Displaying 20 results from an estimated 3000 matches similar to: "caller id ring tones for Asterisk Phone"
2006 Jan 24
1
cannot change distinctive ring polycom phones
Hi,
I'm using asterisk 1.2.1 on a debian sarge distro.
I've followed notes in
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
and
http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO
but I still cannot change ring style via asterisk using
exten => 666,1,SipAddHeader(ALERT_INFO="ring3")
in extensions.conf .
Is it
2006 Mar 29
5
Problem with setting ringtones on Cisco 7960 phone.
Hi All,
I am running into a problem setting the ringtones via _ALERT_INFO on the
Cisco 7960 phone.
I am using * 1.2.1 and have tried setting the variable to several
values. I have also tried setting the phone's software to both 7.5 and
8.2 thinking that it might be a version issue, but with no success.
I have examined the packets and do see the ALERT_INFO header being sent,
but the
2004 Aug 31
3
Cisco 79XX SIP Ring Tones
Hi all,
Has anyone gotten custom ring tones to work using ALERT_INFO with the
Cisco 7940 SIP phone? I've read the wiki, but just can't get this to
work. I'm currently using the 7.2 SIP image.
Thanks,
Chris
2006 Jan 25
4
Setting ringtone on Polycoms
Hi,
I'm having trouble setting the ringtone on my Polycom 501.
The relevant entry in extensions.conf is:
exten => 801,hint,SIP/creative1
exten => 801,1,SetVar(ALERT_INFO="Test")
exten => 801,2,Dial(SIP/creative1,20,Ttr)
In the sip.cfg:
<alertInfo voIpProt.SIP.alertInfo.1.value="Test"
voIpProt.SIP.alertInfo.1.class="13"/>
and
<TEST
2007 Feb 27
5
TE110P: Error ==> Asterisk died with code 1.
Running Asterisk 1.2.9. I just installed a TE110P card and configured
zaptel.conf & zapata.conf. The config files look right to me but I'm
getting the following error when trying to start asterisk:
Asterisk died with code 1.
Automatically restarting Asterisk.
Does anyone have any idea what is wrong with this configuration??
Thanks in advance!!!
Here's my config files:
zaptel.conf
2008 Jan 10
8
IEEE 802.1x capable sip phones
Does anyone know if sip phones from any of the major IP phone vendors
support 802.1x authentication? Any feedback would be greatly
appreciated.
Thanks in advance.
======================
Jeronimo Romero
EUS Networks
Email: jromero at euscorp.com
Cell: 917-332-7238
Office: 212-624-5943
Web: www.euscorp.com
======================
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2006 Nov 20
1
alert_info + Linksys 9xx + custom ringtone
Hello,
I have uploaded a custom ringtone to our SPA-922's for the purpose of
sounding like a door bell chime when the doorbell is pressed. I am using
__alert_info to set this ringtone. It appears that I can only set the
ringtone via alert_info for the ringtones that come from Linksys. Has anyone
else seen this issue?
I am doing the following:
exten => 100,1,SetVar(_ALERT_INFO=doorbell)
2006 Oct 29
4
blind transfers with IP Polycom 501
I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The
only problem I'm experiencing is the following: I can't seem to get
blind transfers to work with my Polycom 501 phones Either through the
feature code or the soft keys.
Feature code blind transfers:
I set up a feature map in features.conf like this:
blindxfer => #
This works for all my
2004 Sep 09
3
Polycom IP500 vs Cisco 7940
Hi Everyone,
I've been asked to determine which phones our organization should go
with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940.
From my travels through google, it's hard to find a definitive
comparison of the two phones. So I thought I would ask the people that
have probably used both.
From what I can tell, the only major benefit the Cisco has over the
2006 Nov 09
3
announcing inbound PSTN calls
I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the
following:
1-once PSTN callers enter their desired extension; they have to record
their name
2-recording then announces that it is trying to locate the user
3-asterisk calls local extension and announces callers recorded name
4-local recipient user can choose to take the call, send it to voicemail
or transfer it to
2006 Nov 15
2
Grandstream GXP2000 -- What's the Catch?
We are doing a medium sized office in NYC with 80 phones. The customer
originally requested Polycom 601 phones. The COO also authorized us to
purchase 2 Grandstream GXP2000 phones for the mail room. We find these
phones much easier to configure and work with asterisk . They support
BLF & intercom right out of the box. They can also be centrally managed
and provisioned. They also sound great
2007 Mar 18
6
T1 cable for Digium T1/E1 Cards
Is there any technical difference between a T1 cable and a cat5e patch
cable as far as using them with Digium T1/E1 cards? Can PRI circuits
terminating at a smart jack connect successfully to Digium cards using
straight through CAT5e cables? If so, are they using all of the pins in
the cable?
Thanks in advance
2006 Dec 11
2
asterisk PLAR
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?
Thanks in advance.
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2007 Feb 05
2
asterisk server as a voicemail server for legacy PBX -- FXO or FXS???
Hey All,
I'll be configuring an asterisk box to be the voicemail server to an old
Merlin system which had an octel 100 voicemail server that is now dying.
My question is simple: do I need to stick an FXO card in the asterisk
box? My logic is that if the Merlin Magix system is actually generating
electrical current, then I would need to have an fxo card. Is this
correct?
2006 Nov 12
2
same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension is
the only one that rings when the
2003 Oct 10
2
ALERT_INFO=1/ Cisco 79x0
Hi,
I've just found:
http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html
which talks about ALERT_INFO and Cisco phones. How do I actually get
this working and what does it do? Do I need to add anything to the configs
for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it
seemed to do nothing at all..
Thanks
Andy
2003 Dec 30
3
SIP phone as intercom
(new asterisk user - currently setting up Polycom IP600 phones)
Does anyone know if it's possible to make a sip phone instantly pick up
on speakerphone when a particular call comes in? Eg so that you can
quickly bother someone across the office without making them reach for
their phone?
2006 Oct 30
1
dealing with blind transfers to invalid extensions
Running Asterisk 1.2.8 kernel 2.6.13.4-1.
Everything in my dialplan seems to be working well except for one
problem.
When calls are blind transferred to an invalid extension I would like
the call to go to the operator on ext 1000?
What is the best way to do this? Thanks in advance
Here's a snippet of my extensions.conf
[default]
exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
2007 Feb 06
1
asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
Thanks. Is there a way I can log into the Merlin Magix to determine
that? How else do I tell?
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric
Germann
Sent: Monday, February 05, 2007 8:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users]
2006 Nov 21
2
FW: CISCO 7960G & Asterisk
I was wondering if people have experienced issues with Cisco 7960G and
Asterisk.
Any feedback on people's experience deploying this phone in production
environments would be appreciated.
Thanks in advance