You can do this with agents, no need for a queue.
Define agents in agents.conf
In your dialplan, instead of Dial(SIP/bedroom) use
Dial(Agent/200)
Let the phones login as agent :)
OK, I know I have to Dial(Agent/200), but how will I login agents if I don't
use queue? If phone log's in as
AMP doesn't do miracles! Look at its dialplan.
I believe he doesn't, but I don't have AMP installed. Next week I think I'll
have enough free time to try it. Will [EMAIL PROTECTED] do the trick?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth
Hi friends !
I want to add stun functionality in asterisk.
can anybody give me some hint that how can i start that.
thanks in advance
Deepak Dhiman
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
On 09:02, Fri 17 Feb 06, Tomislav Par?ina wrote:
You can do this with agents, no need for a queue.
Define agents in agents.conf
In your dialplan, instead of Dial(SIP/bedroom) use
Dial(Agent/200)
Let the phones login as agent :)
OK, I know I have to Dial(Agent/200), but how will I
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
We have got some ATA for only $55 if you are interested?
Sam
Yes Sam, I'm interested. If they work with FAX I'll definitely buy one of them
for testing.
--
Tomislav
[EMAIL PROTECTED]
___
On 2/15/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
Looks like you're making some progress. When I first started using [EMAIL
PROTECTED]
I had trouble getting the outbound dialing to work. I wasn't sure where
to start, so what I did was skip the macros in the dial plan. I wanted
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier
The first Asterisk system is running 1.2 and the second is running 1.0.
When using g726 from the handset all the way thru
Hi Chuck,
my solution may be considered a bit strange but I chose it after trying
asterisk code without success, trying to use Tzafrir patch but I had to
change asterisk user umask too
The right solution could be something like a voicemail_dir_permissions
parameter in voicemail.conf so
Hi List
Is someone out there using one or more GSMgateway(s) from CyberTelecom ?
Me and some friends are interested in buying some of them, but before
we would like to ask, how the experiences are others have made.
e.g.
How easy to setup ?
How reliable ?
How's the voice quality ?
etc.
Any
Since you have no Digium hardware (and thus no connection to POTS or
PRI)... are you routing your phone calls via VoIP? If so, it is not
recommended to run FAX via VoIP. The two don't mix. FAX is not able to
handle packet loss like VoIP. Also, any codec other than uLaw will not
even come
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
AFIK, fax is supported and installed with with app_txfax app_rxfax
If this proves to be true why would you need the ATA?
I'm working on this one. I have to install app_rxfax but I have failed. Soon,
I'll try again (hopefully next week).
Hi there,
Is it possible with the new aastra firmware to have distinctive ring
support? (the wiki says: There doesn't seem to be any way to have the
server request a distinctive ring.)
The rest of the features make this sound like a good phone. (price/quality)
cheers
Why don't you simply rotate the logs with logrotate ?
How to do that?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
2006/2/8, adibar [EMAIL PROTECTED]:
Hi
That does the job (dialout only)
I'm trying with this configuration but I receive the same result.
Checking with ethereal I see the answer from sipdiscount:
asterisk-sipdiscount Request: INVITE sip:the number@sip1.sipdiscount.com
sipdiscount-asterisk
2006/2/17, Ben Dinnerville [EMAIL PROTECTED]:
Hi All,
We are experiencing a a problem when running calls over IAX with g.729.
The call flow is as follows:
Sip handset -(SIP) Asterisk1 -(IAX) Asterisk2 -(SIP) Carrier
if you are calling asterisk-to-asterisk, you should try speex
compression.
The carrier does not support speex, only g729, 723 and 711, so to
minimise codec coversions etc, and due to the fact that licensing 723
is so expensive and 711 is a bit fat on bandwidth (asterisk 1 is
connecintg over 128k ISDN) we are kind of stuck with g729 (not that it
has ever proved to be
yes, with last patch works well. thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adolfo R.
Brandes
Sent: Thursday, February 16, 2006 10:11 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: asterisk t.38 pass
turby wrote:
is
Hi
I hope you dont try to dial with the user test and
the password being also test, that would definitly
end up in an Unauthorized ;-)
Don't forget the register-stuff inside sip.conf, e.g.:
register = YOURLOGIN:[EMAIL PROTECTED]
otherwise it does not work for me either ;-)
If you do a sip show
Hello,
Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list
to ask this question?
Best regards,
Vlasis Hatzistavrou.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vlasis
Hatzistavrou
Sent: Thursday, February 16, 2006 3:43
The UPD I'm using for the * is actually an UPS I used for a much biger
Windows machne, complete with monitor etc. The coleague who used the UPS was
aut of the office when I installed the system and took hid UPS :-) I'm sure
the UPS is good.
I'm saying I'll change the PSU because I've had problems
check out the man page for logrotate
The logrotate script is usually started daily by the cron daemon ( see
/etc/cron.daily/logrotate on redhat boxes)
hth,
cristi
On 2/17/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
Why don't you simply rotate the logs with logrotate ?
How to do that?
On 08/02/06, Alejandro Vargas [EMAIL PROTECTED] wrote:
Sipdiscount has replaced their asterisk servers for another thing.
Then, no more iax. Ok, but I can't make calls using sip also... I'm
getting a forbidden error when using sip1.sipdiscount.com. Anybody
got it working?
A pretty simple
AMP doesn't do miracles! Look at its dialplan.
I believe he doesn't, but I don't have AMP installed. Next
week I think I'll have enough free time to try it. Will
[EMAIL PROTECTED] do the trick?
Yes, I was referring to AAH
Mimmus
___
On 02/16/06 04:45 Prakash Rao Kanthi said the following:
This works but the calling party hears 'prompt02' and the called party
hears 'prompt04' the two parties are NOT connected foa conversatoin -
just like the wiki describes
Does anyone know when the 'G()' flag will be fixed or any
On 02/17/06 08:51 BJ Weschke said the following:
On 2/15/06, Kevin Hanson [EMAIL PROTECTED] wrote:
I am using an Asterisk box as a mini-softswitch and have run into a
minor (hopefully) road block. The far end switch requires CIC (Carrier
Identification Code) in the SIP invite like:
INVITE
On Thu, 16 Feb 2006, Martin Joseph wrote:
I have ordered the wellgate 3701A to see if that helps me any... It's about
twice the price of the SPA3000 ($199), but I know my 2 wire loop is 15000ft+
so I figured the SPA3000 isn't going to help me.
I'll be very interested in your review of this
turby wrote:
yes, with last patch works well. thanks.
Glad to be of service!
Adolfo
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Didnt think of that. Thanks for the insight.
Dovid
--- Matt Florell [EMAIL PROTECTED] wrote:
Hello,
We had over 100 ATA adapters in production 3 years
ago. Now we have
less than 20. They use more power overall than
Channelbanks, they are
not designed to be used 12-16 hours a day every
On 12:17, Fri 17 Feb 06, Vlasis Hatzistavrou wrote:
Hello,
Does anyone know any solution to this? Or is Asterisk-Dev a more suitable
list to ask this question?
It's a -user question to begin with.
Have your agi connect to the manager interface and get the
answer info from there :)
--
Hi,
we send an SETUP message to an SIEMENS (German) Provider.
Our Equipment is pri_cpe, so we may NOT send an IE Display
to the Carrier.
Call Ref: len= 2 (reference 15072/0x3AE0) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info
2006/2/17, adibar [EMAIL PROTECTED]:
Hi
I hope you dont try to dial with the user test and
the password being also test, that would definitly
end up in an Unauthorized ;-)
But... the page of sipdiscount says you can use the user test with
password test to do free one minute calls for testing
2006/2/17, Peter Bowyer [EMAIL PROTECTED]:
A pretty simple setup works for me:
The problem may be the username/password. But the page says this:
SIP Discount offers the possibility to test our service right away,
for free! No need to sign up: just enter the account details below in
your
On 02/17/06 10:13 James Texter said the following:
static int vpmdtmfsupport = 1;
Change this to
static int vpmdtmfsupport = 0;
i'm guessing that this would only be relevant if you were using the newer
TE4XXP cards with the VPM boards attached.
--
Regards,
http://www.trxtel.com/index.php?page=Tollfree_Termination
This is a free service, I am not selling anything with this service. I
just thought that individuals that read this list may enjoy getting
tollfree access free this way (yet another way) given that it lets you
send your caller id and some
I'm having trouble with Asterisk-1.2.4 negotiating codecs with a Sipura
3000 which is running the latest v3 firmware.
The SPA-3K seems to use the preferred codec only and doesn't
negotiate? The SPA is set to no in use only preferred codec.
Does anyone know if Sipura will support gsm at some
Matt,
Whose channel banks did you end up going with?
Hello,
We had over 100 ATA adapters in production 3 years ago. Now we have
less than 20. They use more power overall than Channelbanks, they are
not designed to be used 12-16 hours a day every day. You must
We ended up going with Zhone channelbanks because they are available
very cheaply and are very small. Just make sure you get the B version,
they are much easier to program than the A version. I must note that
Zhone channelbanks are not made anymore so you must buy them on the
secondary market.
Hi
I would sugest, that you just register without balancing your
account. Than use the supplied username/password and it will
work. I doubt that the test/test works.
Greets
Adibar
On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas wrote:
2006/2/17, Peter Bowyer [EMAIL
No, distinctive ring isn't supported in 1.3.1. You only have the option
of setting the ring-tone on a per-line basis.
Gareth
stoffel wrote:
Hi there,
Is it possible with the new aastra firmware to have distinctive ring
support? (the wiki says: There doesn't seem to be any way to have
2006/2/17, adibar [EMAIL PROTECTED]:
I would sugest, that you just register without balancing your
account. Than use the supplied username/password and it will
work. I doubt that the test/test works.
Thanks. This worked. I already had a sipdiscoutn account without
credit, but It never worked
Sorry, this is off topic to asterisk itself, but is about
the list server.
I had a power failure lastnight at home, where my email
server resides, and my network was down for about 20
minutes, that was after 45 minutes of uptime on UPS. Since
power was restored, around 9:45 PM EST on 2/16,
On Fri, 2006-02-17 at 14:32 +0100, Alejandro Vargas wrote:
2006/2/17, adibar [EMAIL PROTECTED]:
I would sugest, that you just register without balancing your
account. Than use the supplied username/password and it will
work. I doubt that the test/test works.
Thanks. This worked. I
The follow should work from the configuration files
(aasta.cfg/MAC.cfg), although I haven't tried it...
audio mode: mode
Where mode is a number between 0 and 3
0 = speaker
1 = headset
2 = speaker/headset
3 = headset/speaker
Gareth
Lee Archer wrote:
In article [EMAIL PROTECTED],
Dinesh Nair [EMAIL PROTECTED] wrote:
On 02/16/06 04:45 Prakash Rao Kanthi said the following:
This works but the calling party hears 'prompt02' and the called party
hears 'prompt04' the two parties are NOT connected foa conversatoin -
just like the wiki
Been around asterisk for two-plus years, but need a little input from the
list on this topic.
Have a potential client that wants to replace their old key system with *,
but they want to integrate a commercial message service (they pay a monthly
fee to have special MOH messages generated) into
If you figure it out, please let me know. I would
actually love to _enable_ such a beep for my agents...
(If it
isn't there already...)
Bob McDowell
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
RichardsonSent: Thursday, February 16, 2006 5:23 PMTo:
Asterisk Users
I have not done this but I could probably send you in the right
direction.
* MOH uses a he standard out of an audio program (ie mpg123) you should
be able to add a custom mohtype in the musiconhold.conf file.
All you need is to 'play' the audio from the line in on your MB and put
it on STDOUT.
On 02/17/06 21:50 Tony Mountifield said the following:
I think it is more useful to transfer to the two separate priorities,
but the documentation should reflect that.
this makes sense. however the help text for 'show application dial' should
then be updated to reflect this. i know this
On 2/17/06, Gareth Owen [EMAIL PROTECTED] wrote:
No, distinctive ring isn't supported in 1.3.1. You only have the option
of setting the ring-tone on a per-line basis.
hm, okay. is it a feature that will be built-in in the future? or can
you say for sure it will not?
thanks,
cheers
Has anyone worked with one of these boxes and Asterisk?
I have the Tenor AX registering 24 extensions just fine with asterisk
but when I try to call one of the configured FXS extensions on the Tenor
AX, I get Got SIP response 415 Unsupported Media Type back from
xx.xx.xx.xx.
I have tried
This is my own GotoifTime section, which works swimmingly I
might add:
exten = s,1,Answer exten =
s,2,SetMusicOnHold(default) exten =
s,3,Set(TIMEOUT(digit)=5) exten =
s,4,Set(TIMEOUT(response)=10) exten =
s,5,Background(fedwelcome) exten =
s,6,GotoIfTime(*|*|1|jan?afterhours,s,1)
; New
Brent Torrenga [EMAIL PROTECTED] wrote:
We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This
should get rid of static on the line (at least any static generated by our
half of the circuit), right?
I am very interested in this too. My main motivation is to get the
improved
Kerry Garrison [EMAIL PROTECTED] wrote:
I dont recall the SPA-941 playing a stutter tone in the previous firmware
but it is driving me nuts, anyone know where to turn it off?
I can't help, but I do understand your pain. I tried to turn this off
with the SPA-2000 with no luck.
Doug
--
Doug
I did some config with one of these. When I got that error it was
because I had only the G729 codec selected on the quintum and did not
have the g729 license for the asterisk. I switch alaw on the quintum
and it worked.
Michael Sampson
Information Systems Manager
Customer Contact Services
My * pc has an integrated soundcard, should be ok for this type of
application, I'd get an RCA-to-line jack cable (radioshack should have
those ;) I know * can play hold music from a streaming server, and I
know some streaming servers can stream from a line in, so the
combination of the two should
Nice one it works. Is there a complete list of all the options you can
use in the config files?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gareth
Owen
Sent: 17 February 2006 13:39
To: asterisk-users@lists.digium.com
Subject:
Hi
Have someome a solution to use the hint function to have the signalling
of the status of a extension on the SPA-941 phone ?
Matteo
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
Hello List,
I work for an IP communication provider in upstate NY
as the engineer assisting our technical support team.
We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble shoot
termination problems I came up with the idea
Hi,
I'm asking to myself what's the main problem in using cheap BRI cards
(30-60Euro, as these HFC-based) vs. great active cards as Eicon DIVA.
Any help?
--
Mimmus
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
Any idea how difficult it might be to get an integrated sound card to
work properly with asterisk? (That seems to be the limiting factor or
more time consuming part of doing this. Adapting the cables and audio
levels is easy.)
My * pc has an integrated soundcard, should be ok for this type of
Rich Adamson wrote:
Been around asterisk for two-plus years, but need a little input from the
list on this topic.
Have a potential client that wants to replace their old key system with *,
but they want to integrate a commercial message service (they pay a monthly
fee to have special MOH
Try turning off iptables (firewall)
service.
MD
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abhimanyu
RapriaSent: Friday, February 17, 2006 2:19 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP Problem
Fedora Core 4 and Asterisk 1.2.4
Fedora:Linux
On 2/17/06, Aloi, Christopher [EMAIL PROTECTED] wrote:
Hello List,
I work for an IP communication provider in upstate NY as the engineer
assisting our technical support team.
We provide a number of different Telco systems to residential subscribers;
and in an effort to more effectively
The
patch you saw is not for the stable branch.
Salu2
Jsalas
Right, but try using this, i adapted
it, no guarantees, i have not made tests, just modified it to apply
properly, it would be great if some one can test it:
http://chewbacca.ivsol.net/asterisk-1.2.1-silence-suppression-4.patch
Has anyone dared to go down the visdn road. www.visdn.org
I want an alternative to zaphfc for the passive HFC-PCI card.
I managed to get the snapshot version of 17th Feb to compile against *1.2.4 et
al.
Now I am battling to get it configured.
Could anyone with a working visdn.conf for 2 HFC
Barix Instreamer takes RCA in and MP3 or ulaw stream out. Asterisk can
use either for MOH.
On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote:
Been around asterisk for two-plus years, but need a little input from the
list on this topic.
Have a potential client that wants to replace their
My Intel board's card works great with * for paging... I haven't ever
tried it the other way.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Friday, February 17, 2006 9:18 AM
To: Asterisk Users Mailing List -
Ok,
So... we've been looking at Intrado as a solution for national E911.
They claim to be able to offer FCC compliant E911 services for VoIP
companies. However, as I look into things further, they don't seem
to have links to all the PSAPs for E911. Now, I understand if the
PSAP is not capable
You
create a context in your dialplan that accepts the DID to call as a variable
using the SetVar: syntax in your .call file. You then set up the context to call
your agent, and when they pick up, the context takes the variable you set in
your .call file asthe dialstring argument for a
Actually, with my suggestion, you would be using the soundcard with
whatever streaming mp3 server you choose to use, some kinda shoutcast
server I guess, so there shouldn't be any asterisk-soundcard
interaction..
* just takes its moh from the streaming server.
On Fri, 2006-02-17 at 09:18 -0600,
Why dont you use Local and router
functionality to find a route to PSTN based agents?
W
From: Aloi,
Christopher [mailto:[EMAIL PROTECTED]
Sent: Friday, February 17, 2006
10:07 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] A unique
Been around asterisk for two-plus years, but need a little input from the
list on this topic.
Have a potential client that wants to replace their old key system with *,
but they want to integrate a commercial message service (they pay a monthly
fee to have special MOH messages
I just double checked my SPA-841. You can change the dial tone in the
Web config on the Regional page. I just copied the Dial Tone: to the
MWI Dial Tone field and it didnt stutter after that. I'm not sure if
its the same with the 941, but i've heard the phone configs are similar.
Hope this
Jesus,
If you recompile Asterisk and still have problems, take a look in
/codecs/Makefile. It'll tell you where Asterisk expects to find stuff
in order to trigger the building of the speex-related objects. If the
build goes as planned, the /codecs directory will contain three
speex-related
Hi,
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.
- Is it a good choice (budget limit of 100 Euro/phone is mandatory)?
- Can be a profitable business the direct buying of 50 phones (to save other
money) or is it a risk?
Thanks in advance
--
Mimmus
I'm using the telnet manager interface with the 'originate' command,
just a little perl script that connects and has asterisk dial the
selected number.
It rings the extension first, if they pick up, it'll dial the remote
number.
It's one of the showcase features of the new phonesystem for us :)
Hi all, I'm trying to setup a custom extension in AMP (yes i can code it
by hand but the on-site admin that does moves changes cannot).
I've tried the following
add cutom extension
600
in the dial box i have
Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED])
this doesnt work as
Hello,
I'm not sure what you mean, could you
elaborate?
Thanks,
--
Christopher T.
Aloi USA
Datanet - Technical Support Engineer 318 South Clinton
Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074
E: [EMAIL PROTECTED] -- --
--
From: Wojciech Tryc
[mailto:[EMAIL
OK I'm answering my own question but if i add a custom extension in AMP
with no dial string.
Then add a dialstring in extensions_custom.conf like
exten = 600,1,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED])
it works
Bails
---BeginMessage---
Hi all, I'm trying to setup a custom
Nik,
This definitely helps! Please check your dial command. You've got
Dial(Zap/0/mynumber) and I think you might possibly want it to be
something like this:
Dial(Zap/1/mynumber) or
Dial(Zap/g0/mynumber)
I don't recall there being a zap channel zero, but it is common to have
a group zero. I
Hi everyone,
I am trying to get iaxmodem up and running.
This is a very basic setup, which at this moment should only answer
incoming faxes.
What I did:
zapata.conf (rest of it should be fine):
faxdetect=incoming
group = 1
channel = 1-2
context=from-pstn
iax.conf:
[200]
username=200
Mark:
I did so, but that did not make asterisk to integrate speex.
Do I have to tweak something in speex after installation?
This is some of asterisk output when I try to use speex:
-- Accepting AUTHENTICATED call from 192.168.2.32:
requested format = speex,
requested prefs =
Colin,
Thanks for your assistance.
Reading over your advice I seem to still be a bit
confused.
My agents are not on the Asterisk server; it appears in
your advice that my the call will travel this path:
WWW interface -- agent enters their DID, platform
to use, and termination DID -- AST
Rich Adamson wrote:
Been around asterisk for two-plus years, but need a little input from the
list on this topic.
Have a potential client that wants to replace their old key system with *,
but they want to integrate a commercial message service (they pay a monthly
fee to have special MOH
That was it. There is way more configuration in these things than I
need and I guess I have to RTFM. VERY impressive box. I just want to
use it as an FXS Gateway. I set the codecs to ulaw and alaw. I
configured the SIP useragents and as I said, it is registering with
asterisk.
Problem now
How is your echo can the issue?
Did you disable the echo can and solve the DTMF issue?
I actually think my echo can had gotten into some odd state, a restart of the
tellabs board fixed the issue.
___
--Bandwidth and Colocation provided by Easynews.com
I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.
- Is it a good choice (budget limit of 100 Euro/phone is mandatory)?
- Can be a profitable business the direct buying of 50 phones (to save other
money) or is it a risk?
if you've never tried a phone, it's always a
You could do something like :
[router-local]
exten = _613XXX,1,Goto(trunklocal,
${EXTEN:${TRUNKMSD3}},1)
exten = _613XXX,2,Congestion
[router-ld]
exten = _1NX,1,Goto(trunkld,91${EXTEN},1)
exten = _1NX,2,Congestion
[trunklocal]
exten =
Same
as before but instead of SIP as the originationchannel you pass
ZAP/g0/XXX (the DID of the agent) to your .call file. In fact, this is
exactly how the www.landmarkhomes.ca
script works (it calls the guy who entered his phone number in the website, when
he picks up, it calls
Christian Lox wrote:
Hi everyone,
I am trying to get iaxmodem up and running.
This is a very basic setup, which at this moment should only answer
incoming faxes.
extensions.conf:
[from-pstn]
exten = fax,1,Dial(IAX2/200)
When trying fo fax, all I get is:
Extension '265399' in context
My guess would be that the mqueu was just too busy.
On 2/17/06, Robert Webb [EMAIL PROTECTED] wrote:
Sorry, this is off topic to asterisk itself, but is about
the list server.
I had a power failure lastnight at home, where my email
server resides, and my network was down for about 20
Elaborating a little more I checked for files suggested by Matthew Roth:
If the build goes as planned, the /codecs directory will contain
three
speex-related files:
- codec_speex.c
- codec_speex.o
- codec_speex.so
Then ran the show modules command and now codec_speex shows as loaded by
On Feb 17, 2006, at 6:36 AM, Robert Webb wrote:
Sorry, this is off topic to asterisk itself, but is about the list
server.
I had a power failure lastnight at home, where my email server
resides, and my network was down for about 20 minutes, that was after
45 minutes of uptime on UPS.
I have some Digium licensed Digium codecs, but when making a call and
transcoding the call is only heard in one direction?
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL
Hi all,
haven't seen many posts about asterisk in Singapore...
Getting a server going there and was wondering if TDM400Ps will be fine in
FCC mode, and if there are indications / cadence values that I should be
putting on there as in other international locations.
Seen an unsettling post on
Thank you, I added both to
SIPDefault.cnf and I am seeing traffic now. Its strange that it would default
to not registering, and wouldnt try to register even if I went into the
phone and did a register 1 1 command.
Im getting a 401 Unauthorized
back from Asterisk now. With the following
Thanks Colin!
Makes sense; I will work on this later
today.
If you can, sending the example would be
great.
Thanks,
--
Christopher T.
Aloi USA
Datanet - Technical Support Engineer 318 South Clinton
Street Syracuse, NY 13202 C: (315) 569 4033 O: (315) 579 7074
E: [EMAIL
Michael Collins wrote:
Just curious to know if anyone uses Festival with * and whether or not
you’ve got a different voice than the default. I’m looking at doing a
commercial application but my boss doesn’t want to shell out the $
before we do some real world testing of * and Festival.
On 2/17/06, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi !
I always use your ARI through AAH, and indeed nice job !
A few comment :
- I have seen that we could use ARI only for the Call Monitor by setting a
value. would it be possible to do the same for only Voicemail ... indeed,
we are
Yes Sir! This is what I use:
http://www.vovida.org/applications/downloads/stun/
Works like a charm! Been running it in production for about a year.
On 2/17/06, Deepak Dhiman [EMAIL PROTECTED] wrote:
Hi friends !
I want to add stun functionality in asterisk.
can anybody give me some hint
1 - 100 of 146 matches
Mail list logo
