Wake Up Call to Extension

Hey Guys,
Does anybody know of a free service that allows you to send wake up calls to an extension through specified phone number? It would really help me out a lot.
Thanks!

You can use my script "alarm call". for CUCME and CME. It can be acquired on the website mentioned in my profile.

Similar Messages

How to make "efficient" wake-up call?

Hi,
Are there any means to force I-Phone to ring longer than 2 seconds for wake-up call? I can not find any setting to make wake-up call ezcepted "meeting" or so in the calendar. Is ther something you can suggest?
Thanks

Is it possible to make an alarm for an event in the calendar app continue to ring until you shut it off?

UC560 Auto attendant calling fxs extension

Hi,
I have an UC 560 with an autoattendant working fine, but when from the pstn need to call to an extension of a analog extension in a FXS port it says that is not a valid extension.
If i check the option: "Allow external transfers" for the autoattedant it works, but in the told fraud prevention practice don´t recommended.
Is required this option to allow the call to the fxs port direct from pstn or what can be wrong in configuration?
Regards,

Hello John,
The configuration for the fxs port is:
voice-port 0/0/0
station-id number 115
caller-id enable
dial-peer voice 1 pots
corlist incoming user-local
translation-profile incoming CallBlocking
destination-pattern 115
port 0/0/0
no sip-register
If i assign a option in the autoattendant to call the extension this work, but i need to call directly to the extension
The prompt messages says: "if you know the extension number, please dial it"
In that moment if i call any sscp extension it works, but calling the extension of the fxs port says that is invalid
I did a test calling directly a hunt group extension with the same effect.
To ilustrate:
FXS extension: 115
Hunt Group Extension: 170
With no allowed external transfer:
Call to AA and dial 115 -> No works
Call to AA and dial 170 -> No works
Assign 1 number of AA to 115, and 2 number to 170
Call to AA and dial 1 -> works
Call to AA and dial 2 -> works
With allowed external transfer:
Call to AA and dial 115 -> works
Call to AA and dial 170 -> works
Assign 1 number of AA to 115, and 2 number to 170
Call to AA and dial 1 -> works
Call to AA and dial 2 -> works
I hope to be clear,
Regards

I was looking to open a WAB document. A page called File Extensions recommended downloading something called MacKeeper. Any experience/advice on this?

I was looking to open a WAB document. A page called File Extensions recommended downloading something called MacKeeper. Any experience/advice on this?

Stay away from MacKeeper like you'd avoid the plague.

TS3938 The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?

The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?

Scot Lee wrote:
The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?
Yes Rosetta has been gone for over two years. You can always search here for how Michael Lax's instructions on how to run SL Server in a virtul environment, but I don't think it is too kind to gamers.
Advice? Buy an old Mac that runs PPC software.
Cheers
Pete

ROBO call blocker is blocking my wake up calls

I've had wake up calls for 3 years using my FIOS landline voice mail. I also Implemented robocall blocker about a year ago.
Within the past week, the blocker is now stopping the wake up calls.
Verizon, how do I get around this? I've added the voicemail number that calls with the wake up call to my accepted list,
still no resolution.

dezyndiva wrote:
that's part of the problem. I don't recall now how I set it up. However, It's illogical that I would have to turn it off
just to get wake up calls. Why now is the blocker recognizing the wake up call as coming from a robodialer,
it never blocked it for the past year.
With FiOS digital voice, there is no native "ROBO call blocker" so you must be using something else. If you're using something like nomorobo then it uses the simulatenous ring feature which can be disabled if nomorobo (or some other service like it) is blocking the call. You check your simulataneous ring setup.

Wake up calls

I have a new landline number and for some reason there is a wake up call set for 5am. I have already called once and the agent could not find it on the account. i had just set up the voice mail. The agent said that should clear out any previous setting. I am still getting these wake up calls and I dont know how to take them off. There is no menu option to delete a wake up call. There is only options to set up wake up calls.. Please help...

There is nothing really built in these apps to do that easily, but there are plenty of 3rd party solution that integrate with CUCM for hospitality market. Novotek comes to mind.
Chris

CUCM Wake-Up Call

I need a product or feature to schedule a call to an IP Phone for use as a Wake-Up call for hotel guests.
This can either be scheduled by the guest themselves or by reception.
IPCelerate sounds like it could offer this, but the price says no ($31k list).
Has anyone else come across anything similar?

Do you still have this? I am new to CUCM and trying to figure out how to do this. Visited your site, but nothing seems to be free... Would you be able to provide a sample of how this is done or a setup guide on how to implement what you have created? Thanks in advance

Ads keep popping up, saying I have an extension called 'None Extension'. I looked for it in add-ons under extensions and plug-ins but I didn't find it.

I looked for an add-on I supposedly have on Firefox called 'None Extension' with no luck. Ad's continue popping up all the time, saying they're powered by this extension. Google won't show me anything about it either and I didn't find it under extensions or plug-ins in add-ons but I didn't find it. What should I do?

You can check for recently installed suspicious or unknown extensions.
*https://support.mozilla.org/kb/Troubleshooting+extensions+and+themes
You can do a malware check with several malware scanning programs on the Windows computer. 
Please scan with all programs because each program detects different malware. 
All these programs have free versions.
Make sure that you update each program to get the latest version of their databases before doing a scan.
*Malwarebytes' Anti-Malware: http://www.malwarebytes.org/mbam.php
*AdwCleaner: http://www.bleepingcomputer.com/download/adwcleaner/ http://www.softpedia.com/get/Antivirus/Removal-Tools/AdwCleaner.shtml
*SuperAntispyware: http://www.superantispyware.com/
*Microsoft Safety Scanner: http://www.microsoft.com/security/scanner/en-us/default.aspx
*Windows Defender: Home Page: http://www.microsoft.com/windows/products/winfamily/defender/default.mspx
*Spybot Search & Destroy: http://www.safer-networking.org/en/index.html
*Kasperky Free Security Scan: http://www.kaspersky.com/security-scan
You can also do a check for a rootkit infection with TDSSKiller.
*Anti-rootkit utility TDSSKiller: http://support.kaspersky.com/5350?el=88446
See also:
*"Spyware on Windows": http://kb.mozillazine.org/Popups_not_blocked

How to call Indesign Extension from C/C++

I have created Indesign extension using creative suite extension builder from my flash builder 4.6. In that extension i have "Run Indesign" button.How can i call that button from C/C++?

@T. Scheider
For Adobe CC 9.0.x-9.2.x, do I need AEDP access to download native application toolkit to compile the flex hybrid extensions?
Also is native application toolkit different for all versions of Adobe Indesign?

Call voicemail extension

Hi,
I use spa502g ip phone with spa500s connect to an asterisk server. I want to know, how I can call the voicemail of extension 7999 directly without the choice of the extension? I try to insert
fnc=sd+cp+blf;sub=*987999@pbxIp
in the attendant console, but it's the same than when I call *98 without the number of extension...
Thank you and sorry for my bad english

Ok, I try with fnc=sd;ext=*987999@myIp but it still doesn't works, I always be redirect to simple *98 service.
This is my spacfg.xml
Static IP
192.168.100.242
ciscospa502g
255.255.255.0
192.168.100.254
212.27.40.240
212.27.40.241
SPA502G
CCQ17040C92
7.5.2
1.0.4
E02F6D629A01
Installed
Open
None
9/17/2013 15:56:22
00:13:44
4
168
670
67029
0
0
301
48160
211
33760
40
19601
39
22350
N/A
100M Full Duplex
Link Down
Registered
9/17/2013 15:54:21
113 s
No
Idle
None
G711u
G711u
Outbound
No
*98
00:00:01
301
211
48160
33760
70 ms
0 ms
Not Available
0 ms
0
0
0
0
0 ms
0 ms
Idle
None
Idle
Not Installed
Not Installed
Not Installed
Yes
80
Yes
SIP
Yes
No
No
Normal
Static IP
192.168.100.242
255.255.255.0
192.168.100.254
ciscospa502g
212.27.40.240
212.27.40.241
Manual
Parallel
No
0
No
Yes
No
Yes
Yes
3
1
No Limit
No
1
70
5
2
$VERSION
$VERSION
application/dtmf-relay
application/hook-flash
No
No
No
No
No
No
No
No
Yes
No
Yes
5060
5080
No
PAID-RPID-FROM
x-sipura
No
No
No
No
.5
4
5
16
16
16
16
16
240
30
1
7200
30
1200
10
7200
10
10001
10040
0.020
0
0
No
No
No
101
98
97
2
96
99
112
113
G711u
telephone-event
PCMU
PCMA
G726-16
G726-24
G726-32
G726-40
G729a
G729ab
G722
encaprtp
No
No
No
No
No
No
No
No
15
No
No
224.168.168.168:6061
Yes
none
Yes
Yes
Yes
2
600
3600
3600
14400
Yes
Yes
Yes
/spa$PSN.cfg
66,160,159,150,60,43,125
https
$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH
$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH
$PN $MAC -- Resync failed: $ERR
Yes
Yes
3600
$PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH
$PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR
$PN $MAC -- Upgrade failed: $ERR
350@-19,440@-19;10(*/0/1+2)
420@-16;10(*/0/1)
520@-19,620@-19;10(*/0/1+2)
480@-19,620@-19;10(.5/.5/1+2)
480@-19,620@-19;10(.25/.25/1+2)
480@-10,620@0;10(.125/.125/1+2)
440@-19,480@-19;*(2/4/1+2)
440@-10;30(.3/9.7/1)
600@-16;1(.25/.25/1)
985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)
350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)
600@-19;25(.1/.1/1,.1/.1/1,.1/9.5/1)
350@-19;20(.1/.1/1,.1/9.7/1)
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)
600@-16;.3(.05/0.05/1)
600@-19;.2(.05/0.05/1)
440@-10;30(.3/9.7/1)
60(2/4)
60(.3/.2,1/.2,.3/4)
60(.8/.4,.8/4)
60(.4/.2,.3/.2,.8/4)
60(.2/.2,.2/.2,.2/.2,1/4)
60(.2/.4,.2/.4,.2/4)
60(4.5/4)
60(0.25/9.75)
60(.4/.2,.4/2)
255
1800
30
.5
10
3
*69
*66
*86
*72
*73
*90
*91
*92
*93
*56
*57
*71
*70
*67
*68
*81
*82
*77
*87
*78
*79
*16
*17
*18
*19
*96
*38
*36
*39
*37
*03
*017110
*027110
*017111
*027111
*01722
*02722
*0172616
*0272616
*0172624
*0272624
*0172632
*0272632
*0172640
*0272640
*01729
*02729
GMT+01:00
Yes
Yes
-16
.1
12dB
ISO-8859-1
en-US
CISCOSPA502G
CISCOSPA502G
*97
Default
Text Logo
Auto
No
300
Background Picture
1
$USER
private
2
Scrollable
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
1
No
n=Classic-1;w=3;c=1
n=Classic-2;w=3;c=2
n=Classic-3;w=3;c=3
n=Classic-4;w=3;c=4
n=Simple-1;w=2;c=1
n=Simple-2;w=2;c=2
n=Simple-3;w=2;c=3
n=Simple-4;w=2;c=4
n=Simple-5;w=2;c=5
n=Office;w=4;c=1
n=Pulse;w=5;c=1
n=Du-dut;w=6;c=1
0
0
0
0
0
0
pggrp=224.168.168.168:34560;name=All;num=800;listen=yes;
No
Enterprise
No
None
Trusted
No
No
em_login|1;acd_login|1;acd_logout|1;astate|2;avail|3;unavail|3;redial|5;dir|6;cfwd|7;dnd|8;lcr|9;pickup|10;gpickup|11;unpark|12;em_logout
lcr|1;miss|4
redial|1;dir|2;cfwd|3;dnd|4;lcr|5;unpark|6;pickup|7;gpickup|8;starcode|11;alpha|12
dial|1;delchar|2;clear|3;cancel|4;left|5;right|6;starcode|7;alpha|8;dir
endcall|2
hold|1;endcall|2;conf|3;xfer|4;toggle;bxfer;confLx;xferLx;park;phold;flash;
hold|1;endcall|2;xfer|4;toggle;
hold|1;endcall|2;conf|3;toggle;
hold|1;endcall|2;join|4
endcall|2;
resume|1;endcall|2;newcall|3;redial;dir;cfwd;dnd
answer|1;ignore|2;toggle|4
newcall|1;barge|2;cfwd|3;dnd|4
resume|1;barge|2;cfwd|3;dnd|4
Yes
private
3600
No
No
No
$NOTIFY
$PROXY
0x68
3
0xb8
6
high
up and down
UDP
5060
No
Yes
No
4
No
0
none
0
No
No
Yes
Yes
none
No
No
No
No
4
86400
No
No
No
Yes
No
No
192.168.100.240
No
Yes
Yes
No
300
No
No
No
3600
Normal
No
CISCO
8001
No
G711u
No
G711a
Unspecified
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Auto
0
0
No
Default
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Yes
No
20
Yes
No
No
No
No
No
Yes
Speaker
No
12hr
month/day
Yes
Yes
automatic
source
media
Yes
No
No
Yes
Yes
9
8
10
10
Auto
Default
Yes
8
10 s
1800
30
Yes
1
Yes
Asterisk
No
*8
*68
*88
Yes
12
7
fnc=sd+cp+blf;[email protected]
fnc=sd+cp+blf;[email protected]
fnc=sd;ext=*[email protected]
and this is the log of the call in asterisk
<------------->
[2013-09-17 16:42:40] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996
From: "CISCO" ;tag=5826b042144b7d5do0
To:
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "CISCO"
Expires: 240
User-Agent: Cisco/SPA502G-7.5.2
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 3640 3640 IN IP4 192.168.100.242
s=-
c=IN IP4 192.168.100.242
t=0 0
m=audio 10035 RTP/AVP 0 8 2 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (14 headers 18 lines) ---
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Using INVITE request as basis request - [email protected]
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found peer '8001' for '8001' from 192.168.100.242:5060
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.100.242:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996;received=192.168.100.242;rport=5060
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as09f233a1
Call-ID: [email protected]
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2847dbe8"
Content-Length: 0
<------------>
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
ACK sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-834fc996
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as09f233a1
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "CISCO"
User-Agent: Cisco/SPA502G-7.5.2
Content-Length: 0
<------------->
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (10 headers 0 lines) ---
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
INVITE sip:*[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66
From: "CISCO" ;tag=5826b042144b7d5do0
To:
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*[email protected]",algorithm=MD5,response="f4a1ef5ed0d7e5bec2c603a783fe04ff"
Contact: "CISCO"
Expires: 240
User-Agent: Cisco/SPA502G-7.5.2
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 3640 3640 IN IP4 192.168.100.242
s=-
c=IN IP4 192.168.100.242
t=0 0
m=audio 10035 RTP/AVP 0 8 2 9 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (15 headers 18 lines) ---
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Using INVITE request as basis request - [email protected]
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found peer '8001' for '8001' from 192.168.100.242:5060
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] netsock2.c: == Using SIP RTP TOS bits 184
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] netsock2.c: == Using SIP RTP CoS mark 5
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 0
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 8
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 2
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 9
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 18
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 96
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 97
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 98
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found RTP audio format 101
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format PCMU for ID 0
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format PCMA for ID 8
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G726-32 for ID 2
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G722 for ID 9
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format G729a for ID 18
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-40 for ID 96
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-24 for ID 97
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found unknown media description format G726-16 for ID 98
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Found audio description format telephone-event for ID 101
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Peer audio RTP is at port 192.168.100.242:10035
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: Looking for *98 in from-internal (domain 192.168.100.240)
[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: set_destination: Parsing for address/port to send to
[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: set_destination: set destination to 192.168.100.242:5060
[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: Reliably Transmitting (NAT) to 192.168.100.242:5060:
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK5b2eaf36;rport
Max-Forwards: 70
From: ;tag=as1ae3104c
To: "CISCO" ;tag=4b051e1ec62e863d
Contact:
Call-ID: [email protected]
CSeq: 103 NOTIFY
User-Agent: FPBX-2.11.0(11.5.0)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 207
<?xml version="1.0"?>
confirmed
[2013-09-17 16:42:46] VERBOSE[32503] chan_sip.c: == Extension Changed 8001[ext-local] new state InUse for Notify User 8001
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c: list_route: hop:
[2013-09-17 16:42:46] VERBOSE[32523][C-00000006] chan_sip.c:
<--- Transmitting (NAT) to 192.168.100.242:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66;received=192.168.100.242;rport=5060
From: "CISCO" ;tag=5826b042144b7d5do0
To:
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Length: 0
<------------>
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:1] Answer("SIP/8001-00000008", "") in new stack
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Audio is at 10032
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding codec 100004 (alaw) to SDP
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.100.242:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-6132ab66;received=192.168.100.242;rport=5060
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as466725aa
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 2104859674 2104859674 IN IP4 192.168.100.240
s=Asterisk PBX 11.5.0
c=IN IP4 192.168.100.240
t=0 0
m=audio 10032 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
SIP/2.0 200 OK
To: "CISCO" ;tag=4b051e1ec62e863d
From: ;tag=as1ae3104c
Call-ID: [email protected]
CSeq: 103 NOTIFY
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK5b2eaf36
Server: Cisco/SPA502G-7.5.2
Content-Length: 0
<------------->
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
ACK sip:*[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-e8d91c8c
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as466725aa
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*[email protected]",algorithm=MD5,response="f4a1ef5ed0d7e5bec2c603a783fe04ff"
Contact: "CISCO"
User-Agent: Cisco/SPA502G-7.5.2
Content-Length: 0
<------------->
[2013-09-17 16:42:46] VERBOSE[32523] chan_sip.c: --- (11 headers 0 lines) ---
[2013-09-17 16:42:46] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:2] Wait("SIP/8001-00000008", "1") in new stack
[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:3] NoOp("SIP/8001-00000008", "app-dialvm: Asking for mailbox") in new stack
[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] pbx.c: -- Executing [*98@from-internal:4] Read("SIP/8001-00000008", "MAILBOX,vm-login,,,3,2") in new stack
[2013-09-17 16:42:47] VERBOSE[32747][C-00000006] file.c: -- Playing 'vm-login.gsm' (language 'fr')
[2013-09-17 16:42:50] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: REGISTER
[2013-09-17 16:42:51] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: REGISTER
[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
BYE sip:*[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-def5f005
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as466725aa
Call-ID: [email protected]
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="8001",realm="asterisk",nonce="2847dbe8",uri="sip:*[email protected]:5060",algorithm=MD5,response="8a4e6470356a8e1ea82eb36413e682cf"
User-Agent: Cisco/SPA502G-7.5.2
Content-Length: 0
<------------->
[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c: --- (10 headers 0 lines) ---
[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c: Sending to 192.168.100.242:5060 (NAT)
[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c: Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)
[2013-09-17 16:42:52] VERBOSE[32523][C-00000006] chan_sip.c:
<--- Transmitting (NAT) to 192.168.100.242:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.242:5060;branch=z9hG4bK-def5f005;received=192.168.100.242;rport=5060
From: "CISCO" ;tag=5826b042144b7d5do0
To: ;tag=as466725aa
Call-ID: [email protected]
CSeq: 103 BYE
Server: FPBX-2.11.0(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] app_read.c: -- User disconnected
[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/8001-00000008", "") in new stack
[2013-09-17 16:42:52] VERBOSE[32747][C-00000006] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/8001-00000008'
[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: set_destination: Parsing for address/port to send to
[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: set_destination: set destination to 192.168.100.242:5060
[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: Reliably Transmitting (NAT) to 192.168.100.242:5060:
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK60445d1d;rport
Max-Forwards: 70
From: ;tag=as1ae3104c
To: "CISCO" ;tag=4b051e1ec62e863d
Contact:
Call-ID: [email protected]
CSeq: 104 NOTIFY
User-Agent: FPBX-2.11.0(11.5.0)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 208
<?xml version="1.0"?>
terminated
[2013-09-17 16:42:52] VERBOSE[32503] chan_sip.c: == Extension Changed 8001[ext-local] new state Idle for Notify User 8001
[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c:
<--- SIP read from UDP:192.168.100.242:5060 --->
SIP/2.0 200 OK
To: "CISCO" ;tag=4b051e1ec62e863d
From: ;tag=as1ae3104c
Call-ID: [email protected]
CSeq: 104 NOTIFY
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK60445d1d
Server: Cisco/SPA502G-7.5.2
Content-Length: 0
<------------->
[2013-09-17 16:42:52] VERBOSE[32523] chan_sip.c: --- (8 headers 0 lines) ---
[2013-09-17 16:42:58] VERBOSE[32523] chan_sip.c: Really destroying SIP dialog '[email protected]' Method: BYE
[2013-09-17 16:43:02] VERBOSE[32744] asterisk.c: -- Remote UNIX connection disconnected
Thank you

Call handler extension cant be dialed?

Hello
I setup a callhandler to be my AA...extension 5051....when i dial it from my phone i get that extension cant be dialed consult my directory...any ideas to whats preventing this? The phone im calling from has a full access CSS...
TIA

Do you have a dummy phone with extension 5051 created (and set to CFwdAll to vmail)?
Callmanager doesn't have any concept of the callhandlers on Unity unless you create a dummy phone or route point with the DN.

Wake Up Call

Just getting back from the holidays I'm beginning to hear a lot of stories from people who left Verzion Wireless. As a holder of a not insignificant amount of shares of VZ stock I find this very disconcerting. I also was one who just switched 4 phones from Verizon to AT&T. I was a long term Verizon customer. Verizon claims to have the best network and I believe that to be true. However we have found there really isn't anything wrong with AT&T's network in the areas we use it. Verizon is clearly OVERVALUING the quality of its network. Unless Verizon stops being Very Greedy, and starts valuing it's long term customers, the bleeding will continue. I think it's time to sell all of my VZ stock. (Special thanks to NATE in Newington, CT for letting me switch my 4 phones to AT&T). Wake up Verizon before you lose many more of your 'customers'.
- A former Verizon customer................

You have just been demote to Bronze. Contributor

Using mm50 and wake up call

I now exchanged my ipod classic with a nano 5G and wanted to use it same way in the morning for wake up as my classic. So I put the nano into my logitech mm50 at night with the wake up time set to 5:45am. But instead of being quite the nano wakes me every 20 minutes. This behavior is only shown when docked in the mm50, not if I use the build in loud speaker. And the classic didn't show this behavior either.
What did I do wrong?
Thanks and best regards
Anke

You have just been demote to Bronze. Contributor

The great 2 am wake up call

I'm really starting to get irritated and ****** off. Every day at 2 am my husbands phone resets over and over again until we power it off. From looking around I cannot find if there is a resolution to this or not. We both have the Droid incredible and it is only his phone that is doing it. My husband is at the point where he is ready to throw the phone in the garbage because of this issue. Can anyone tell me what I should try. He hasn't installed any new apps recently and this has been going on for a week.

check for new updates, it might be the phone trying to install a new update? my thunnderbolt did that when the 2.3 update came out...

Wake Up Call to Extension

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