I've used chan_ooh323 with Call Manager version 3.3, 4.0, 4.1
and now 5.0 with great success.
Which version of Asterisk-addons are you using and which
version of Asterisk?
I have a very simple config. I seem to remember an issue
if bindaddr was not set, or left to 0.0.0.0, but I might
be
Hi Dan,
I used asterisk 1.2.10 with asterisk-addons 1.2.3.
I did two successfull calls, but with dtmf=rfc2833, dtmf was not sending at all.
Then when I made some changes, I could not get any calls to go through. The call would just hangup
after first ring.
Did you get calls going in both
Do you mean to use insecure=very in general? I have set it in
[general] of sip.conf but the problem still here. As I am using ARA,
I insert a record to the user table of the UA2 with field
insecure=very. However, the result is the same.
On 9/28/06, John covici [EMAIL PROTECTED] wrote:
I had
Hopefully this didn't get posted to the list already. I think I was having some
email problems and lost most of the days postings.
Anyway, I was wondering is anyone knew if the Gnet VP320S phones are any good?
Tim
___
--Bandwidth and Colocation
On fonalities web page, i see they offer pstn failback as a feature of their asterisk package. i've also heard before of failing back to a pri line if your t1 voip line fails. my question is. in order to have pstn or pri failback, dont you basically have to have all the equipment there on standby,
inline ...
On 9/28/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 28, 2006 at 03:24:27PM +0200, Cesc wrote:
Hello people!
I have an inquiry (not a doubt ;D ). Actually, two.
I am trying to run asterisk on an embedded Power PC platform on which
we have a linux with a 2.4.2x kernel.
1. Good box, see above
We used IBM, HP/Compaq and Fujitsu Siemens. None of them came close to
supermicro opteron servers. The Serverworks-HT1000 Chipset rocks (apart
from the broadcom nic). Things just work and I tell it exactly which
IRQs to use for which slot.
And boy, do they feel fast
Colin,
for the record I think this post was exellent and deserves a
compliment. It is probably one of the best outlines of what is needed
for a professional system I have ever seen.
Marnus van Niekerk
Colin Anderson wrote:
I concur with your approach, but "Tier 1" means as little
Colin Anderson wrote:
I concur with your approach, but Tier 1 means as little here as it
does when evaluating Internet backbone carriers. could you expand on
what evaluation criteria you use? I'm going to be pre-speccing some
stuff myself this month...
Sorry I should have been more clear.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
stan ford wrote:
On fonalities web page, i see they offer pstn failback as a feature of their
asterisk package. i've also heard before of failing back to a pri line if
your t1 voip line fails. my question is. in order to have pstn or pri
Hi,
I have a site where we currently have * and a legacy PBX (Samsung DCS)
both with BRI lines coming in. The two in linked together with four
analogue channels which works most of the time.
The local Telco in South Africa will (eventually) install a PRI with
100DID's next week and we want
It seems a number of phones have separate settings for SIP Server and
Outbound proxy.
More specifically, the Linux X-Lite softphone I've been playing with has
SIP Proxy and Out Bound Proxy settings, and it seems the GrandStream
BudgeTone 100 has SIP Server and Outbound Proxy; see
hi list,
i need some help here...
ihave the following setup
1. openser running on port 5060 - succesfully registering endpoints. all good.
2 asterisk 1.2 running sip on port 5070 on the same machine.
3. asterisk 1.09 running sip on port 5070 a different machine.
i have 2 routes in my
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Crocker
Sent: Thursday, September 28, 2006 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk - Tekelec T6000
(Vocaldata, voiss)
Should work fine without problems. Sirrix has default NT wiring, thus
you can use straight CAT5 cables.
regards
klaus
Marnus van Niekerk wrote:
Hi,
I have a site where we currently have * and a legacy PBX (Samsung DCS)
both with BRI lines coming in. The two in linked together with four
Hi,Any advice on this one ?Sometimes off-hooking a Snom 320 handset doesn't answer a multi-extension call : phones keep ringing while handset remains silent.This occurs for 1 to 5% of incoming calls.
The setup is bristuffed 1.2.10 Asterisk with Junghanns QuadBRI and 6.2.x firmware Snom 320.Every
I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint.
Pretty sure I know what the problem is now. It's not limited by devices, its limited by the length. This is when it would be nice to know C. I'm assuming the
Title: VMware and Digium TDM400P card
Hello,
I have recently installed and upgraded trixbox in vmware on a win2000 server.
Everything works as expected but I can't seem to get vmware to see the card (it was installed after the vmware/trixbox was set up)
Should vmware be able to see
Lacy Moore - Aspendora wrote:
I have one extension that rings in many places. It has just come to
my attention that I can only monitor 4 devices within a hint.
Pretty sure I know what the problem is now. It's not limited by
devices, its limited by the length. This is when it
I'm running Asterisk 1.2.12.1 on a Solaris 10 box. I've built mpg123
but it doesn't want to play well under Solaris so I want to replace it
madplay.
I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls
for mpg123 to madplay with the appropriate options.
The madplay
Hi,
the scheme is this
:
xlite ---
Asterisk --- SIP gateway --- PSTN
When i make a call
with xlite (sip) to asterisk on the display of xlite i see that the call is
connected but the phone is still ringing ..
What is the problem
??
Thanks
___
I just implemented a Dell SC430, Sangoma A101, Adtran TA750 with 8FXO and 16FXS, and about 8 Poly IP430's.Everything is working pretty well, however users are complaining about the speakerphone quality and a loud HUM on the line. I see in the
sip.cfg that gains are set to all kinds of wacky
Hi Frank,
Frank Tarczynski schrieb:
The madplay executable works find on this box from the command line but
is giving a segmentation fault when called from Asterisk.
Has anyone already done this switch? Can they share some pointers?
I run here Asterisk on FreeBSD with the buildin MOH from
Hi,running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:sometimes, call transfer works as expectet, and sometimes not. So far, I couldn't figure out any pattern in this behaviour,features.conf
:featuredigittimeout = 1500atxfer = *3-works:# user enters
Andy Green wrote:
I have recently installed and upgraded trixbox in vmware on a win2000
server.
So, you have VMWare running on Windows 2000 Server (host) and are trying
to run trixbox within a VMWare session (guest), do I follow you correctly?
Everything works as expected but I can't seem
Olivier wrote:
Hi,
Any advice on this one ?
Sometimes off-hooking a Snom 320 handset doesn't answer a
multi-extension call : phones keep ringing while handset remains silent.
This occurs for 1 to 5% of incoming calls.
The setup is bristuffed 1.2.10 Asterisk with Junghanns QuadBRI and 6.2.x
Not having a [globals] section (even if it is empty) has caused Asterisk
to screw things up in the past. I think it causes contexts to not be found.
Brian Candler wrote:
On Thu, Sep 28, 2006 at 09:44:07AM -0500, Eric ManxPower Wieling wrote:
You need the [general] and [global] sections
Try:
progressinband=no
in your sip.conf.
-Brodie
On Friday 29 September 2006 08:07 am, antonio wrote:
Hi,
the scheme is this :
xlite --- Asterisk --- SIP gateway --- PSTN
When i make a call with xlite (sip) to asterisk on the display of xlite i
see that the call is connected but the
stan ford wrote:
if you have to setup an office of 100 users now. would you rather setup a sip
trunk,a t1-pri, or even a t1? and why?
Always a PRI. PRIs have fast call setup, are reliable and work well.
___
--Bandwidth and Colocation provided by
Hi again,
as I wrote before, I'm new to Asterisk. And so, many many new questions
pop up .
For example:
I have here a very small telephony system. We have only 5 (or so)
extensions. (4 phones, 1 fax).
So I wonder if there is disadvantage if we use only 1 digit extensions
(1 for boss, 2 for
On Thu, 28 Sep 2006 21:33:13 -0400, Jay R. Ashworth wrote
On Thu, Sep 28, 2006 at 11:31:29AM -0500, Eric ManxPower Wieling wrote:
Naija Man wrote:
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten =
I tried by adding answer() to the dial plan but the problem still exists.
I am not sure if I am doing this right.
Attached is the log file from asterisk while making the call to the conf
bridge after adding answer()
Could you please let me know if you find anything out of this log file?
thanks
I got a bad batch of 360's where the hookswitch was damaged in shipping.
Snom fixed this by sticking a piece of packing foam between the switch and
the hook socket, wedging it into place. While this worked fine, I found I
had to be careful unpacking the phone - if you just yanked on the foam, a
On Tuesday 26 September 2006 12:12, C F wrote:
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
http://bugs.digium.com/view.php?id=5779
Yes, but unless there is a way of setting a local channel's state, there's no
way to achieve what
I have just loaded Fedora Core 4 (minimum installation / updated). Does
anyone have an updated list of required packages or dependencies that need
to be installed prior to basic (no real-time DB) Asterisk / Zaptel / libpri
1.4 Beta install? This is what I have so far and have used in the past
It's the
same
Thanks
Try:
progressinband=no
in your sip.conf.
-Brodie
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Also with PRI:
-Fax works
-No 911 issues
-SIP provider may or may not honor your arbitrarily set caller ID - PRI
always will if your telco isn't a dick
-Easier to break out an analog channel if needed (give me a channel bank
over an ata any day)
-Faster to troubleshoot - if you get red alarm
I'm running Asterisk 1.2.12.1 on a Solaris 10 box. I've built mpg123
but it doesn't want to play well under Solaris so I want to replace it
madplay.
I've edited app/app_mp3.c and res/res_musiconhold.c to change the calls
for mpg123 to madplay with the appropriate options.
I'm using a
Stan,
I agree with the comment below, we switched from analog lines to a PRI and
it's not always as reliable as some people think. We are in a somewhat rural
location and we have outages regularly. 1-4 hour outages every few months
are not uncommon for us. Outages of 60 seconds or so are even more
Can I find out whether it is possible to achieve this.
I would also like them to work with a dialler, cutting out Voicemails, no
answers and out of service calls giving them only calls that are live
With the dialer Ideally I would input a group of numbers into some type of
system (i.e
Hi, sorry for the question, i've been searching for a real time billing
system for asterisk with zap/sip support, for use in post paid systems
like locutorios, do you know of or use any ?
thanks
--
patoVala
Linux User#280504
Hablando en http://www.elprimoalcahuete.com
El tiempo cura los
The O'reilly Book Asterisk, the Future of Telephony is a good book.
I don't operate Asterisk; I just mess with the internal code and stuff
like that. This book helps me configure and understand how asterisk
works from the user side.
Race Vanderdecken
Code Tyrant
[EMAIL PROTECTED]
828 221 2636
I have no clue that was just a refference, you tell me.
On 9/29/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 26 September 2006 12:12, C F wrote:
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que me
parece mas fácil es que te hagas
uno propio, echale una ojeada a lo que hay en
http://www.voip-info.org/wiki-Asterisk+manager+API, lo
único que haría falta sería un poco de conocimientos deVB 6 y de como
trabajar con
Looking to set up an outbound only Asterisk installation for 5 to 10
attendants that will cold calling phone numbers in a database. The
customer would like the server to call the numbers as needed and
transfer the call to an open attendant if a voice response is detected.
The customer called
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Have you tried this:
http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications
Pato Valarezo wrote:
Hi, sorry for the question, i've been searching for a real time billing
system for asterisk with zap/sip support, for use in post paid systems
On Tuesday 26 September 2006 05:26, Geoff Karl wrote:
I am running today's SVN of the 1.4 branch, on Ubuntu dapper.
I compiled a custom kernel (2.6.15.7). Created modules of the rct and
the rtc modlue loads fine.
As soon as I load ztdummy the syslog fills up with:
rtc: lost some
Gerald Drouillard wrote:
Looking to set up an outbound only Asterisk installation for 5 to 10
attendants that will cold calling phone numbers in a database. The
customer would like the server to call the numbers as needed and
transfer the call to an open attendant if a voice response is
I have like clients several spa 3000, problem is that spa3000 is not
registered or something by the east style problem must to be by
bandwidth? spa3000 verifies bandwidth qeu can use and that is
registered or no?very I am intrigued with this problemilla. Thanks your help.
I have installed
asteriskand TrixBox (newbie) and have configured it fine.
However, when I try to upload a .wav file (8 khz mono)and use it for
the primary IVR, nothing is played.Also, when I try to locate
the file on the box, I cannot find anything with the nameI gave it
(either .wav or
Hi,I'm having some issues with the manager api when it tries to redirect a call. If a call gets transferred to a person and the person doesn't answer, after the voicemail greeting the call gets dropped. As well when I try to redirect a call to a queue, there is only one way audio. If you need any
VMWare specifically states that it doesn't provide guest access to any
non-standard devices. The guest operating systems only have access to
the disks, network, sound card, and USB ports (if they are not already
being used by another guest or host OS). As such, VMWare is not a good
solution
Hi Gerald,
What about initiating the calls inbound via the web? Check out
www.cognation.net/Mexuar
We have also implemented a proof of concept using email to deliver the
same campaign mechanics as well.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
Several companies/organizations are doing this now with Asterisk and VICIDIAL.
http://astguiclient.sourceforge.net/vicidial.html
MATT---
On 9/29/06, Gerald Drouillard [EMAIL PROTECTED] wrote:
Looking to set up an outbound only Asterisk installation for 5 to 10
attendants that will cold calling
I guess the goal is to be able to give any (not just device or channel
related) to a SIP subscription.
I can see many concepts that could use this:
Day/night mode.
Flash a light when nagios gets a red alarm.
Non phone related presence status. (IM, etc.)
I do not think that anyone would want to
While no voicemail/answering machine-detection method is perfect, the
app_amd Asterisk module works pretty well if you tune it to your
setup.
The two Asterisk-based GPL Dialers both have this capability:
GnuDialer - http://www.gnudialer.org/
VICIDIAL-
On Thu, 2006-09-28 at 15:05 +0100, Brian Candler wrote:
I have an anomoly that I am unable to explain.
...
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)
[test]
exten = 611,1,Answer()
exten = 611,2,Playback(hello-world)
exten = 611,3,Hangup()
[internal]
thanks for the responsesa couple things, if you guys could clear up for me.A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for
Yusuf wrote:
Hi Dan,
I used asterisk 1.2.10 with asterisk-addons 1.2.3.
I did two successfull calls, but with dtmf=rfc2833, dtmf was not
sending at all. Then when I made some changes, I could not get any
calls to go through. The call would just hangup after first ring.
Call Manager's
Oops, I did find them, but they
are (correctly?) set to asterisk owner in custom...Any ideas?
David MoringMC 2188http://www.fleet4.com
-Original
Message-From: "David Moring" [EMAIL PROTECTED]To:
asterisk-users@lists.digium.comDate: Fri, 29 Sep 2006 13:21:01 -0400
Subject:
Hi,
Is it possible to either pick up calls to phones that go
through a queue (using Pickup/DPickup), or not notify other users when phones
are ringing because of a queue?
When the call goes through a queue, there is no extensions
that is being dialed, and therefore just using
Hi, I am a salesman currently using asterisk to contact my customers.
So far, I have asterisk connected to two PSTN analog lines where I
only receive phones calls.
Then, I have asterisk connected to a VoIP service company for
terminating my phone calls.
I also kept one PTSN phone line to place
a couple things, if you guys could clear up for me.
A) If i have a bunch of sip ip phones, and if i were to fail to my PRI. I should still be able to use my ip phones right? i assume the signal will be changed by my T1 card? and the reverse i would assume is true for incoming calls.
On Fri, Sep 29, 2006 at 10:39:44AM -0500, Shawn Kelley wrote:
You also have to be careful like mentioned below, if you get 2 PRI's, even
from different CLECS, the will normally still come out of the same Central
Office and travel side by side on the cable. So it's likely if 1 goes down
then
On Fri, Sep 29, 2006 at 09:37:06AM +0100, Conrad Wood wrote:
7. Relationship with provider. What is their SLA? Is it the
incumbent or the clec? An incumbent will be more expensive
and more difficult to deal with but they will tend to be more
reliable. A clec will be cheaper and they
Message: 9
Date: Fri, 29 Sep 2006 08:23:29 -0700
From: Ken Godee [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Replacing mpg123 with madplay under
Solaris?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL
Thanks -
This worked. I swear I was getting a 503 or something weird before
when I did this but it seems to be working now.
On 9/28/06, Andres [EMAIL PROTECTED] wrote:
Mr. Jones wrote:
Hi Folks,
I'm curious if there's anyway to force Asterisk to transcode for
certain handsets.
All you
For some reason I'm having problems with DISA. This is what I have:
exten = s,1,Answer()
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,DISA(no-password|from-internal)
I can generate tones with no problems on this system
But I cant hear a dialtone using DISA, anyone
All,
The queue function AgentCallBackLogin() would take the name of an agent as an
argument, and would log that agent into the queue they where associated with.
We programmed our appearances on our Polycom phones to have a different
appearance for each queue, and we'd send the caller id of the
On Fri, Sep 29, 2006 at 08:47:27AM -0400, Frank Tarczynski wrote:
I'm running Asterisk 1.2.12.1 on a Solaris 10 box. I've built mpg123
but it doesn't want to play well under Solaris so I want to replace it
madplay.
Just stating the obvious: the only please where you'd need mpg123 is for
Not that server model, right now we have a dell server for testing
(which puts the server cost ata round $2000). I am hoping to get one
in to see if it will play nice with the TDM cards. This appealed to
me because of it's small rackable form factor and cheap price. For
that price I can have a
Make sure you have a /etc/asterisk/indications.conf
Shidan wrote:
For some reason I'm having problems with DISA. This is what I have:
exten = s,1,Answer()
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(10)
exten = s,4,DISA(no-password|from-internal)
I can generate tones with no
Thanks Dan,
that was awesome, and really made sense about what was really happening. :)
Will try a newer.
BTW: I did get it to successfully route inbound calls to asterisk with
oh323, and DTMF and transfers worked fine.
Yusuf wrote:
Hi Dan,
I used asterisk 1.2.10 with asterisk-addons
From where can I download the collection of Asterisk Native Sounds?
I tried the www.astlinux.com link, but I was not able to uncompress them
because they seem to be corrupted.
___
--Bandwidth and Colocation provided by Easynews.com --
Well if you currently have a T1/E1 just get a 2 port card and place it
inline to the PBX. All calls will go to asterisk so they can all be
billed and you can handle voicemail.
However the most complex issue would be voicemail notifications
On 9/26/06, Patricio A. Bruna [EMAIL PROTECTED]
B) How about DID's, how would that be handled. is there a DID failover
as well? I have my VOIP service with one company, if i had my PRI
service with another. how would those DID's get failed to the other
provider, if thats even possible at all in a timely manner.
I have yet to come up with
The cheapest suggestion is to buy a Buffalo WHR-G54S, it is the same
as the GPL linksys routers, load the DD-WRT firmware and then you can
use it as a 5 port wireless Ethernet bridge, they cost less than USD
50.
On 9/22/06, Brian Candler [EMAIL PROTECTED] wrote:
Sorry, one other equipment
The VoIP version of DD_WRT runs Ser by default
On 9/24/06, David Gagnon [EMAIL PROTECTED] wrote:
You could take a WRTSL54gs, install openwrt then openser
David
-Message d'origine-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Kennedy
Envoyé: 24 septembre 2006
To rich for my blood. Googled it. Looks like it is about $12000, I
hope to stay in the $1500 range. We are but mearly a private school.
On 9/29/06, James [EMAIL PROTECTED] wrote:
I use the Lucent MAX TNT.
They are cheap, will do up to 24 T1's, have 12 fans and I've never had one
fail.
I also
On 12-Sep-06, at 3:14 PM, Richard Klingler wrote:
Hi Jason
Hi! Sorry for the delay! :/
loadInformation6 model=IP Phone 7970SIP70.8-0-4SR1S/
loadInformation6
1. Stick with the 8.0.2 SIP image as it works best with asterisk...
at least for me (o;
- Here are TFTP server logs to
Backbone folks are noting that a fiber cut went down on the west coast,
sometime after 0945PDT today, between Washington (I think that's
State) and LA, on the order of 5 OC-48's.
So if you're having troubles with VoN today, that's probably why.
Cheers,
-- jra
--
Jay R. Ashworth
Try eBay
Forrest Beck wrote:
To rich for my blood. Googled it. Looks like it is about $12000, I
hope to stay in the $1500 range. We are but mearly a private school.
On 9/29/06, James [EMAIL PROTECTED] wrote:
I use the Lucent MAX TNT.
They are cheap, will do up to 24 T1's, have 12 fans and
On 9/28/06, Douglas Garstang [EMAIL PROTECTED] wrote:
All,
I've recently been told that the AgentCallBacklogin() application is buggy, and
I should not use it. Apparently I should use AddQueueMember() instead. I see
though that AddQueueMember() does not take the location to call back as an
On Fri, Sep 29, 2006 at 07:49:00PM +0200, Michael Neuhauser wrote:
The order of include statements is important in 1.2, I don't know if
this still holds for trunk/1.4. Could you please try to include the
'invalid' context as the last one (i.e., AFTER include = test, not
before) in both
On Thu, Sep 28, 2006 at 03:17:15PM -0400, Jeronimo Romero wrote:
Has anyone tried RedFone?? It is supposed to offload a lot of that bus
overhead to the external unit doing TDMoE.
Offloading? What exactly?
A quad E1 (4 E1 cards, much more than is needed for the 64 lines
mentioned in the topic)
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
Chapeti wrote:
Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que
me parece mas fácil es que te hagas
uno propio, echale una ojeada a lo que hay en
http://www.voip-info.org/wiki-Asterisk+manager+API, lo
único que haría falta sería un poco de conocimientos deVB 6 y de como
Well this is how I got around it, and since it's just a test box it'll
do. I commented out most of the function that's responsible for
playing the tones so that now it is just a wrapper around this:
ast_tonepair_start(chan, 350, 440, 0, 0); So looks like somethings
off with the data coming back
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 28, 2006 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Queue AddQueueMember()
On 9/28/06, Douglas Garstang [EMAIL PROTECTED] wrote:
All,
Running * 1.0.9, sip.conf allow=all is set.
Based on the advice of -users earlier this week I've ordered an Asstra 480i
CT for evaluation. Phone is up, sees Asterisk, tries to register, Asterisk
refuses. I though it might be codec mismatch so I specified allow=all. Valid
account, password OK,
[8247]
username=8247
type=peer
secret=
quaify=no
port=5060
pickupgroup=
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
allow=ulaw
allow=alaw
context=from-internal
callerid=Colin Anderson 702-8247
canreinvite=no
Tried peer , friend allow=all etc no go
We are trying a couple of the Intertex - seems to work so far
On Sep 29, 2006, at 2:59 PM, Andrew Joakimsen wrote:
The VoIP version of DD_WRT runs Ser by default
On 9/24/06, David Gagnon [EMAIL PROTECTED] wrote:
You could take a WRTSL54gs, install openwrt then openser
David
-Message
On Fri, 2006-09-29 at 16:48 -0500, Pato Valarezo wrote:
Chapeti wrote:
Hola pato, hasta donde yo sé no hay nada que sea código abierto, lo que
me parece mas fácil es que te hagas
uno propio, echale una ojeada a lo que hay en
http://www.voip-info.org/wiki-Asterisk+manager+API, lo
único
Lacy, can you confirm what i was saying about SIP Phones. if i fail from my voip connection to my pri, would i need to swap out my SIP phones with another type of digital phone?Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: a couple things, if you guys could clear up for me.A) If i
And Hi again,
I wrote:
I think I can remember something from the Asterisk-TFOT book saying
that one must not use 1-digit extension numbers. But I can't remember
that very well and can't find it in the book any more
I found it! On page 90 the book says:
quote
... (Well, almost.
If you are using Asterisk, I wouldn't think so. Asterisk will choose what media the outbound call is on. If no Voip connections are available, it can continue (by what you have defined in the dial plan) on the a pri.
On 9/29/06, stan ford [EMAIL PROTECTED] wrote:
Lacy,
can you confirm what i was
I'm not too sure on that. I don't really know, unless maybe for an IVR and you want a menu of options (which can only be 1 digit) and also dial direct to the extension. That would limit you in that regard.
I have a suggestion regarding dial plan. When I first started I saw no reason to have to
On 01:35, Sat 30 Sep 06, Norbert Zawodsky wrote:
And Hi again,
I wrote:
I think I can remember something from the Asterisk-TFOT book saying
that one must not use 1-digit extension numbers. But I can't remember
that very well and can't find it in the book any more
I found it! On
antonio wrote:
Hi,
the scheme is this :
xlite --- Asterisk --- SIP gateway --- PSTN
When i make a call with xlite (sip) to asterisk on the display of
xlite i see that the call is connected but the phone is still ringing ..
You must configure your gateway to NOT answer the call before
Setting up a new system, have two sip phones that give dial tone and appear to
dial, but do not complete,
giving a busy.
Watching the CLI thing, get this message,
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/200-0825b648,
recordingcheck|20060929-195420|1159574059.5) in new
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