Automatic outgoing calls with CUE?

Hello,
is it possible to make automatic outgoing calls with a CUE?
That is, can we give X numbers to the CUE and let him place X automatic calls, with a prerecorded message (for example, for a telemarketing campaign)?
Thanks and regards.

Hello,
I think to have solved by myself, by the command "Place Call" of the CUE Editor.
Regards.

Similar Messages

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: ACK

I 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

I cannot make outgoing call with uc500.

please help
i have uc500 router. i have conifured my dial peers very simply . i can receive the call from outside . but i cannot make a call. whenever i am trying to make a call.
just Ring Out is appearing with starting buy tone or some time it going telecom company. this is my simple dial peers configuration
dial-peer voice 50 pots
description long distance
preference 5
destination-pattern 9T
port 0/1/2
forward-digits all
no sip-register
dial-peer voice 51 pots
destination-pattern 9[2-9].........
port 0/1/2
forward-digits all

shakirullah,
Please collect below debugs from uc500 and attach it here.
Router(config)# logging buffer 5000000
Router#debug voip ccapi inout
Router#debug vpm signal
Router#debug voip vtsp default
Router#debug voip vtsp session
Router#Clear log
Now make one test outgoing call and re-create the issue.
Router# undebug all --->turn off debugs
Router# show log----->collect the entire output of this and attach it here
Mention calling and called numbers.
-Abhi

VCS-E unable to make outgoing call with search rule not found error

Hi
I deploy telepresence device registered to CUCM and connect with VCS-C and VCS-E. The issue now is I am not able to make outgoing call and my search rule show not found, request time out.
I use DNS zone to make the outgoing call and my VCS-E is able to resolve DNS SRV record. However search rule result show not found.
Any advise are appreciated.

Hi
I managed to make a call to address that I am trying to reach. But it supposed to come up with PIN number to enter instead I saw the message that no incoming video.
Can you please advise what could be the issue.
tvcs: Event="Call Connected" Service="SIP" Src-ip="10.84.83.101" Src-port="27224" Src-alias-type="SIP" Src-alias="sip:[email protected]" Dst-alias-type="SIP" Dst-alias="sip:[email protected]" Call-serial-number="24a63c27-42b4-4950-ba02-6a548b279a09" Tag="f060faef-441f-47a2-9c75-e1b0623b5e14" Protocol="TCP" Call-routed="YES" Level="1" UTCTime="2015-02-25 14:38:32,604"

FaceTime always fails on outgoing calls with my MacBook Pro. Is there some way I can fix it?

Have I accidentally disabled a connectivity preference that subsequently disabled FaceTime? My connection is great, so that must be out of the question. I have tried calling it from another Mac and from an iPhone, but only then does it cooperate. I don't want to downgrade the FaceTime that came with my MacBook, so can someone help me out?

coxorangepippin wrote:
The message that comes up when I click on "TomTom Home Install" is "You cannot open Home Install because Power PC applications are no longer supported." Does this mean that the TomTom, which is absolutely vital to my life - well, not literally but you know what I mean - can no longer be used with my MacBook Pro?
What that means is that is can't be used with Lion 10.7. PowerPC was the chip Apple used up until 2006 when they switched to Intel. At that point, they created the Rosetta emulator to allow for PPC support for 5 years. They finally dropped the support with the release of Lion. It will only work with OS X 10.6 or earlier. The other option is to purchase a new GPS, which can be had for $100 for a decent one.

PAP2T-NA will not make outgoing calls with Asterisk

I have a PAP2T-NA connected to a Uniden cordless phone and an Asterisk server with working trunks. Inbound calls work fine - the phone rings and sound works both ways. However I am unable to make outbound calls through the PAP2T, even to other local extensions or just to voice mail. After dialing an extension, I hear a short pause and then a fast busy signal.
Dial plan is (*xx|x.). This is all I see with syslog when calling extension 1000 through sip1 (* server):
Feb 26 08:13:51 LinksysPAP [0]Off Hook
Feb 26 08:13:56 LinksysPAP Calling:1000@sip1:0
Feb 26 08:13:56 LinksysPAP [0:0]AUD ALLOC CALL (port=16440)
Feb 26 08:13:56 LinksysPAP [0:0]RTP Rx Up
Feb 26 08:13:56 LinksysPAP RSE_DEBUG: reference domain:sip1
Feb 26 08:13:57 LinksysPAP RSE_DEBUG: reference domain:sip1
Feb 26 08:13:57 LinksysPAP [0:0]AUD Rel Call
Feb 26 08:13:57 LinksysPAP CC:Failed w/ Calling
Feb 26 08:13:59 LinksysPAP [0]On Hook
This is all I see in the * log:
Feb 26 08:16:12 DEBUG[29924] acl.c: ##### Testing 192.168.1.179 with 192.168.1.0Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Setting NAT on RTP to 0
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Stopping retransmission on '[email protected]' of Response 101: Match Found
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Setting NAT on RTP to 0
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Checking SIP call limits for device 100Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Stopping retransmission on '[email protected]' of Response 102: Match Found
I've tried various codecs and regional settings to no effect. Setting an outbound proxy makes no change. Is there something simple I'm missing? Thanks!

HI.....
           Is your voice provider Vonage then Vonage Supports 7-,10- and 11- digit dialing.Use 7-,10- or 11- digit dialing for calls within the same area code as your Vonage phone number. Use 10- or 11- digit dialing for calls outside of your area code.
           Also forward the ports 53, 69, 5060-5061and 10000-20000 for the adapter  IP address.

Outgoing calls with my iphone4 are mute but incoming calls ok?

Problem has just arrived with my iPhone 4s. thus i can only use my iPhone if somebody gives me a call. Has anybody made similor a experience? Is there any Chance for repair?
thx pitti

Try resetting your iPhone 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.
Next would be a Restore with your backup via iTunes.
You might also try a new SIM card.
Does it work on Speakerphone? If so, maybe the earpiece speaker is shot.

No audio in on outgoing calls with the spa112 ata

Hello,
Everything works fine if I forward ports but I don't want to do that for security reasons so I'm guessing its a NAT issue. Stun is not helping. Can anyone help? the NAT support paramaters in the SPA112 configuration look interesting but incomprehensible to me. I enabled logging and this is what I got:
Jun 14 13:51:29 SPA112 kern.err [17179569.412000] physmap-flash physmap-flash.0: map_probe failed
Jun 14 13:51:29 SPA112 kern.err [17179571.384000] ksz8873 0-005f: failed with status -1
Jun 14 13:51:29 SPA112 kern.err [17179579.468000] br0: Dropping NETIF_F_UFO since no NETIF_F_HW_CSUM feature.
Jun 14 13:51:29 SPA112 kern.err [17179604.692000] Created coma thread for processing coma control message
Jun 14 13:51:29 SPA112 kern.err [17179637.320000] Created coma thread for processing coma control message
Jun 14 13:51:29 SPA112 kern.err [17180921.884000] Created coma thread for processing coma control message
Jun 14 13:51:29 SPA112 kern.err [17185305.068000] Created coma thread for processing coma control message
Many thanks

Hi
I've had a look at the postcode and there has been some planned works on 3 or 4 masts in the area. The work is showing to be completed by the end of today.
Once planned works are completed it can take a few days for any updates to complete fully.
If you are still having problems over the weekend please Contact Us and speak to our tech team so they can investigate it for you.
Thanks.

Contact groups and outgoing call Protocol

I have two Problems with phone essentials:
1. making contact groups without external App.
2. Getting a Detail Protocol for outgoing calls ( with Time Budget) without ext. App.
Any idea?
Thank you

1) To the best of my knowledge this is not possible.
2) If I understand what your asking, go to Phone>Recents and tap on the blue arrow at the right of each call listing. It will tell you the time and duration of the call.
Best of luck.

OUTGOING CALL LOG

HOW DO U FIND UR OUTGOING CALL LOG ON VERIZON.COM, IM TRYING TO LOOK AT MY HOME PHONE OUTGOING CALL LOG

I've been nagging Verizon on this one alot. If you have one of the international long distance plans, you get 300 or 500 minutes of international long distance, then any time over this allotment is billed by the minute. The problem is that you have NO WAY of knowing how many minutes you have used, until the bill comes at the end of the month.
I've had customer support say:
- You need to track the time yourself - we suggest a stop watch.
- I get several calls a month about this , but we don't have an answer.
- We are trying to resolve this issue.
- You don't need to know how many minutes you used. (They indirectly imply this)
- We can see how many minutes you used, but can only tell you to resolve a fraud alert issue regarding unusual long distance activity. (This bugs me as much as suggesting I use a stop watch)
- If you switch to the more expensive 500 minute plan, you won't have to worry as much.
Call assistant would be the logical place to put outgoing calls, with international plan minute totals, but what do I, a customer, know.
CM
Message Edited by ComposMentis on 05-18-2009 07:26 AM
Message Edited by ComposMentis on 05-18-2009 07:27 AM
Message Edited by ComposMentis on 05-18-2009 07:42 AM

Setting a time limit for outgoing calls SPA3102

I need to put a time limit, say, 5 min, on outgoing calls with a certain prefix (are code). Is it possible with SPA3102 dialplan or otherwise? Thanks.

Based on my best knowledge - no, it's not possible with just SPA3102 only.
It is switch (PBX) job to terminate the call.

Handsfree bluetooh outgoing call delay

Here is a big issue of mine.
When using my phone to make outgoing calls with my car's built in bluetooth, the ringing is not heard through the car at all...and its not till 5-10 seconds after someone answers the phone that it is transmited through the cars speakers. Very odd. Most people are hanging up before the connection is finally made. They tell me they kept saying hello hello hello, but nobody is there. If they wait for more than 10 seconds after they pick up, then the bluetooth connection is finally made into the cars speakers.
Incoming calls pick up right away.
I have had many bluetooth phones with my car....never a problem...until now.

That's really odd, but since I'm not seeing it with a Plantronics 510 bluetooth headset I'm thinking it's something between your car and the iPhone specifically.
I always press the button to wake up my bluetooth connection (what once would have initiated voice dialing) before dialing on the iPhone. Is there a similar action you could try? I'm thinking that perhaps your car is not realizing at first that dialing is occuring, since you say it's only on outgoing, which on an iPhone is manual.
If you're still having problems you might want to start a new thread with your car name in it to find other people with the same car and see if they've solved it.
Aym

IPhone 4s 6.1.3 problems with 3g,In and Outgoing Calls

Hi Everybody!
I got a factory unlocked (originally was Vodafon locked) iPhone 4s with firmware 6.1.3.When I insert my sim (Hungarian Telenor) when only EDGE is on it founds the signal and seems like everything is working.After a few hours or sometimes even a day passes by a phone stays on full signal with Carrier logo and Edge on, but my friends reporting that they cant call me(like im offline) and i cant make calls either.The interresting stuff is that the Edge connection is fully working during that im offline for my friends via Calls and i can send and recieve imessages and viber messages.
When i switch 3g on it disconnects from service drops the EDGE and never finds the way back and stays 'searching' or 'no service' at all..
I thought that its my Carrier's fault so i went to spoke with them and tried 5different micro Sims( cause i wrote that 4s-s are sensitive to SIM card types).. this thing not worked either,
Then i tried the phone with a T-mobile SIM card.. it does the same..connects to the services,but with 3g switched on not finding service..so totally like with my sim.
Here Comes the bumm... i Tried the phone with vodafone sim and everything works.. simple connectivity to 3g with full internet usage,can make and recieve calls...I spoked with all of the Carriers and also Apple and no body can give me nothing..the phone is off varranty so changed to a new one isnt an option..
Anybody? Something? im beginning to give up .. please help if you can..Thanks!!
ps.:i restored over 20times,reset network settings,switch off and on automatic setting time.. every little possible things that people wrote on forums before..

neverstop wrote:
1. Settings > General > About - Carrier sais Carrier 14.0 when i restore and startup with my Telenor sim.. ..
1. In order
2. ...... everyone is pointing on the other and i cant do anything about it..just the whole 3g/call failed stuff is bothering me...
2. It appears configuration issue. I use three iPhones (in my family) on four unsupported carriers. Never had any problem except on one. Outgoing calles to one particular party failed. Set right by Carrier.
3. Is there something like the Unlock process was not fully succesfull...
3. Never heard. It is digital : Yes or No. In the instant case, unlock is successful.
Download firmware afresh via iTunes or from here iPhone4,1_6.1.3_10B329_Restore.ipsw
and save. Insert Telenor SIM in the iPhone. Backup. Option+Restore (Mac) Shift+Restore (PC) pointing to saved firmware. Reject Carrier updates, if any offered by iTunes. Setup as New.
Observe.

Automatic mute in outgoing calls

Hi all,
after upgrade to V6 bundle 1879 mute goes on automatically in all outgoing calls (Curve 9300).
Someone has some suggestions?
Thanks in advance

Hello,
In this case we can try to reload the software on the BlackBerry smartphone to make sure it is running the latest version and there are no issues with the OS.
Here is how to backup your BlackBerry smartphone http://bit.ly/aSediX
Once you have backed up your BlackBerry smartphone please follow the link below to complete a clean reload of the BlackBerry smartphone software.
Link: http://www.blackberry.com/btsc/KB11320
Once completed test it and proceed with selective restore here is how to restore http://bit.ly/bSbfl3
Thank you
-DrP
Come follow your BlackBerry Technical Team on Twitter! @BlackBerryHelp
Be sure to click Kudos! for those who have helped you.
Click Solution? for posts that have solved your issue(s)!

When making an outgoing call from my i phone, an icon of a microphone with a slash acroos it appears. I can hear the receiving party speak, but he cannot hear me. What has been shut off and how do I turn it on?

When making an outgoing call from my i phone, an icon of a microphone with a slash across appears. I can hear the recipient speakin but he cannot hear me. What has been turnedd off, and how do I turn it on?

I think mute is on, tap the microphone icon with the slash running through it

Automatic outgoing calls with CUE?

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