Can't make outgoing calls
Hello
My problem is that whenever I dial a number I get the engaged tone and am told “the number is busy and to use ring back press 5”. I can still receive incoming calls. Have checked the line and there are no reported faults, BT line checker says the line is fine. Have plugged the phone direct into the master socket but still unable to make outgoing calls. Have checked to see if there's any call barring in place but upon entering the * # 34 # am told “this feature is not available”. Have tried cancelling all the other call features detailed in the “calling features user guide” but am still unable to make outgoing calls. No matter which number I dial I get an engaged tone and am offered the ring back service. Please help! Many thanks in advance.
Hi
As you have tried different phones in the test socket it does sound like the fault is not with your equipment. I suspect that it is an exchange fault and not a line fault.
You will now need to report the fault to your comms provider, If it is BT call from your home number on 151. I suspect that when you report the fault the line will be tested and it will test ok as the test they run is only a physical line test and as I said I suspect it is an exchange fault. Just be persistent as they will insist there is nothing wrong.
(If I have helped you in any way to say "Thank You" please click on the star next to the message. Thank You)
If I have solved your Issue please click the "Mark as accepted solution" button.
Similar Messages
-
Can't make outgoing calls but receives fine (I believe it's the ISP)
Hi.A small company were maintaining that uses a uc500 on a dedicated voip line suddenly can't make outgoing calls.This has occured on another two occasions when they were using a different ISP and It was due to them not paying their bill on time.This time we've called the ISP and they Insist everything Is fine on their end and that the company hasn't been disconnected.I can't find anything wrong with the UC.I can ping the ISP gateway.I can ping their SIP server and everything looks fine.I've added two debug files and would appreciate any assistance as my debugging knowledge Is not that great !
Turns out the ISP was receiving calls In G729 and they use G711 Alaw.We didn't change any configuration on our side so I'm not sure what happened.I've set the sip trunk to use g711 alaw via CCA but calls are still going through as g729.I'm not sure what to configure via the cli.I assume It has something to do with transcoding but my telephony cli knowledge Is limited.
Should I change the configuration for Individual dial-peers or Is there some global config. I can enter ?
I added g711alaw to a dial peer and now I can't make any inbound calls either.I removed the g711alaw from the dial peer but It hasn't changed anything.
Running debug ccsip message I receive "error 304 media type(s) unavailable. q.850 cause=65"
I didn't expect such a small change to cause so much havoc. -
I can't make outgoing calls from my Iphone 3GS. I am using Vodafone as my Carrier
I can't make outgoing calls from my Iphone 3GS. I am using Vodafone as my Carrier
I would but I work on a remote mine site, closest apple store 2 hour flight away. Hence the reason I'm asking here
-
Constant Buzzing and Can't make outgoing calls
For the past month or so, our phone line has had a constant buzzing. Now, we cannot even get a dialtone to make outgoing calls. We do receive calls, but it doesn't do us much good since the buzzing prevents us from talking to anyone anyway.
I am having the same exact complaint, "Constant Buzzing on line, and inbound or outbound conversation cannot be performed". I am in southern New Jersey just outside of Philly, and this problem was first recognized approximately September 19th. I went out of town and felt that the problem would be remedied by my return. On September 24th the same problem was still present on the line. I used to be a Computer Network Engineer years ago and have a telephone handheld "butt set" (handheld phone that the phone techs use to connect directly to the wires. I disconnected all of my premise equipment and connected to only the line coming in from the pole, and the problem still persists which rules out any "interior" equipment malfunction.
Original service appointment was scheduled for Thursday September 29 between 8AM and Noon, however when Thursday arrived, I noticed on the internet that my expected service visit time had been moved back (without any heads-up) to Friday September 30 between 8AM and Noon. Friday 12 Noon comes and goes with no results yet, and upon looking at the internet again I noticed that the expected service visit time had been moved back once again to Friday September 30 at 8 PM.
Very poor repair service for a "phone not usable" condition. -
Can't make outgoing call with Skype Connect
I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
My config:
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://111111111111111:[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=11111111111111
fromuser=11111111111111
secret=abcd12345
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
SIP Debugging enabled
[2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
> doing dnsmgr_lookup for 'sip.skype.com'
> ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 63.209.144.201:5061:
REGISTER sip:sip.skype.com:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 32495 REGISTER
User-Agent: Asterisk PBX 10.5.2
Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
Expires: 120
Contact: <sip:[email protected]:5061;transport=TLS>
Content-Length: 0
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
Call-ID: [email protected]
CSeq: 32495 REGISTER
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
Expires: 45
Contact: <sip:[email protected]:5061;transport=tls>;expires=45
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
<--- SIP read from UDP:192.168.1.16:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
Expires: 240
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 234
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 88651316 88651316 IN IP4 192.168.1.16
s=-
c=IN IP4 192.168.1.16
t=0 0
m=audio 16484 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.16:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.16:16484
Looking for 19739928881 in home (domain asterisk.test.com)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
== Using SIP RTP CoS mark 5
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.2
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301052 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
-- Called SIP/skype/+19739928881
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 407 Proxy Authentication Required
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.5.2
Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301053 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 180 Ringing
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: SipGW 8
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
-- SIP/skype-000000b1 is ringing
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 408 Request Timeout
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
== Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.16:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: ACKI wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
Scott
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=user
fromuser=user
secret=pass
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes -
Can not make outgoing calls???
When I dial out, the E for edge network disappears and I can not hear a ring. The iphone makes the call but the it won't ring and I can't hear anything when the person picks up. Right now my iphone has no calling function whatsoever. When someone calls me and I pick up they I can not hear them and they can not hear me.
Hey cmilz,
Has the iPhone ever been able to make and recieve calls? Does it work with the headset?
Try restoring the iPhone.
http://docs.info.apple.com/article.html?artnum=305744
Jason -
Receiving incoming phone calls but can't make outgoing calls
while connected to my vehicle's hands free device my outgoing call was dropped...when I tryed to redial nothing occurred to connect to the number; once home tried to make a call free from the hands free and same results; however, I'm able to receive calls to the device.
if your on sprint, they are experiencing a nationwide outage with certain ios devices, no ETA available, no work around as of now
-
Hi my Iphone 4S today wasn't able to make calls or send texts. When trying to dial it wouldn't give back a ring tone despite me having full bars. I turned the phone off and back on hoping to resolve the call service. It didn't work. At that point I called AT&T and they said I should have service. I checked with others with Iphones and they all were working so I know for a fact it isn't AT&T. I then decided to reset the entire phone to its original state. After that I hooked it up to Itunes and put my backup file back on. At this point the phone began to work again but then two hours later it did the same exact thing and I'm unable to text or call. I don't want to keep reseting it and reinstalling my backup as it is time consuming and a chore for such a short amount of time. I read some other posts and they blame 5.0.1 for the problem. I don't know if there is any facts to those posts. Does anyone know if Apple has come out and said that the new update has bugs? Or could it be that my phone is defective? I appreciate all of your help.
You use Restore to get the latest firmware build, not the Update button. You can only use the Update button, if there is an iOS update, say iOS 5.1.
this link talks about updating to 9A406 to fix the "No Service" issue with iPhone 4S:
http://iphone-and-i.blogspot.com/2011/12/fixing-iphone-4s-signal-problem.html -
I am freaking out! I am a college student and owner of my Iphone 3Gs. I just got it this past August and I don't understand what is wrong. I did not drop it or come into contact with any water or disabling substance. I am literally stuck on the calling page with favorites, recents, contacts, keypad and voicemail. When I push my Home button I am not brought back to my menu. When I hold it down voice control is enabled but I still can't get to my menu. Then when I try to hold the sleep button to shut off nothing happens. What's wrong with my phone? I can't even go into contacts and hit text message nothing happens. Also, when I get a message in nothing happens when I push view. Please help me.
Try a reset by pressing the home and sleep buttons until you see the Apple logo, ignoring the slider. Takes about 5-15 secs of button holding and you won't lose any data or settings.
-
How can i hide my extension number when i make outgoing calls
Hi All,
How can i hide my extension number when i make outgoing calls,what configuration should i make on the call manager server.
Regards,
LesterWe use a Route Pattern that mimics the telco feature *67. The RP is *679.xxxxxxx with a Transformation mask of 000000000 (just to get beyond the individual blocking on some private lines.
-
I have a iPhone 5 and I am not able to toggle between the show my caller Id option on and off. How can I do that ??? It always shows greyed out and also because of which I am not able to make outgoing calls. So please help
Not all carriers allow that to be set using preferences in the phone. Contact your carrier.
-
not able to make outgoing calls in iphone4. Regularly i face this problem, at the same time i receive incoming call, can do net surfing, can access email and sms, only outgoing calls get failed every time. please answer to solve this problem
Good day Sidharth Namrta,
It sounds like you are unable to make any calls, but you can recieve them, and everything else seems to work fine. I recommend you use the troubleshooting in the following article to help you get that resolved, named:
iPhone: Troubleshooting issues making or receiving calls
Follow the steps below to resolve this issue. Please test after each step.
Toggle airplane mode: Tap Settings > Enable Airplane Mode, wait five seconds, then turn off airplane mode.
Check your phone settings:
Check your Do Not Disturb settings: Tap Settings > Do Not Disturb.
Check for any blocked phone numbers: Tap Settings > Phone > Blocked.
See if Call Forwarding is turned on: Tap Settings > Phone > Call Forwarding.
Ensure that your software is up to date:
Check for a carrier settings update.
Check for an iOS software update.
Note: Some updates may require a Wi-Fi connection.
If the iPhone has a SIM card, reseat the SIM card.
If the iPhone 4 or iPhone 4s is on the Verizon network, dial *228 from the iPhone and select option 2 to update the Preferred Roaming List (PRL). The PRL determines the cellular towers the phone uses for cellular service, selecting those with the best signal strength.
Reset the network settings: Tap Settings > General > Reset > Reset Network Settings.
Try to make or receive calls in another location.
Attempt to isolate to one network band:
If you're having the issue on LTE, disable LTE, if possible, and try again.
If you're having the issue on 3G/4G, disable 3G/4G, if possible, and try again.
Contact the carrier to check the following:
Your account is properly configured to use the specific iPhone that has the issue.
There are no localized service outages.
Your account doesn't have a billing-related block.
Your calls don't have errors on the carrier system.
Restore the phone as new.
If the above steps don't resolve the issue, go to an Apple Retail Store, carrier, Apple Authorized Reseller, or contact AppleCare to send the phone in for service.
Thank you for using Apple Support Communities.
All the very best,
Sterling -
I'm able to receive calls, but cannot make outgoing calls. I'm also able to receive/send texts and emails. Someone please help!
Help is here for you, ahues1214. Our forum friends did provide some great comments. If you're receiving an error even when dialing the 10 digit pattern, please let us know the message and we'll be sure to find the reason for your calling difficulties. Please also let us know if this is effecting anyone else in your area, along with your zip code.
JenniferH_VZW
Follow us on Twitter www.twitter.com/vzwsupport -
My iphone 5 won't make outgoing calls
my iphone 5 won't make outgoing calls....i tried to restart network settings and then my whole phone all together....also i tried on wifi and off wifi and it wont ring......What should I do????
Settings > General > Reset > Reset Network Settings.
Shaker Ahmed wrote:
whats can i do any one from apple suprted replay mee fast
These are user to user support forums, as clearly stated in the Terms of Use that we all agreed to when signing up here. Apple does not read or respond here. -
IOS 8.02 not letting me make outgoing calls
I don't know what's going on. At first I thought it was a fail-safe when bad I was in bad service. But even on LTE my phone REFUSES to let me make outgoing calls. I can't even type in the number and call. Any way I try, it fails. Won't even pull up the dialing. Just registers as an empty press.
HELP! I can't even call Apple support.try a reset, no data loss. Hold down the home/sleep button together until you see the apple logo and then release, then wait for the phone to boot back up.
then check again.
Maybe you are looking for
-
About 6 months ago I upgraded to 16GB RAM and have had no problems. Have checked the system report for my memory and it says both of them are OK. Within the past 2 or 3 weeks I've been seeing crazy lagging at odd intervals with a message coming up t
-
Unable to link the values with subnodes in Tree
Hi All, I need to create a tree structure where main node is Sales Order. It has various subnodes based on document type. Then sales order based on each document type are to be displayed under corresponding subnode. The above requirement is getting p
-
First i can't download iis.Do u know a good link to download it? Is there way to open publish the site from easy php? Where can i find a video to see how to publish?
-
OS X Mountain Lion 10.8.5 and Server 2.2.2 on both Mac Mini Server and Macbook Pro. They are on the same WiFi network. I want to use the MBP to manage the Server on Mac Mini Server and use the MacMiniServer to provide Caching Service for my MBP, iPa
-
Keithley 2400 driver for Labview 8 not working
I searched the boards and found this driver for a Keithley 2400 that should run with Labview 8. I have Labview 8.6. http://sine.ni.com/apps/utf8/niid_web_display.download_page?p_id_guid=25B255F3AA83660EE0440003BA7CCD... I put the driver folder in th