Displaying 8 results from an estimated 8 matches for "callernumb".
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callernum
2008 Mar 02
0
Cisco 7970 - register with NAT phone
...gt;
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>7b452e87-4496-4762-e11f-b26751a1884b</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>9</featureID>
<featureLabel>label</featureLabel>
&...
2012 Jan 15
0
configuring a Cisco 7961 so that different line appearances register to different SIP proxy addresses
...<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>study_line1</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID&g...
2008 Jan 25
2
Unprovisioned 7961
Hi Everyone,
I am having some issues getting my 7961 working with Trixbox. I have loaded
the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an
unprovisioned state. A status message shows up and says ?Error Verifying
Config Info?.
I have read quite a bit on this topic (getting 7961?s to work with Asterisk
and TB) and only came across a few postings where other people
2011 Mar 02
1
Registering Cisco 7942G IP phone with Asterisk!.
Hi,
?
We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)).
?
Recently we have bought a cisco 7942G IP phone.
It currently has SIP 42.9-0-2SR1S firmware loaded on it.
We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk.
?
Do we need to
2011 Dec 19
0
ChanSpy in whisper mode - low quality audio
...o many cuts so that they cannot understand what is said in the audio file. Is this because the total RTP bandwidth is too much?. Which codec and format should I use for the files played with Playback application?
Dialplan is like this:
same => n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER}))
[macro-inject]
same => s,1,Originate(Local/trunk at injectWarning,app,Playback,Message-Callee)
same => n,Originate(Local/${ARG1}@injectWarning,app,Playback,Message-Caller)
[injectWarning]
exten => _+34[69]XXXXXXXX,1,Answer()
same => n, ChanSpy(SIP/${EXTEN},qw)
same =>...
2006 Jun 27
4
siemens pbx and asterisk
Hello all,
I'm new to asterisk. Our company wants to setup an asterisk server and will
eventually move to IP centric phones, but they don't want to just throw away
the old Siemens PBX, so during the process we want to integrate it with
asterisk. Is it possible? and how?
thanks.
Lito
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2011 Dec 15
3
Play audio file for both Caller and Callee in a call
Dear all,
Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee?
A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor.
Thank you!
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