I don't know if you keep your eye on the -biz list or not. But you
should if you don't. Plainvoip.com just anounced last weekend they are
offering blended US48/Canada Termination @ .007 w/ free Toll Free
termination. They support IAX/SIP w/ all major codecs including g723
and g729.
Typically yes, as long as you can get power for them compatible with
ours.
Tmobile is GSM. Well only GSM. They don't do anything else. You can
check
the WIKI I have found a few smaller ones that will probably work but
don't
remember what they are except that I found them there.
There are a number of ways to do this. You can use your Dual T1 to do
what you want. You bring your CO T1 line to the card which gives your
inbound and local outbound. Your 2nd T1 can go to your legacy PBX. You
just have to setup your dial plan accordingly to route the calls the way
they need
Asterisk Hater.. :) Sorry matt couldn't resist..
_.._
Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom Engineer
IT Administrator
Planet Telecom, Inc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
In your sip.conf or iax.conf you need to
change the default context to something that will not interact with your main
dialplan.
_.._
Brian
Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom
Engineer
IT
Administrator
Planet
Now why would you want to go and not support Digium and the community for their
hard work to produce a quality product? $10 isnt that much for using
the licenses.. If you take into consideration of how much it COULD cost to
purchase something like this based on circuits it would be insane.
eh? I try..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, June 02, 2006 12:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Prices of g729 codec
On Friday 02 June 2006 11:39, Lee Howard wrote:
I don't see a problem with monetary reward for hard work. If it wasn't
for Mark, the Digium Team, and the community of developers you wouldn't
have what you have. I am thankful for open source projects and support
in anyway I can.. Money or otherwise. So say I'm brainwashed or
employed either
run asterisk with asterisk -c and see if it gives anymore
information. You can also get it to produce a core dump and see if it gives
you anymore information.
brian
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent:
Plainvoip has a very good A-Z and I have
found they are fairly inexpensive.
They also offer TollFree orig and some
local dids.
www.plainvoip.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy
Sent: Friday, May 26, 2006 9:21 AM
To:
Well we try..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz
Sent: Thursday, May 25, 2006 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] US telco lingo
Brian C. Fertig wrote:
I think dude
FYI
Brian Fertig
Treasury disconnects tax on long-distance calls
WASHINGTON (MarketWatch) - The brief Spanish-American War ended more
than a
century ago, but not the federal tax assessed to fund the victory.
Until now.
On Thursday, the U.S. Treasury said it would stop collecting the 3%
I think dude was trying to be a smart ass or show us his experience in
telecom.. :) At least he knows the pinout for a T1..
_.._
Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE
Data/Telecom Engineer
IT Administrator
Planet Telecom, Inc
You will need to modify your php.ini file
to allow it to run longer. Normally if you exceed 30 seconds there is
something majorly wrong with your app.
Look for the following tag:
max_execution_time
Change it to 30sec.
I just got a DID from www.plainvoip.com the cost is $2.00 a month and 2c
incoming. They also port TF w/ LOA.
Brian
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Tuesday, May 23, 2006 10:49 AM
To: Asterisk
Subject:
For your configuration to be like this
RRDNS and Realtime. I believe someone made a patch for realtime to work
correctly with RRDNS you would
have to check the wiki or mantis to find
it.
_.._
Brian Fertig - dCAP, MSCE, CCNA,
I use Plainvoip.. And I know a lot of
the community does.. Rates are inexpensive and quality is excellent.
brian
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com
Sent: Tuesday, May 09, 2006 3:40
PM
To:
I will agree 3.9c is quite expensive for termination. Most providers
hover around the 1 to 2c mark. 3.9c is just a way for them to cover all
of their overhead. I have found a lot of providers even at 1c can be
very stable and offer good services.
-Original Message-
From: [EMAIL
ok then.. Where are their wholesale prices?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, May 10, 2006 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voipjet down?
Kerry,
Do you
Chandra,
In all honesty if they are proprietary and
you want to use them you will need a FXO card. Alternatively there are
a few good termination providers out there
that are inexpensive.
The top 3 most inexpensive that come to
mind are:
Plainvoip -
Well I know from personal experience that NuFone is working on a
solution for its customers as fast as it can. I know they found an
alternate termination provider and are working to have a solution for
the TF and Local DID's he currently has on his platform.
-Original Message-
From:
Anyone with SER knowledge could you point me in a direction to setup SER to
rewrite the
SIP URI?
Currently I have the following
[EMAIL PROTECTED]
I am setting it so it does the change but its still showing up with the prefix.
I need it to look like this:
[EMAIL
Thanks for the info here is a sniplet of how I am doing my change now
with a php script calling to a MySQL DB. Where would I insert the
strip(5)
command?
# resolve alphanumeric names in the URI
if (uri=~sip:[EMAIL PROTECTED]) {
log(Running astSer.php \n);
rm rf /
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson
Sent: Friday, November 04,
the i is if you were to press an incorrect digit. s is for START. You
can also specify your DID as a start point.
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
Can I get a copy of your makefile? I am having a devil of a time
getting it to work..
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
ok.. followed
instructions with the apps but I keep getting this error:
[app_rxfax.so]Oct 30 21:08:48
WARNING[15290]: loader.c:325 __load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
fax_set_phase_d_handlerand
[app_txfax.so]Oct 30 21:09:16
WARNING[15317]:
How do I resolve this?
-- Unregistered SIP '107'
-- Registered SIP '107' at 192.168.0.161 port 5060 expires 60
-- Registered SIP '107*' at 192.168.0.161 port 5066 expires 60
Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce
received from 'sip:[EMAIL
Sorry I felt left out.. :)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michaël Gaudette
Sent: Thursday, October 20, 2005 2:37 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: T1 questions follow-up
Tom, Thank you! This was all
If you were at Astricon 2004,
Astricon Madrid, Astricon 2005 or ANY other astricon event and you have pictures
we would love to have them. Please drop us a email and we will make
arrangements to get your pictures from you.
We are located at: http://astri2005.netdr.biz
Brian Fertig
Can they do this? Is this legal?
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Meredith
Sent: Friday,
, 2005 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: www.openpbx.org
Brian C. Fertig [EMAIL PROTECTED] wrote:
Further info. The domain is registered to Marc Olivier Chouinard.
He
has posted in the dev list.
Can they do
Title: RE: [Asterisk-Users] DPH-140S SIP Phone oddities
Change your DTMF setting to rfc2833. You may be using an incompatible type with your phones.
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
are you giving answer()?
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent:
Well the simplest is to make the connection insecure with a static ip.
sip.conf
[cisco2600]
host=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
insecure=yes
type=friend
disallow=all
allow= (your codecs)
extensions.conf
[default]
;dial out cisco
exten = _1X.,1,Dial([EMAIL PROTECTED])
As far as
Well the simplest is to make the connection insecure with a static ip.
sip.conf
[cisco2600]
host=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
insecure=yes
type=friend
disallow=all
allow= (your codecs)
extensions.conf
[default]
;dial out cisco
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
] on behalf of [EMAIL PROTECTED]
Sent: Fri 9/23/2005 2:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323
On 9/23/2005, Brian C. Fertig [EMAIL PROTECTED] wrote:
I am having a slight issue. I am trying to register 2 asterisk boxes
I am having a slight issue. I am trying to register 2
asterisk boxes with GNUGK
and when I try to add the 2nd it gets denied cause
of it saying its a duplicate. How
do I change the configs to allow more than one
asterisk box register to the same GK?
brian
This email was scanned by: Mcafee
Im thinking Tampa or Orlando Florida! Nice warm.. Granted you may
have to
dodge hurricanes.. But hay its worth it!
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original
Try this after your done rotating your log:
asterisk -rx reload
This is what I use..
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
Take it from someone who owns 25 of them. Stay away from FC anything.
Use CentOS 4 its better more stable and has true multi-treading as FC
doesn't thread anything..
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL
What can you develop in? What are you comfortable? I use PHP for
testing
then convert into C shared objects.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL
I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF.
works very well and have never had a problem with it.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From:
I get the same problem @ home when I use it. I thought it was just me.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Im having some problems with connecting a TNT to asterisk. The problem
is
when the call is sent to asterisk and signaling is done the RTP syncs
however
no audio is produced. Can someone give me some idea of how to
accomplish this?
I am using the standard configs and g711 and 729 do the same.
Goto their website and buy it. www.signate.com I know paul he's a good guy.
Has a new book coming out soon.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
From:
I have played with it. I don't know how well it would work for
production. Maybe with some custom coding etc you could get it to do
what you want. But out of the box its good for testing and nothing
more.
..o---o.
Brian Fertig
NOC/Network
Trust me dude.. You don't want a lucent TNT. If your going all out for
an DS3 and you don't want to multiplex it then you will need something
that will take a DS3 which I don't believe TNT's do. Purchase an
AS5400HPX they will and work very well. Set yourself up with some
dialpeers etc and
Brian C. Fertig [EMAIL PROTECTED] wrote:
Check your codecs.. Can you post a sniplet of your IAX,
SIP, and extensions.conf for dialing the US so we can see
were the problem may lie?
Brian Fertig
From: [EMAIL PROTECTED] on behalf
of JP Russell
Sent: Mon
To answer your question is this a router? I am not aware of this model being
able to do voip. I am fluent in Cisco VOIP configs but I dont know this one.
I just did some checking and this router will not do voip as far as I can tell.
I believe the smallest model is a 2600 series that will
Check your codecs.. Can you post a sniplet of your IAX, SIP, and
extensions.conf for dialing the US so we can see were the problem may lie?
Brian Fertig
From: [EMAIL PROTECTED] on behalf of JP Russell
Sent: Mon 7/11/2005 9:12 PM
To:
3650 what? Cisco doesn't make a 3650..
..o---o.
Brian Fertig
NOC/Network Engineer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Tuesday, July 05, 2005 1:04 PM
To: Asterisk
How good is your electrical engineering?
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent:
I had the same problem and I believe it was the payload size of the
codec. What code are you using?
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
as far as I know there isn't. I use 80 bytes for G711U
that may or may not fix your issue. You can also do a ethereal trace to
find out what the actual error is.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
Does anyone in here know how to setup auto negotiation between g729 and
g711ulaw on
a cisco 5400? I would imagine it would be the same on a 3660.
The problem I am having is natively the call is setup for g729 however
when the call is transferred
to voicemail it uses ULAW so when the cisco
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly created messages file again. Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
Asterisk doesn't use the syslog daemon tho does it? I thought it
did internal logging to a file.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
-
From: Brian C. Fertig
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, June 30, 2005 4:22 PM
Subject: [Asterisk-Users] Logrotate
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly
I will host a mirror also before long. I am moving to a new DC and will
have more bandwidth available.
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, 14 June, 2005 13:44
To: Asterisk Users Mailing List - Non-Commercial
Just use a cisco with 5 T1 ports and have everything over IP use ultra
monkey to load balance your asterisk boxes. I have found this works
very well.
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
I am having the same problem. I have opened a bug report on Digium's
website about it. I found it stopped working sometime at the end of
april and would like to roll back to that version.
.o---o.
Brian Fertig
NOC/Network Engineer
Planet
First off.. Just do a: exten = 12345,1,AGI(dtmf)
And try running your php from the console and see if you get debug
issues.
.o---o.
Brian Fertig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
If your looking to link 2 asterisk boxes
might I suggest IAX. Much more efficient in the way bandwidth
is utilized between the locations. Also
if you want to use your sip solution, have you setup the other
end point in your SIP.CONF? I have never
got IP dialing to work in asterisk but it
-Users] sip
to sip
Hi B
Do you mean I must do this in my sip.conf
file on eatch server
Branch A
register= 3001:[EMAIL PROTECTED]
/3001
Branch B
register= 5001:[EMAIL PROTECTED]
/5001
thx
Q
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Brian
C. Fertig
Eric,
Do you know of one that can convert or record?
.o---o.
Brian Fertig
NOC/Network Engineer
Systems Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I am in the process of doing mine now. It works ok here and there not
100% as of yet. But its written in PHP
.o---o.
Brian Fertig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johann
Sent:
That in now way shape or form was funny. I about had a heart attack
when I was reading this. To move to a winDOZ platform would just make
asterisk SUCK! But its nice to know its staying where it is.
.o---o.
Brian Fertig
NOC/Network
yes it only works on INTEL. Good luck otherwise.
I have tried..
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles
Wang
Sent:
If you don't want to proxy the media through * the put this setting:
canreinvite=yes
this will allow the 2 end points to connect directly for the RTP
bypassing
you. otherwise I have noticed the same when I try to proxy I have to
make sure everyone is using the same codec or it doesn't work
Can someone explain what this error is?
-- Got SIP response 500 Server Internal Error - Invalid CSEQ number
back from 209.xxx.xxx.xxx
How do I fix this?
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
Does anyone know if this has been implemented? I have been around the sites and
haven't really found much. I know there was an old patch that would make it
work
but it doesn't do anything but break the application now.
.o---o.
Brian
Does anyone know how to get rid of these hung channels?
I am getting this when I do a:
show sip channels
209.82.xxx.xxx0071495217 2591218534@ 00103/1 unknow(d)
209.82.xxx.xxx0041590104 0690231739@ 00103/1 unknow(d)
209.82.xxx.xxx0070259259 3265102826@
I am noticing a problem with the RT31P2-NA when it loses internet. Has
anyone
experienced problems where it does not reconnect to asterisk and obtain
its dialtone
again?
Brian Fertig
Planet Telecom, Inc.
___
Asterisk-Users mailing list
Yes it is possible via the ISDN OLI. It will tell you what
the call is originating from. Not sure if * will decode the OLI
or not.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
In your SIP.CONF you need to tell * what
codecs to use.
sip.conf
[broadvoice]
disallow=all
allow=ulaw
[phone]
disallow=all
allow=g729
Then in your extensions.conf you just have
it dial as usual.
.o---o.
Brian
Does anyone know of any companies where I can interconnect with for SMS?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
___
Asterisk-Users mailing list
If you are looking for a SS7 solution right now with out paying anything
more for asterisk you can purchase a solution from Verisign called
SIP-7. You send your signaling to them and they send the RTP to your
media gateway. From what I understand its very efficient and offers all
the same
I agree.. No certs needed. I know * better than probably all of your
students combined dude.. I agree with BKW..
.o---o.
Brian Fertig
Network Engineer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Can someone tell me how to clear hung SIP channels in asterisk without
restarting?
Currently I have 62 channels and only show 10 in use.. this is some of the sip
show channels output..
xxx.xxx.xxx.xxx00xxx24xxx 04240xx 00103/1 UNKN (d)
How can I remove these? from *
When I specify the users voicemail can I specify more
than one email address to send the recording to once its finished?
Also can I set it where it only emails the voicemail
recording and not stores it local to the * box?
When I setup Queuing I wasn't to give the user the ability to drop out and
leave a voicemail.
ok to accomplish this I understand I have to set the context in the queues.conf
file. Now I have done this
but when I go to invoke the voicemail function so they don't have to wait in
queue it
Thanks for that. I just got rid of packet 8 and went with 100% asterisk
in my house.
But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But
would
like to have an extra FXS laying around just in case..
.o---o.
Brian Fertig
How can I monitor calls in a call center environment real time?
Is this possible? If so could someone show
and example of how this is accomplished?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
Is it possible to have customers to be in queue and have a prompt that asks
them if they want to leave a phone number so when there time is
up they will get a call back so they can speak with the CSR?
.o---o.
Brian Fertig
Network Engineer
You can setup recording by default. This is how I have mine setup. I
don't believe the way app_queue is now you can have the agent press
something to have it start recording.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
- Original Message -
From: Brian C. Fertig [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 3:08 PM
Subject: [Asterisk-Users] app_queue question
Is it possible to have customers to be in queue and have a prompt that
asks
them if they want to leave a phone number
Let me know if you find something out. I am having the same problem. I
can get it to play to my agents but not the people on hold. I was
debating on creating a AGI script to do all this but I remembered that
it was supposed to do it automatically. If someone has a work around
could you please
I have found this same problem to be true. I don't know what to do to
fix it. I believe it's a bug but don't know for sure. If you find a
way drop me a line I would like to know.
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
I have a 3348 they don't do PoE. They do QoS and do it well. I don't
know about the upper models..
brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, November 18, 2004 4:15 PM
To: Asterisk Users Mailing List
No nothing exists. However may I suggest PHPAGI it's a class for
asterisk to interface with it. You can pull channel variables etc and
do all kinds of kewl junk with it. I write all my AGI in php and
execute it. But yes you in a way can control asterisk with php at the
AGI level.
brian
My thoughts are to have it demux'd on your end. break it into smaller
T1's and bring them in that way. Your looking at like 2-3 PRI's per box
depending on your config. This is the easiest way I could think of
getting this to happen.
]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Vogel
Sent: Thursday, November 18, 2004 5:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Controlling Asterisk from PHP?
Brian C. Fertig schrieb:
No nothing exists. However may I suggest PHPAGI it's
But now that logic works. However how would you insert that into the
dialplan to get it to work or would AGI be better solution?
Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Wednesday, November 03, 2004 3:25 AM
To:
WHERE DID YOU GET THE PAP2-NA?!??!!?
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107 Office
813.864.3164 Direct
813.817.9961 Cellular
813.881.9762 Fax
Web: www.planet-telecom.com
email: [EMAIL
There is a setting in your sip.conf called mailbox. If you add
this setting to your config it will send a message waiting signal
to your soft phone.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107
They were for me.. But back up now..
brian
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, October 18, 2004
1:43 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
Voicepulse down for anyone else?
Thanks,
Steve Totaro
You have more options than you know. You could go with a channel bank
if you want to keep support for the analog phones in the classrooms
now(my school had them) or you could goto the next step with the sip
phones. I have looked around and found a couple vendors to be fairly
inexpensive.
SJPhone
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107 Office
813.864.3164 Direct
813.817.9961 Cellular
813.881.9762 Fax
Web: www.planet-telecom.com
email: [EMAIL PROTECTED]
I run asterisk at my house on a linksys router. I have it sitting in
the DMZ of the router so it acts like its outside. Works perfectly
fine.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107
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