIP Port: 5060
Proxy: 10.1.1.45Register: No.
Call without Reg: Yes
Call for years without Reg: Yes
Full name: 423
User ID: 423Password: 123456
Numbering plan 1: (S0<:gw0>)
Dial Plan 2: (S0<:10.1.1.45>)
PSTN ring across the line 1: No.
ATA186 Config:
Firmware: 3.02.01
Under SIP settings:
UID0: 423
Password0: 123456
SIPRegOn: 0
Proxy: 10.1.1.144
Anyway, I can't get these two to work in this configuration. When I try to make a call from a PBX to the ATA186, I see from the SPA3102 will receive an invitation to the ATA186. The ATA186 responds with a "404 Not Found".
When I try to make a call from the ATA186 to the PBX, I see an invitation of the ATA for the SPA. Inside this prompt, there "sip:
@10.1.1.144. Then I see the answer of the SPA with a "status: 100 Trying" followed by a "status: 503 Service unavailable '. No indication on how to connect the two devices together will be very appreciated! Thank you!
Assuming that your Dial Plan 1 is the VoIP calling by default DP under the Voip to PSTN gateway, I think the dial plan should be (xx.). Maybe the (S0<:gw0>) works, I don't know, try (xx.) and see if that makes a difference. You told me that was planted in the line PSTN tab UserID to 423. If the ATA call 423 ([email protected]/ * /: 5060) the SPA should return a tone and wait the digits for numbers. If the ATA call the PBX extension number ([email protected]/ * /) where xxx is the number of post, then you have a phone on a floor and the SPA3102 should dial this number on the FXO port after there he won.
The INVITATION contains the number sent to the SPA3102.
If you are running WireShark he reformatted the sip signaling so that you can see exactly what is happening. After you capture a test call click you on the phone, then on "Voip calls". WireShark then puts up a screen display detected than VoIP calls. You click your test call to highlight, and then click 'stream '. Wireshark is implementing another magnificent screen that shows the SIP signaling between the ATA and the SPA3102. My version of WireShark is Version 1.6.4. There may be future versions.
Another debugging tool is the Trace of the Sip Debug. You install a program on a local computer syslog, put the ip address of the local computer under debug the server on the SPA3102 System tab, set the level of debugging on 3 in the System tab and set the Option Debug Sip to FULL-on the tab line PSTN (then on the tab line too if you use). You can download a simple Windows pc syslog program here:
https://supportforums.Cisco.com/docs/doc-9862
Traces of Debug Sip will show internal things going on inside the SPA3102, which do not appear in the Trace of WireShark, but WireShark trace shows exactly what is happening inside and outside the SPA3102 on the ethernet link.
The duration of CPC Min setting you mentioned is for the detection of a signal coming into the attached FXO analog line port CPC. As you know a signal CPC is a complete the voltage drop on the line of signalling the far part disconnected from the call. You would increase the minimum to stop detecting voltage drops false that disconnect an ongoing call. I'm surprised that increase the duration would solve your problem of disconnection.
-
can't make an external call with CuCM and H323 gateway
Hi experts, I have a problem with CuCM and h323 Gateway. I use a 2811 router to make a gateway H323, CuCM server and Switch layer 2 build a VoIP LAB, this my network:
and my router configuration:
Current configuration: 2335 bytes
!
version 12.4
horodateurs service debug datetime msec
Log service timestamps datetime msec
no password encryption service
!
router host name
!
boot-start-marker
boot-end-marker
!
forest-meter operation of syslog messages
!
No aaa new-model
!
!
!
dot11 syslog
IP source-route
!
!
IP cef
DHCP excluded-address 192.168.1.1 IP 192.168.1.10
!
IP VOICE dhcp pool
network 192.168.1.0 255.255.255.0
option 150 ip 192.168.1.2
default router 192.168.1.1
!
!
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
service additional h450.12
Fax protocol cisco
!
!
!
vocal h323 class 1
H225 timeout tcp establish 3
!
!
!
translation of the voice-rule 1
rule 1 / ^ 9.
!
!
voice translation-profile PSTN
translate 1 called
!
!
voice-card 0
!
!
application
Service flash aa1: its - cisco.2.0.2.0.tcl
paramspace English language en
paramspace English index 1
Param aa-driver 0839959204
paramspace location English flash:
prefix English paramspace
!
!
!
Archives
The config log
hidekeys
!
!
interface FastEthernet0/0
IP 192.168.1.1 255.255.255.0
automatic duplex
automatic speed
H323-gateway voip interface
port of link voip H323-gateway 192.168.1.1
!
interface FastEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
IP forward-Protocol ND
no ip address of the http server
no ip http secure server
!
!
!
control plan
!
!
!
voice-port 0/0/0
cover of the echo - cancel 32
connection ERA opx 3100
Description 08395959204
activation of the caller ID
!
voice-port 1/0/0
!
voice-port 0/0/2
!
voice-port 0/0/3
!
!
!
1010 Dial-peer voice pots
Service aa1
incoming called number 08395959204
direct line to inside
port 0/0/0
!
voice pots Dial-peer 2020
PSTN description
translation-profile outgoing PSTN
destination-model 9 t
Setup progress_ind allow 3
progress_ind enable progress 8
port 0/0/0
Forward-digits all the
!
Dial-peer voice 10 voip
destination-model 1...
Setup progress_ind allow 3
progress_ind enable progress 8
h323 voice-class 1
session target ipv4:192.168.1.2
DTMF-relay h245 alphanumeric
Codec g711ulaw
No vad
!
!
!
!
Line con 0
line to 0
line vty 0 4
Cisco password
opening of session
!
Scheduler allocate 20000 1000
end
I have configured the gateway h323 and attach files to router parttern on follow CUCM
What is the problem and how to solve?
Thanks for all the help.
Your DP pointing CUCM is wrong, or your ÉRA OPX, depending on what you need.
voice-port 0/0/0
cover of the echo - cancel 32
connection ERA opx 3100
Description 08395959204
activation of the caller ID
Dial-peer voice 10 voip
destination-model 1...
Setup progress_ind allow 3
progress_ind enable progress 8
h323 voice-class 1
session target ipv4:192.168.1.2
DTMF-relay h245 alphanumeric
Codec g711ulaw
No vad
Your DP is waiting for 1... and you send 3100, they must match.
You might need to change the significantt numbers and entering CSS, but only you can know
For outgoing, you are throwing 9 predot, but again, expect it in your RFP, again, change as you need.
voice pots Dial-peer 2020
PSTN description
translation-profile outgoing PSTN
destination-model 9 t
Setup progress_ind allow 3
progress_ind enable progress 8
port 0/0/0
Forward-digits all the
You cannot transfer the numbers all send you also 9 for each call
Change your configuration you want / need.
HTH
Java
If it helps, please note
-
I have a few computers on a small network ethernet, via a residential gateway (i.e., router), which allow computers to share files and printers, and share an Internet connection cable. Most of the computers on my network is not yet have PSTN modems and (fortunately), I had the need to use remote access services in more than six years. However, in my IE Options CONNECTION tab it says that MSN remote connection has been implemented.
Sometimes, I received a box of popup with some sort of message about the remote connection and work offline. Of course, I'm not looking for a modem connection. I have changed since my connections to NEVER A CONNECTION settings, but I really want to just remove the connection distance for not having any conflicts in my system or Internet access in the future.
My question: would I be causing no problem for me on the road if I just remove this connection remotely, and if deleting it would not be a problem, so I have correctly and safely how so? Thanks in advance for your help.
I don't think that delete a unused dial-up connection can cause problems. To be completely safe:
1. make a system restore point.
2. open the network connections, right click on the connection file and delete it.
3. in case of problems, restore the system to the way it was before deletion.
Another response of the community of Windows XP newsgroups -
help with SPA 3102 (question graphcal)
HI guys here is my situation (I draw so it would be easier for future reference):
I want to pick up A phone and dial the Ext 101 101, 102 for Ext 102 and so on.
also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.
castro69 wrote:
also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.
any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.I guess that's an analog PBX, otherwise you wouldn't need the SPA3102 through the internet.
For communications between the SPA3102, I would use direct ip call, using the external ip address and the sip port numbers. Think that the SPA3102 is two separate cards inside the box where treat you everyone with its sip port number added to the external ip address common.
I would setup port sip distinctive numbers on each of the baths to keep things straight. You have a number of separate port for tabs of line 1 and line PSTN. You will need to send these to the SPA3102 adapter port numbers in their respective routers or firewall of the router will reject incoming packets on the internet. I would also convey the port range rtp for voice, flow packs.
On each tab of the line 1 and RTC, you define NAT Mapping Enable: YES, not record, make call without Reg Yes, years call without Reg No. I put your external ip address on the Sip tab under EXT IP. This will tell the SPA3102 to use this address in the sip signaling. I assume you are using static external ip addresses. On each tab of the line 1 you would activate IP Dial Yes.The analog PBX is connected to the FXO port on one of the Spa. You should check the voltage level hung up and won and then set the line parameter usage on the RTC of the SPA line tab to halfway between the two readings. You can read the levels of tension on the PSTN line tab. Calls to the PBX of the PSTN line tab will go through the voip to PSTN gateway. I set up the catwalk with http authentication and configure a user name and password.
Details are starting to become quite complicated. I'd get running through steps. Get a job step before moving on to the next step.
The 1st step would be to get A phone call/receive calls to a PBX. You can configure the line 1 for FXS phone attached A to use port location PSTN 2 as the proxy using http authentication, and you can then dial the extensions you want to call. Location 1 SPA3102 will send a guest of the sip Protocol to the tab location 2 SPA3102 from pstn line and the SPA3102 will dial the number on the FXO port to the PBX.
For calls coming from the other direction of a PBX to slot 2 SPA3102 the only place where you can connect a voip call is in the SPA3102 numbering plan. If you want to call only phone that is easy, install you just dialers-messengers automatic telephone in the pstn-to-voip dial plan.
I'm not clear about what you want to do with phone B I take is Extension 104.
I like your designs. Can save a lot of words.
-
difference between gateways, voice and Porter
Hello
Someone might well want explain (or provide a url link) the main difference between the voice gateways and a guard?
Concerning
Hello
Your best source of information is the book of Cisco Press 'Cisco Voice Gateways and guardians (GWGK)' which explains these 2 more in detail. I have included information to help...
The role of voice gateways:
In a VoIP environment, voice gateways are the interface between a VoIP network and the public switched telephone network (PSTN), a private branch exchange (PBX) or analog devices such as fax machines. In its form the simplest, a voice gateway has an IP interface and a legacy phone interface, and it manages the many tasks involved in translation between protocols and transmission formats.
At least one gateway is an essential element of any IP telephony system that interacts with the PSTN or analogue devices. In addition, when the front doors are configured correctly, many can resume when it is inaccessible to a Cisco CallManager.
The gateway that enables communication between the two networks by performing tasks such as these:
Interfacing with the PBX and the PSTN or IP network.
Support IP call control protocols, more than time-division multiplexing (TDM) call control protocols.
For calls between VoIP and PSTN closing networks and REPETITION of the call media and noting the call setup and teardown artist.
Providing additional services, like call hold and transfer.
Relay of DTMF multi-frequency (DTMF) tones.
Supporting modems and analog fax via the IP network.
In a Cisco CallManager network, a gateway must also do the following:
-Support the CallManager redundancy in replacement of spare CallManager.
-Support survival call when no CallManager is available.
Gateways communicate with other gateways, guards, their end points or their control agents, such as Cisco CallManager call or a PBX.
Protocols that use Cisco gateways for voice signaling and media are:
Media Gateway Control Protocol (MGCP)
H.323
Session Initiation Protocol (SIP)
Skinny Client Control Protocol (SCCP)
Real-time Transport Protocol (RTP)
The role of guardians of the voice:
Access controllers help scale of large VoIP networks. Businesses that are geographically dispersed, networks of voice, or that have become so important that they are difficult to handle, could opt to segment their network. In a network of CallManager, you can create multiple clusters. In this case, you need to configure a full mesh of connections via the WAN IP to connect all segments or clusters. You need to configure the dialing information for each remote location on each bridge and CallManager cluster.
It is preferable to use the guards. In a network that has doormen, trunks are necessary only for the keeper and the goalkeeper keeps information of remote endpoint.
When you use access controllers, gateways and CallManager to register with their goalie. Doormen divide the network into 'zones', or groups of devices that fit to a particular access controller.
When an H.323 gateway receives a call that is intended for a remote phone, he asks the guard for the location of the endpoint.
If the call is for a different area, you can configure the guardian to allow it only if sufficient bandwidth is available. In more complex networks, you can use a directory gatekeeper to maintain information on all areas.
You can configure routers Cisco with the Cisco IOS appropriate as guardians H.323.
Please note the useful messages...
-
Fax protocol t38 cisco 2921 rescue % entry invalid
Hello friends,
I m to help cisco T38, but 2921 said an invalid entry.
Can you help me please?
Router2921 (config) #voice voip service
Router2921 (conf-voi-serv) cisco aid hs-redundancy 0 0 ls-redundancy of the t38 #$l
Fax protocol t38 ls-0 hs-redundancy redundancy 0 backup cisco
^
Invalid entry % detected at ' ^' marker.__________________________________________________
Router 2921 #sh run
Building configuration...Current configuration: 3639
!
! Last modification of the configuration to 16:38:58 UTC Friday, May 24, 2013
version 15.2
horodateurs service debug datetime msec
Log service timestamps datetime msec
no password encryption service
!
router host name
!
boot-start-marker
boot-end-marker
!
!
card type e1 0 0
!
No aaa new-model
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
IP cef
!
!
!
!
!
!
No ipv6 cef
Authenticated MultiLink bundle-name Panel
!
!
!
!
primary-net5 ISDN switch type
!
!
!
voice-card 0
dspfarm
DSP services dspfarm
!
!
!
voip phone service
Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
!
voice class codec 100
preferably 1 codec g729r8
g711ulaw codec preference 2
g711alaw preferably 3 codec
!
voice class h323 200
H225 timeout tcp establish 3
H225 timeout Setup 3
!
!
!
!
!
!
license udi pid CISCO2921/K9 sn FTX1712AJLJ
licence accept the end-user agreement
HW-module pvdm 0/0
!
!
!
!
redundancy
!
!
controller E1 0/0/0
No.-CRC4 framing
!
controller E1 0/0/1
!
synwait-time of tcp CSDB 30
CSDB tcp idle time 3600
CSDB tcp finwait-time 5
CSDB tcp reassembly max-memory 1024
CSDB tcp reassembly queue-max-length 16
CSDB udp downtime 30
CSDB icmp idle time 10
CSDB max--a session 65535
!
!
!
!
the Embedded-Service-Engine0/0 interface
no ip address
Shutdown
!
interface GigabitEthernet0/0
no ip address
Shutdown
automatic duplex
automatic speed
!
interface GigabitEthernet0/1
no ip address
Shutdown
automatic duplex
automatic speed
!
interface GigabitEthernet0/2
no ip address
Shutdown
automatic duplex
automatic speed
!
IP forward-Protocol ND
!
no ip address of the http server
no ip http secure-s
!
!
!
!
!
control plan
!
!
Voice-port 1/0/0
!
Voice-port 1/0/1
!
voice-port 1/0/2
!
Voice-port 1/0/3
!
voice-port 2/0/0
!
voice-port 0/2/1
!
!
!
!
!
!
profile MGCP default
!
!
dspfarm profile Conference 1
Codec g711ulaw
Codec g711alaw
Codec g729ar8
Codec g729abr8
Codec g729r8
Codec g729br8
maximum sessions 4
associate the PCRS application
!
voice pots Dial-peer 1000
Llamadas PSTN description
destination-model 92 [2-9]...
direct line to inside
Forward-digit 8
!
voice pots Dial-peer 1001
Llamadas Nooduitgang description
destination-model 91 [34].
-More-
!Owner Cisco fax relay is not supported by PVDM3 DSP. You use the pass through or none.
Please remember useful frequency responses and identify useful or correct answers.
-
Slow download speed after downgrading to Windows 7 64 bit
Hi all!
I'm new here! My first laptop is a Toshiba Satellite since 2003. Since then, I got one couple of others.
Now I bought a new: Satellite P70-A-11V, a portable professional high-end computer, which came with Windows 8, which I upgraded to 8.1.Unfortunately, the degraded system so bad that all pendrives made plant. I have connected my Toshiba camcorder, and it also crashed the laptop. The search indexing was not working outside the C: drive. The pointer of an optical mouse has started to freeze for seconds on the screen.
This last problem, which I think was due to the low power USB ports. So, I really had no choice after 2 restorations system total and installed Windows 7 Enterprise x 64 SP1. The operating system seems ok, but I always have problems, and one of them is that my download speed is very low, as well with LAN cable or wireless. I can't tell how is download speed.
My internet provider provides 100 MB/s internet access via optical cable. I test on Speedtest.net and normally, it gives a range of 50 to 80 Mbps actual download speed. Then, if I check the speed of the connection Wireless wireless connection status, I experience a maximum of 130 MB/s, which is constant. Large.
Yet, no download occurs at a speed of wretched 400 to 500 kbit/s, as if it was a dial-up connection. This problem was not present before.I tried to change several settings according to the forums, but nothing helped. I disabled the firewall and stimulated wireless (duplex 100 MB) and also tried through LAN cable. The rated speed is great, but apparently the slowness problem is inside the laptop, not outside. Any ideas will be greatly appreciated!
I checked the Toshiba download page and your laptop is fully supported for Win7 64 bit. Drivers LAN and WLAN are available also so I hope that you have installed all the drivers, tools, and utilities including LAN and WLAN. Can you confirm this?
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Siri is unable to place phone calls.
Siri is unable to place a telephone call to one of my contacts for about a month. After a long wait she say sorry I don't do make your call. I reset my phone without success. Anyone have any ideas?
If Siri dial a specific contact, the problem may be with the contact. Try to delete the contact and re-enter.
If you encounter a problem with Siri in general, this article can be useful:
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How can I ask Siri to make a phone call, if the telephone number is not in my contacts?
Hi people
I do a lot of driving on the road. I health home where I travel to the patient's home and do some physiotherapy with them in their homes. Many times I find myself adjust my schedule, while on the road in order to accommodate a schedule patient or let know that I'm running a little late. One of the best features of the Siri is that it can call the numbers for you while you are busy with another task, IE. at the wheel. Well lately, I noticed that she tells me she can't dial a phone number because the number is not in my contacts. Unfortunately, I can't put the numbers in my contacts because it is my personal phone and any personal information of patients cannot be on my phone, computer, etc., I usually have to call on the road. Before she could dial the number without any problems. Now, when I say"Siri, please call 352-xxx-xxxx, she said," I'm sorry I can't find this number in contacts, you would I look for this number somewhere else? No matter how I word it, even just saying "Siri, I know this number isn't in my contacts, but I still need to call 352-xxx-xxxx" I always get the same prompt of Siri.
That said, is it possible to fix this? It would certainly make things a little easier for me while driving instead of having to look at the bottom of the phone to dial the number. For any help or suggestion would be so appreciated. Thank you in advance.
Michael
I just tested a call by saying "call 800-xxx-xxxx on my iPhone 6. He knew immediately. I'm in iOS 9.2.1. Have you tried leaving out the ' Siri, please... " » ?
Have what troubleshooting you done? Update to the latest version of iOS? Reset the phone? Restore network settings? Restored from a return to the top? Restored like new?
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