I have an application that creates a call file and drops it into the
/var/spool/asterisk/outgoing directory. This file places a call and
presents a menu to the called party.
This works perfectly over SIP phones but will not work via my ISDN PRI
(Digium TE410P).
If I dial IN to the menu, DTMF
Hello everybody,
Theres a problem I am facing with the queues and agents. When the Agents
login using the AgentCallbackLogin ( ) and I use the show agents command, I see
the output on the Asterisk CLI stating both the agents are logged in. When I place
a call, all the agents are supposed
Not to mention any additional latency that you could be introducing..
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, 19 April 2005 3:38 PM
To: Javier Godinez; Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
Wondering if there is a timer provided on TDM cards?
I don't have use for TE110P and it seems expensive just to get it for
timer function.
I do have ztdummy running but it is hovering on 99.975586% and I'm not
sure if this is good enough or not.
Any info is appreciated.
This message
The citrix ICA protocol is fundamentally BAD for audio/video application in
general, let alone something as time sensitive as VOIP.
- Original Message -
From: Javier Godinez [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Sent: Tuesday, April 19, 2005 12:30 PM
Subject:
Paul Hales wrote:
From our experience (with a HP 2626-PWR) it hasn't been an issue.
PaulH
Paul,
Would you like to share wich phones you're using on this switch?
Cheers,
Kristof.
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Marc Storck wrote:
does the Junghans QuadBRI Card and qozap module support Fax detection?
You could have a look at the fax detection code in AMP, maybe that helps?
But I think it should work. We're not using it, because we're using a
fixed number for fax, so if that number gets a call, the
I have compile Asterisk many times on Woody without any problems. I am
not sure about the error your are receiving but some things that you
might want to watch for.
* You need to recompile the kernel with the source for Woody I think
this is currently 2.4.18 I noticed you have 2.4.20.
* Use
You might want to confirm this but I think the basic analog cards have
the same functionality and they are very inexpensive.
Scott Henderson
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Snom 200's Polycom 600's.
Polycom 500's with the special lead.
Later,
PaulH
-Original Message-
From: Kristof Hardy [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 19 April 2005 5:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re:
Hello all * usrs,
Does anyone know how to change your *'s primary codec and still keep the g723.1
pass through capability? Or altenativly make the phones renegociate the codec
to use when a channel is about to be bridged?
Thank you,
Etienne
___
Hey
I want to implement billing in Asterisk for a calling card type application.
My scenario is like this: PSTN = Asterisk =(IAX)= Asterisk = PSTN.
I used the ${ANSWEREDTIME} and ${DIALEDTIME} variables but
${ANSWEREDTIME} always gives a value even if the call is not answered.
e.g. If I dial on
Hi all,
When I am making call with with my Cisco 7960 SIP phone, I
found only codec g711 works.
The network call is like this:
CISCO AS5300 ---sip Asterisk sip---
CISCO7960
Peer
User/ANR Call ID Seq
(Tx/Rx) Formatcisco7960
30511694 152828207b0 00102/0
ulawciscoAS5300
Guys,
I have configured Astrisk , because until now I don't have any TDM card
in my box , I configured trunk between Astrisk and Cisco 5350 . So I am
sending all calls to Cisco 5350 , that is connected to TDM.
Here is output from my extension_additional.conf
[ext-local]
exten =
My one from www.broad-tel.com works perfectly.
On 4/11/05, Sahil Gupta [EMAIL PROTECTED] wrote:
I'm having similar issues using an X100P Ambient Chipset Clone Card
any ideas?
Regards,
Sahil Gupta
VoiceValley
On Mon, 11 Apr 2005, Dave Weis wrote:
I've got a X100P in a
I do.
On Tue, 2005-04-19 at 02:05, Wang Xiangzhou wrote:
Hi,
Does anyone run asterisk on MIPS architecture successfully?
Thanks,
Fox
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U better add line to your /etc/asterisk/modules.conf
noload = chan_phone.so
before [global] section
(of course in case U don't need that module...)
and it will start
On Tue, 2005-04-19 at 02:37, Paul A Brown wrote:
I have just installed 1.0.7 from the debian source using apt-get.
Now it
On Mon, 2005-04-11 at 15:57, Doug Millsaps wrote:
I use a headset w/out any problems, except for if my cell phone is close by
and rings. Otherwise, volume is ok and no humming. Could it be your headset?
The Labtech noise canceling headsets are the ones which seem to cause
the poblem and
Reading http://www.voip-info.org/wiki-SPA-841 I have configured Asterisk
with the following dialplan to ring using 'Simple-1' but it is not
working.
exten = 6150,1,SetVar(ALERT_INFO=Simple-1)
exten = 6150,2,Dial(SIP/6152,4,t)
exten = 6150,3,Dial(SIP/6152SIP/6148,4,t)
exten =
On Sun, 2005-04-17 at 17:14, Walt Reed wrote:
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said:
Eric Wieling wrote:
I have put into each phone settings (sip.conf and zapata.conf) in my
office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from another
Hi, does anyone ever tried assigning a DID to an extension
of similar value?
Example:
Extension 6945199
DID 6945199
It doesnt seem to work in my system.
Please advice.
Thanks.
Cheers,
Angelo
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- Original Message -
From: Vladyslav [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 19, 2005 9:02 AM
Subject: Re: [Asterisk-Users] Upgraded now Asterisk won't start
U better add line to your
You can get one at around US$6 from www.broad-tel.com, including
installation instruction.
On 4/5/05, Tore Hansen [EMAIL PROTECTED] wrote:
You do need a proper FXO card to connect your POTS line
However, that need not be expensive. A suitable card is
available by mail order in the U.S. from
Hello,
I have QuadBRI and asterisk 1.0.6 bristuffed but fax reception works ugly.
My faxes are missing many
rasters and even sending does not work well. Can you tell me with version of
asterisk , spandsp, app_sndfax etc
you use to have a good result?
Thanks for you kind help
- Original
Hi,
I am new to Asterisk. Can i use Asterisk as a session target from
softswitch/Call Agent. I mean, is it possible to initiate a SIP call
to Asterisk. My PRI terminates onto Cisco 2800 and i want to send few
numbers to Asterisk to do some application related call control.
Please advice
Hi
Since Im using the mISDN drivers and no zaptel stuff, I had to install
ztdummy to get MeetMe to work. Well, that was the plan. Now, after getting
the latest zaptel version over CVS (Im using Kernel 2.6), uncommenting all
the modules except ztdummy in zaptel.sysconfig file and compiling this
Why not to use one of the existing CallingCard solutions such AstCC
AreskiCC! There are pretty mature already and perhaps it would be better
to add your efforts on one of them!
BTW you can look on the sources to see how we manage
ANSWEREDTIME DIALEDTIME!
Rgds, Areski
On Tue, 2005-04-19 at
Vladyslav wrote:
On Sun, 2005-04-17 at 17:14, Walt Reed wrote:
On Fri, Apr 15, 2005 at 10:39:44PM +0800, Ronald Wiplinger said:
Eric Wieling wrote:
I have put into each phone settings (sip.conf and zapata.conf) in my
office:
callgroup=1
pickupgroup=1
I cannot pickup any calls from
I have 2 * server, each connects to a IPtel SER server.
i have another ATA which also connects to IPtel SER server.
when * call * thru IPtel server, i have jitter and voice quality problem,
but when i use the ATA to call both * server, there is no problem, the
quality is superb. the same thing
Thanks Tzafir,
Tzafrir Cohen escribió:
What version of zaptel?
zaptel-1.0.6
st-install tor2 /sbin/ztcfg
post-install wcusb /sbin/ztcfg
post-install wcfxo /sbin/ztcfg
post-install ztdynamic /sbin/ztcfg
post-install ztd-eth /sbin/ztcfg
post-install wct1xxp /sbin/ztcfg
post-install wct4xxp
Hi yet again everyone,
I've installed 4 E1 lines into a TE405p and its almost done. 3 of the 4
E1 lines are now working perfectly (many thanks to Andres who told me
about NFAS).
But I'm having problems with one E1 line (span # 4). When I cat
/proc/zaptel/4 I get one of two messages in the
Hi!
I want to send fax to PSTN with Asterisk, but by now I
can't.
I am using the following boxs:
Internet Zap E1
Phone/Fax Gateway(H323)--- Asterisk--- PSTN
The gateway H323 has T38 and T30. Before I began with
Asterisk, I used Cisco to connect with PSTN,
Eugenio De Vena wrote:
I have QuadBRI and asterisk 1.0.6 bristuffed but fax reception works ugly.
My faxes are missing many
rasters and even sending does not work well. Can you tell me with version of
asterisk , spandsp, app_sndfax etc
you use to have a good result?
I have used bristuff-0.2.0-RC7k
http://www.asterisk-support.ru/Members/litnimax/HowTo/DTLHowto/
My howto for using asterisk with any billing.
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Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
I use DIDs for incoming faxes as well, but we have several users with
combined Telephone-Fax-Hardware. As humans make errors and are very
lazy, these users don't want to dial another prefix when they send a fax.
This is what I try to do:
exten = _XX.,1,SetVar(NUMBER=${EXTEN}) ;save the number
Rizwan Chaudhry wrote:
Hey
I want to implement billing in Asterisk for a calling card type application.
My scenario is like this: PSTN = Asterisk =(IAX)= Asterisk = PSTN.
I used the ${ANSWEREDTIME} and ${DIALEDTIME} variables but
${ANSWEREDTIME} always gives a value even if the call is not
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
On Tue, 19 Apr 2005, Derek Conniffe wrote:
But I'm having problems with one E1 line (span # 4). When I cat
/proc/zaptel/4 I get one of two messages in the first line:
Span 4: TE4/0/4 TE410P (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RECOVERING
ClockSource
Span 4: TE4/0/4 TE410P (PCI) Card 0 Span 4
Damian Funnell wrote:
When I asked them for further information
on how to improve this they replied:
** Extract begins **
SCSI RAID can cause the problem. If disabling hyper threading does not
resolve your problem my next suggest would be to revert to a PATA IDE
hard drive solution configured
Hi There
Having difficulty displaying more than 32 directory entries on my Cisco 7960
handset. I have a Php script which returns all of the directory entries (
50 for this example tested by putting the URL into a Web browser eg.
http://mywebserver/directory.php ) found in the database, however
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
Hi,
I have an Asterisk installed on a FC3 with a Digium e100p card and an
E1 (ISDN PRI).
I'm in Brazil and using Embratel as carrier.
After few troubles I get it working to make calls, from a SIP channel
to an Fone through the carrier. But when I receive a call, this one
is transfered to the
Hi,
-Original Message-
Any comments or suggestions would be greatly appreciated.
Also if anyone
knows how to insert a CiscoIPPhoneDirectory object after
every 32 entries
using Php then please let me know.
You can't. Not with php, not with anything else. The Cisco will accept 32
On Sunday 17 April 2005 15:14, Paul Hewlett wrote:
On Saturday 16 April 2005 14:00, [EMAIL PROTECTED] wrote:
On Fri, 15 Apr 2005, Paul Hewlett wrote:
I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards -
lspci reveals these as :
The problem is that under certain
Hello,
this is not automatic, you need to set up the proper dialing rules.
the fact that a DID dumps a call into the system and that there is an
extension with the same numbers do not mean they will be automatically
connected.
post your config files...
On 4/19/05, Nathaniel Angelo A. Torres
At a hotel we have two separate internal networks, one for office and one
for guests. I want to find an elegant way to have phones on both networks
and to allow some traffic between the networks but control most of it. I am
thinking of using a motherboard with a LAN interface and adding a dual LAN
What PoE Switch are you thinking of using or own? The 802.3af Spec, as I
understand it, uses a special Fast Link Pulse to detect if the Powered
Device needs to receive power.
However, there are vendors that have pre-spec (before 2003) Switches that
MAY NOT follow the spec.
There are also
Hi :)
I've just received a couple of these units and they're working nicely as a
basic unit for making/receiving calls.
Alas, neither the HOLD, FWD or FLASH buttons do anything so I'm probably going
to end up using Park and speed-dials ( #700 etc.) to implement these basic
features unless
Have asterisk installed, and working with my 2 quicknet phonejacks and 1
linejack cards.
I cant seem to get my way thru getting the linejack to answer, and give some
choices to the callers. I cant get the phonejacks to work when I change there
entry to mode=dialtone in the phone.conf, thats
Thanks Peter - I'll try a bit of cable swaping checking - I think it
sounds like a cable fault.
Derek
Peter Svensson wrote:
On Tue, 19 Apr 2005, Derek Conniffe wrote:
But I'm having problems with one E1 line (span # 4). When I cat
/proc/zaptel/4 I get one of two messages in the first line:
Chris Mason (Lists) wrote:
Has anyone done this, can you recommend software that will give me traffic
shaping and a robust firewall with VPN capabilities?
Yes - Debian Linux (a cleaned up knoppix will give you easy install).
Use n' customize wondershaper script for traffic shaping.
OpenVPN will
[ext-did]
exten = 6945194,1,Goto(ext-local,6945194,1)
[ext-local]
exten = 6945194,1,Macro(exten-vm,novm,6945194)
DID is 6945194
Extension is 6945194
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak
Sent: Tuesday, April 19, 2005 7:30
I use monowall in an executive building with 18 different LANS all ran
through the same firewall for Internet and IP phones with dual asterisk
servers.
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hi,
Ive got a linux firewall with Private/Public ip address on two nics.I need clients to connect via SIP from the private networkto a public Asterisk PBX.Ive tried the the configs from the wiki and have not come right.
Has anyone managed to get this working and if so please share the configs
On Mon, Apr 18, 2005 at 12:09:35PM -0400, Mailing List wrote:
Does an underscore work?
Yes, and underscore seems to work fine. Thanks for the suggestion.
On Mon, Apr 18, 2005 at 11:34:03AM -0400, Michael George wrote:
I have a Snom-190 that I've successfully used on a * box with the LED's
Hi!!
i need to know the exact time when a particular user picks up a
hardphone connected to Asterisk via the FXS module. 'AnsweredTime' is
giving weird integer results.Is there any way of finding out when a
called party has picked up the phone?
thanx
sara
Hi Flavio,
I asked for help to start with asterisk some weeks ago.
Thanks for your help and thanks to other people who reply my mail.
At this moment I have configured asterisk and I have two clients ( I'm
using Kphone software like SIP client), the asterisk regist correctly
the clients, it's
How do you get 18 interfaces on one machine?
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Henry Devito
Sent: Tuesday, April 19, 2005 8:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Yes you can,
We do exactly this to provide voicemail services through the softswitch.
Your specific configuration will depend upon the softswitch setup and how
they intend to deliver calls or provide trunk signalling. You will need to
work with the softswitch vendor (or installer) to determine
I'm not fond of debian, can you use it with redhat (Centos)
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Tuesday, April 19, 2005 8:10 AM
To: Asterisk Users Mailing List -
On Tue, 2005-04-19 at 13:22 +0200, Florian Overkamp wrote:
Hi,
-Original Message-
Any comments or suggestions would be greatly appreciated.
Also if anyone
knows how to insert a CiscoIPPhoneDirectory object after
every 32 entries
using Php then please let me know.
You
I'm setting up the LED keys on a Snom 190 and it is working fine for my other
SIP clients. However, one of the extensions is an IAX2 channel to another *
server.
It can be dialed like any other extension (x200) and it can dial into the
system. However, * will not send a NOTIFY to my Snom when
On Tue, 2005-04-19 at 01:04 -0400, steve szmidt wrote:
On Monday 18 April 2005 22:30, Pedro wrote:
Every so often I get a report from a customer that DTMF stops working
while checking voicemail. The customer has to hang up and check for
messages again. I have actually had this happen to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Mason (Lists)
Sent: 19 April 2005 13:57
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] VPN/Asterisk combo
How do you get 18 interfaces on one
Is there any way to modify this text coming from
comedian?
Just wanted to let you
know you were just left a 0:06 long message (number 6)
in mailbox 14878 from Cell Phone TX
214xxx, on Tuesday, April 19, 2005 at 01:51:25 AM so you might
want to check it when you
get a chance.
I am looking into using Asterisk for an application where the upsteam switch
will provide MF digits instead of DTMF after establishing a call. This is
not during the call set up but after the call is established additional MF
digits will be passed to indicated features to provide to the caller.
Citrix is bad for anything for that matter. First and foremost it is not
secure as you have to open your PC ports for Citrix to access the remote
client. And same is the case for remote client as it Needs ActiveX
installation.
I have been flooded with Spyware once I did this for a client of
Title: Vici Dialer
It's
quite easy to do multiple campaigns on VICIDIAL(just create a new campaign, load
leads, assign the leads to that campaign and have the agent log into the new
campaign), you might want to post you question on the astGUIclient-users mailing
list though:
Hello everyone, I have some doubt on the QoS matter and I hope that someone
could
bring me some light.
I see my 3Com 3300 switch supports 802.1p priorization ,
I see through Microsoft network monitor that the packet coming from my SIP
phones
have in IP header a field which is marked Precedence
Chris Mason (Lists) wrote:
How do you get 18 interfaces on one machine?
Chris
A commodity computer would not be up to the task. For that many you
would need a security switch.
For example:-
3Com® Security Switch 7280
http://www.3com.com/prod/cz_CZ_EMEA/detail.jsp?tab=featuressku=3C13512
Tim Connolly wrote:
Is there any way to modify this text coming from comedian?
Just wanted to let you know you were just left a 0:06 long
message (number 6)
in mailbox 14878 from Cell Phone TX 214xxx, on Tuesday,
April 19, 2005 at 01:51:25 AM so you might
want to
u can get ser to register you also, at sip providers, I think its in the
uac module
Iqbal
Elton Machado wrote:
Hi guys,
I'm working of configurations examples, how to get Ser and Asterisk
working together in peace ;)
What I wan to do is to have ser doing all the hard work, registering,
Chris Mason (Lists) wrote:
How do you get 18 interfaces on one machine?
Another way, if you're not worried about performance, is to use bonded
nics and VLANs. Bonding is not required, you could technically route
between as many vlans as you like through a single 10baseT card if you
really
On 4/19/05, Tim Connolly [EMAIL PROTECTED] wrote:
Is there any way to modify this text coming from comedian?
Just wanted to let you know you were just left a 0:06 long message
(number 6)
in mailbox 14878 from Cell Phone TX 214xxx, on Tuesday, April 19,
Probably VLANs and a router before the firewall.
If you use VLANs, woud all the computers be able to access all the resoures
on the network?
I want the two networks to be seperated and only share one resource, the
PBX.
Chris Mason
www.anguillaguide.com
The additional CiscoIPPhoneDirectory objects have to come on subsequent
HTTP requests. Take a look at
http://www.jaredsmith.net/misc/cisco7960/Directory-0.1.tgz which helped me a
lot with this, although I'm using LDAP information and not database records.
It gives you a pretty good base to start
I have a several friends registered on my asterisk box that are experience
problems with their ISP blocking SIP default ports 5060 and tftp port
69. Is there any way around this problem or are they forever doomed to
VOIP since their ISP is pretty much the only ISP company on that
island. So
Well, I bought two of these when they were first released.. They seemed like
VERY nice phones for the money except for the fact that the headset jacks
did not work at all on either device. Tried multiple headsets none of them
worked. I had to return the phones..
I also remember the buttons
Is ther eany way to increase the levels or am I the only one that finds them
too low to be useful?
Chris Mason
www.anguillaguide.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JD
Sent: Tuesday, April 19, 2005 9:36 AM
To: Asterisk Users
I have a 7 slot MB with 5 4 port Ethernet cards each port is it's own VLAN.
- Original Message -
From: Chris Mason (Lists) [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, April 19, 2005 7:57 AM
Subject: RE:
I have IPv6 (via tunnel) available.
Is there a solution for IPv6 available?
bye
Ronald
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Hi Ruben. You can make a direct IP call. If the 2 sip phones can ping
each other (that is, both are reachable in the network), then in
kphone select the option File New Call, then type
sip:[EMAIL PROTECTED] , the 'number' is the number wich is configured
in kphone, sipdeviceip will be the IP of
Is there a Soft VIDEO phone available?
Have you tried it? What is your comments about it?
bye
Ronald
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I have seen folks mention FireFly softphone on the list many times. I went
to their website but could only find a version which connects directly to
their service, it did not seem configurable to use with *.
Is FireFly in fact usable with *?
Thanks!
Can't you use IAX or alternatively, setup a VPN connection? With a VPN you
will circumvent the port restrictions.
Cheers!
On Tue, 19 Apr 2005, Joel Jn-Francois wrote:
I have a several friends registered on my asterisk box that are experience
problems with their ISP blocking SIP default ports
Michael Sanders wrote:
Hi,
Ive got a linux firewall with Private/Public ip address on two nics.I
need clients to connect via SIP from the private network to a public
Asterisk PBX.Ive tried the the configs from the wiki and have not come
right.
Has anyone managed to get this working and if
Wow, that's quite a setup. What do you use for routing and firewalling?
Chris Mason
www.anguillaguide.com
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Yes, there is a solution.
Use IAX2 both on Server and Clients and bypass all that Port mapping
crap.
There are a few brands of VOIP Phones available in the market that can
do IAX2.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel
Maybe they could start by finding the info on the lawsuit that was brought
against the last ISP that tried this. They could then forward it to their
ISP and see if that gets them anywhere.
I guess this could also get them disconnected from the only ISP available
so... Don't listen to me... :)
im not expert at all, but i really think that is faster to use the
RealTime. And talking about agi, i use AGI for more complex tasks,
as looking for authorization for calls, and user defined preferences
in the calls, so agi just identify users and set some variables so
asterisk change its
I want to equip my havey duty users with headset to plug into the sipura 841
phones, can anyone recommend a suitable unit?
Chris Mason
NetConcepts
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I think we are getting off the subject, I personally don't have 18 lans. I
have two, which is good as I can hardly handle those.
I would prefer to use two different LAN adapters and allow some ports to
route between the lans, I have a music server that both lans will share and
I have to share the
i have not done such a thing. but may be a couple of questions may
point you in the right direction. Does your firewall allow
communication in this ports? (source and destiny shown):
# External SIP phones Signaling (Imcoming and outgoing)
ACCEPT all all tcp -
MF works with FGD FGC signaling.
Are you taking FGD with a tandem connection?
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael B.
Murdock
Sent: Tuesday, April 19, 2005 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Gareth Blades wrote:
Reading http://www.voip-info.org/wiki-SPA-841 I have configured Asterisk
with the following dialplan to ring using 'Simple-1' but it is not
working.
exten = 6150,1,SetVar(ALERT_INFO=Simple-1)
exten = 6150,2,Dial(SIP/6152,4,t)
In the 'distinctive ring' section on that page is
Citrix is bad for anything for that matter. First and foremost it is
not
secure as you have to open your PC ports for Citrix to access the
remote
client. And same is the case for remote client as it Needs ActiveX
installation.
I have been flooded with Spyware once I did this for a client of
Yes, you need the third party version found on the download page at http://www.virbiage.com/firefly/download/...
the direct URL for that is at http://www.virbiage.com/firefly/download/firefly-thirdparty.exe.
Enjoy!
Regards,
John
- Original Message -
From:
Me
To:
The SIPURA 3000 allows some dialplan programming.
You might check them out.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anton Krall
Sent: Monday, April 18, 2005 5:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Probably VLANs and a router before the firewall.
If you use VLANs, woud all the computers be able to access
all the resoures
on the network?
Only if you route between the VLANs. You can enable dot1q tagging in
Linux and sub-interface the PBX, then configure that port as a dot1q
VLAN
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