Displaying 20 results from an estimated 9000 matches similar to: "Different ringing tones ..."
2008 Apr 24
1
No CallerID Transfer Problem
Came upon a problem today that I thought I'd see if it's by design, if
I'm missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with
others, but for some reason when they would transfer an outside call
anywhere it would pause for a few seconds and hang up the line.
Well, after spending most of the day on
2007 Oct 14
5
AA50 Paging
Hi
I just got an AA50 from Digium and the paging command reboots
asterisk when you use it. Digium says it is a requested feature and is
of low priority. Is there any other way to page 10 Grandstream gxp2000
phones with meetme or some other command than the page command.
Thanks in advance.
Kelly
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2005 Jul 26
2
Stumped on vMail problem, any ideas?
Hello all,
I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow
something is not quite right with my vMail setup. I would have sworn this was
all working, but maybe I was just dreaming.
Anyway here is what is happening, say I am on extension 200 and I want to
call to extension 201. If extension 201 is no connected, then it rolls right
into vMail with the message the
2006 May 26
1
Not able to make any calls
Hi All,
I have registered "abhijit" for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name "abhijit".
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI>
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120
May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:
2006 Apr 18
1
Granstream GXP2000 Distinctive tones
I recently posted a question RE the Sipura 941 and using different ring
tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to
achieve this but trying this on the GXP 2000's didn't seem to do the
trick?? Has anyone one had any luck on this topic?
Also haven't been able to find any info on an auto-answer for the GXP
2000, again, I have succeeded in doing so with the
2009 Apr 09
2
notifyringing=no does not work
"
Hello,
I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it.
Here is how i have my subscriptions setup:
extensions.conf
[demo]
exten => 6100,hint,SIP/100
exten =>
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2014 Oct 22
1
SPA504G auto answer
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging). I
have tried the following SIP headers (not all together), but without luck:
SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);
Any other ideas?
Leandro
PS
I have set
2007 Jan 27
1
How to fix error when paging
I am trying to page my Grandstream GXP-2000 phones
and keep getting the error message:
Jan 27 12:55:04 WARNING[30401]: app_page.c:183 page_exec: Incomplete
destination '' supplied.
How can I fix this error?
The two contexts below do either one-way paging or two-way paging to all
Grandstream phones in a list.
[One_Way_Page_GROUP] ; one to many page
exten =>
2006 Jan 27
7
AAH out bound routing problem
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try your call later
and when i see in the console i get this mesage
any help
Called easycall/19197543700
2008 Apr 04
4
Advice on best operator phone (with attendant console)
One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and call
volume. We get the following problems randomly:
1. menu stops working
2. transfer key stops working
3. Line 1 LED gets
2007 Sep 25
4
Grandstream GXP2020 / 2000
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on .... not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Thanks!
Kind Regards,
Erik
2012 Jul 13
8
How to set SIP to auto answer in the dial plan .
Hi,
I am trying to write dial plan for sip to auto answer (auto attend) the
incoming call to the sip phone.
- If i call from sip1 to sip2 then sip2 should automatically answer the
call and play some sound file.
I am trying to do this but as new to the asterisk dial plan configuration ,
so not able Todo this.
help me if anyone already done this setup.
Regards
Upendra.
-------------- next part
2010 Jul 28
1
Random DTMF Tones Only on heard on ATA
I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are
receiving random DTMF tones on their side, but that are not heard by the
outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have
always had this issue. I am only using SIP on the Asterisk server and all
extensions and trunks are set to rfc2833; outside of this issue DTMF
operation works fine.
2005 Mar 14
18
Grandstream GXP-2000
FYI, spoke with Grandstream this morning, the GXP-2000 release has been
delayed again. Looking like April now before these hit the street.
--
Cory Andrews
Senior Partner
VOIPSupply.com
+++++++++++++
V: 800.398.VOIP X22
F: 716.630.1548
E: Cory@VOIPSupply.com
2009 Jan 16
4
Snom 300 vs Grandstream gxp
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
Julian
______________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email
2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
2009 May 05
4
AMI + AGI for outbound click to dial
Hey Gang,
Trying to figure out how I can do the following (have each part working
individually but drawing a blank on combining)
1) click on-screen which sends an AMI originate (works fine)
2) the originated call is to an internal extension that looks up the number
to be dialed (works)
3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works
separate from the above)
What I
2015 Mar 23
1
Auto Answer
Hi , I'm having some problems with functions enable auto answer in
some Grandstream GXP 1405 , I have enabled this feature in the snom
821 phone and work gr8 , in the gandstream not work, I enable the
function on the phone
"Allow Auto Answer by Call-Info: yes
Dialplan:
exten => 501,1,SIPAddHeader(Call-Info: answer-after=2)
exten => 501,n,Page(SIP/140&SIP/110,d)
exten