To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Help with IAX Trunk
On 09:48, Sat 02 Dec 06, Dave Morrow wrote:
H.interesting thought. Not sure how to do it though...
I found this this morning. I think it might be the answer I seek
http://www.trixbox.org
I wonder if anyone has experienced an issue I have found with the
Linksys SPA-841 phone.
On my Asterisk (Trixbox 2), to login to a queue, a user must enter the
queue number, followed by the * key. This works fine on my Companies
mix of phones, with the exception of the Linksys (Sipura) SPA-841.
Hi all. I have an IAX trunk between 2 Asterisk servers. Everything is
working correctly dialing between the servers as well as through the
PSTN (a T1 connected to one of the servers).
The second Asterisk server routes all calls to the PSTN via the first
server. Calls to local 10-digit, and
-Commercial Discussion
Subject: Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote:
My long distance provider requires that a billing code be entered
after dialing a long distance call. From the directly attached
Asterisk server, these calls work when the user enters their PIN after
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Saturday, December 02, 2006 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with IAX Trunk
Dave Morrow wrote:
Unfortunately, the codes are private for the individual
I have a need to
have a single extension actually ring on 2 phone lines which are not extensions
(they are analog phone lines). Does anyone know a suitable extensions.conf
config for this?
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
Colp
Sent: Friday, July 28, 2006 6:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] One extension to ring on multiple outside
lines
- Original Message -
From: Dave Morrow
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial
] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it
should be somthing like this [featuremap]automon =
*1
--Giridhar Bandi
On 5/12/06, Dave
Morrow [EMAIL PROTECTED]
wrote:
Thanks
for the response.How would I change the DTMF transfer
mode?David MorrowTechnical
, May 12, 2006 3:41 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it
should be somthing like this [featuremap]automon =
*1
--Giridhar Bandi
On 5/12/06, Dave
Morrow
: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it
should be somthing like this [featuremap]automon =
*1
--Giridhar Bandi
On 5/12/06, Dave
Morrow [EMAIL PROTECTED]
wrote
automon = *1 in your features.conf ?? it
should be somthing like this [featuremap]automon =
*1
--Giridhar Bandi
On 5/12/06, Dave
Morrow [EMAIL PROTECTED] wrote:
Thanks
for the response.How would I change the DTMF transfer
mode?David MorrowTechnical Systems LeadAutodata
Hi all. I was
reading a sample config someone had posted relating to call forwarding, and in
it, they use a Dial command with components that I cannot find any reference
to.
Can someone point me
to a reference which could explain the difference between
Dial(SIP/100|20|Ttr,,wW) and
]
mailto:[EMAIL PROTECTED]
From: Dave Morrow Sent: Friday, May
12, 2006 10:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] features.conf *1 Call
Recording
It's quite strange. When I press *1 I do not hear a tone
indicated that it's even trying
632, Of. 67.
La Plata, CP B1900AMZ
Buenos Aires, Argentina.
Tel. +54 221 445 0244 Ext. 301
Fax. +54 221 445 0245
www.trestech.com.ar
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Dave Morrow
Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.
Para: Asterisk
Hi
all.
I am attempting to
setup Asterisk to allow me to press *1 while in a call to use automon to record
the call but have had absolutely no success. Is there a trick to
this?
In
extensions.conf
[globals]
DYNAMIC_FEATURES=automon
[default]
: [Asterisk-Users] features.conf *1 Call Recording
2006/5/10, Dave Morrow [EMAIL PROTECTED]:
I am attempting to setup Asterisk to allow me to press *1 while in a
call to use automon to record the call but have had absolutely no
success. Is there a trick to this?
May be a problem with the way you
Hi all. I posted this earlier but
nevergot any advice that helped. If anyone knows how to get this
going, I'd appreciate some
advice.
I am attempting to
setup Asterisk to allow me to press *1 while in a call to use automon to record
the call but have had absolutely no success. Is there a
Hi all, I was just
wondering ifanyone knows of any gotchas with respect to upgrading Asterisk
to the latest 1.2.7 ?
Is the procedure the
same? Config files remain intact? Just untar/make
install?
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
Hi all. I am
trying to find out what the most popular soft phone for Windows is for use with
Asterisk. SIP or IAX?
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get
Title: Call Transfer
Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
Thanks all for the replies.
I've narrowed it down to the phones dislike for my older 3COM switch. I
noticed on the weekend that when these missed calls occur, if I ping the
phone, the first few packets are dropped..almost like it's gone to
sleep..
David A. Morrow
Technical Systems
Title: Music on Hold Error
Can anyone help with;
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3'
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player
Dec 2 12:20:16 WARNING[2562]:
Title: Linksys SPA-841 Missing Calls
Hi all, I have been plagued by an issue with my SPA-841 phones. The issue is that frequently, usually after a period of inactivity on the phone, an incoming call will be missed by the phone. The call works, cause the caller ends up at voicemail, but the
Title: Linksys SPA-841 Disconnects from Asterisk
Thanks for the reply, however, I am already running the
latest 3.14a
It seems it may have something to do with the "Registration
expires" setting on these phones. This value is set at the default
3600. After this interval, the phone
Title: Linksys SPA-841 Disconnects from Asterisk
Thanks for the reply, however, I am already running the
latest 3.14a
It seems it may have something to do with the "Registration
expires" setting on these phones. This value is set at the default
3600. After this interval, the phone
Title: Linksys SPA-841 Disconnects from Asterisk
Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones.
They work fine generally, but occasionally, incoming calls are missed. It's like the SIP registration is expiring. Does anyone
Title: Sipura SPA-841 Disconnects from Asterisk
Hi all, I am hoping to find someone who has run into this issue with the Sipura SPA-841 phone.
Although my phones appear to be working fine, occasionally, they do not ring. When this happens, if I make a call on the phone, it seems to
Thanks, that's exactly what I have done...
But I am still trying to find out what the Shared line appearance is
on these phones?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY
Title: Sipura SPA-841 Second Line Help
Hi all, I recently purchased Sipura SPA-841 phones for a group of users. While the phones are functioning great, I am having some troubles configuring one aspect. Hopefully someone will know what I am doing wrong.
On each of the phones, I have
AM, Dave Morrow wrote:
Hi all, I recently purchased Sipura SPA-841 phones for a group of
users. While the phones are functioning great, I am having some
troubles configuring one aspect. Hopefully someone will know what I
am doing wrong.
On each of the phones, I have configured Line 1
)
you must mean
exten = s,1,Dial(SIP/110SIP/112,20,tr) ? Just append all extensions
you wish to ring, separated by
ampersands
(). The first one to answer will be winner.
That's what I think you're asking, at least.
Moj
Dave Morrow wrote:
Hi all. I wonder if anyone out there has
Title: CVS HEAD - app_muxmon
I just upgraded to the latest CVS HEAD and found that the install reported app_muxmon.so as being incompatible for this version of Asterisk. Had to remove it from /var/lib/asterisk/modules in order to get asterisk started.
Just an FYI
David A. Morrow
Title: Extension Ring on Multiple Phones
Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones.
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519)
Title: Options for 3-way or Conference Calling
Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user
Title: How to do Call Forwarding
Hi all. I am attempting to setup a dial plan which will allow me to forward an extension using the handset. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding however it does not work correctly. Does
I was hoping there would be something considerably more simple.
For example, on my legacy PBX, all I need do is press the Call Fwd
button on my phone, followed by an extension. Something similar (like
*72#ext) would be nice.
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote:
Thanks Steve. It almost works, but never dials the extension. Also
Title: Call Forwarding
Hi all. I am attempting to setup a dial plan which will allow me to forward an extension. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding however it does not work correctly. Does anyone have some expertise
Title: Help with Dial Plan
Hi all. So far this list is proving it's worth, even on my first day using it!
I hope that someone might know an easy solution to this one.
I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface
:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote:
Hi all. So far this list is proving it's worth, even
Title: Newbie Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning.
Since all inbound calls come through my T1, I would like to setup a dial plan that
Thanks,
Steve Totaro
- Original Message -----
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just go
up just do it
over.
Thanks,
Steve
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:41
AM
Subject: RE: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
I do
Title: Forwarding Extensions using dialplan
Hi all. So far this list is proving it's worth, even on my first day using it!
I hope that someone might know an easy solution to this one.
I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of
Title: Soft Phone
Can anyone recommend a good soft phone that's easy to configure under Asterisk and works well on a typical Windows XP system?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519)
Title: AMP or Asterisk
Hi all. I have been using Asterisk for sometime now and have recently come across AMP for the first time. I am wondering if someone could enlighten me a little as to the advantages and disadvantages to using AMP as opposed to the do-it-yourself Asterisk? Is this
Title: Minimum Setup
Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. I am think ISDN as I
-Users] AMP Anyone?
On Tue, Jan 11, 2005 at 09:56:39PM -0500, Dave Morrow wrote:
Hi all, I have been using Asterisk for a while now, and loving it.
Just about to update to 1.0 (running like 0.93) I was wondering if
anyone has any expertise in the implementation of AMP onto an existing
Asterisk
]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis
Boylan
Sent: Wednesday, January 12, 2005 6:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP Anyone?
On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow
Title: AMP Anyone?
Hi all, I have been using Asterisk for a while now, and loving it. Just about to update to 1.0 (running like 0.93)
I was wondering if anyone has any expertise in the implementation of AMP onto an existing Asterisk install? The instructions for it all deal with a fresh
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