Hi,
I want to know, if someone has tried to use clustering
in asterisk to increase its scalability and make it
distributed??
If yes, how easy it is to cluster?
Can someone please ive me details about the same
Thanks
Varun
__
Do you Yahoo!?
Hi
From what I have heard, Asterisk does not allow for iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the jitter
processing.
If lost packet concealment doesnt work with ilbc, I can
assume the same applies to other codecs who claim to have
this
Trilogy India wrote:
Hi,
I want to know, if someone has tried to use clustering
in asterisk to increase its scalability and make it
distributed??
If yes, how easy it is to cluster?
Can someone please ive me details about the same
Thanks
Varun
This has been discussed a number of times in the
On Tue, 3 Aug 2004, Steve Underwood wrote:
Eh? G.729 has no particular features to allow more effective packet loss
concealment. iLBC has, but at the cost of a substantially higher bit
rate. In fact G.711 is a little ahead of G.729 in the regard, since
packets are completely independant.
Thanks for help.
All works now.
Problem was in codecs on different sides
Definity: display ds1 1b14 CRC? n
Interface Companding: mulaw
And when making call via asterisk
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Hello everybody,
I have a strange comportment with oh323 and asterisk, I'start testing
asterisk but with this I can't understant plesae help me !
Thanks
Eltorio
--
1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a
Hi,
I want to know how we can use TDMoE to cluster
asterisk??
And, how many asterisk servers it can cluster and
how??
Varun
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Yahoo! Mail Address AutoComplete - You start. We finish.
http://promotions.yahoo.com/new_mail
On Tue, 2004-08-03 at 03:43, Trilogy India wrote:
Hi,
I want to know how we can use TDMoE to cluster
asterisk??
And, how many asterisk servers it can cluster and
how??
If you want to know how, search the archives, or read the wiki. Once you
have specific questions instead of blanket
On Mon, 2 Aug 2004 12:32:38 -0400, AJ Grinnell [EMAIL PROTECTED] wrote:
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything else will fix it. I just need the one
file from the latest cvs. 8-1-04. Please help
Delete your corrupted app.c
On Mon, 2 Aug 2004 12:54:59 -0700, Alain Bautista
[EMAIL PROTECTED] wrote:
Anyone had experience 'marrying' the two?
We had setup * to front end Artisoft's Televantage.
It works with some issues need to be resolved:
- Inbound calls could not properly handled and routed by Televantage's
Call
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
* -- SIP -- CISCO -- PRI -- PSTN
The PSTN sees no callerid.
*--- PRI[zaptel]-- PSTN
Callerid is there... which makes me think it's the cisco, not the
PRI/PSTN/telco.
CISCO PRI-- * PRI [zaptel]
Callerid IS
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
* -- SIP -- CISCO -- PRI -- PSTN
The PSTN sees no callerid.
*--- PRI[zaptel]-- PSTN
Callerid is there... which makes me think it's the cisco, not the
PRI/PSTN/telco.
CISCO PRI-- * PRI [zaptel]
Callerid IS
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 August 2004 18:39, Deti Fliegl wrote:
Your Extension has to match your MSNs. You have to configure all MSNs
you have in a comma separated list like
msn=27849,27852,27869,27861
and you must only use these MSNs as caller id.
Hi :)
HI ALL;
Is there anybody who use app_radius(astersik radius
module)???
is it stable?
Regards
mohammad
On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote:
From what I have heard, Asterisk does not allow for iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the jitter
processing.
Can you provide a source for that statement? I am not
On Tue, 3 Aug 2004 11:40:28 +0200, Maurizio Marini
[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 August 2004 18:39, Deti Fliegl wrote:
Your Extension has to match your MSNs. You have to configure all MSNs
you have in a comma separated list like
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Jason :)
On Tuesday 03 August 2004 12:07, Jason Williams wrote:
I would set the MSN's to 855285 and 859609
They do not usually include the area code.
[local]
exten = _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1
exten = _9XX.,2,Congestion
exten
Hi all,
Im using a AVM Fritz! Card (TE-Mode) and a Longshine LCS-8051 with HFC-S-Chip (NT-Mode) together with the chan_misdn.
I build the system like it was explained at http://isdn.jolly.de.
At first I used the pbx4linux software from jolly (http://isdn.jolly.de) and then I changed to
Hello All,
Can any Australians who have any info or current patches relating to
Caller ID in Australia please drop me a line? There is little or no
info on the Wiki regarding this topic, although I am aware of a
related patch mentioned in the bug tracker.
Regards,
Rob Barnes
On Fri, Jul 30, 2004 at 03:40:58AM +0200, Jan Czmok wrote:
Date: Fri, 30 Jul 2004 03:40:58 +0200
From: Jan Czmok [EMAIL PROTECTED]
Dear Skinny/SCCP lovers :-)
I've just completed uploaded to the cvs the newest version with fixed
redial key AND implementation of speed dials. please test
Rob,
Caller ID all depends on which hardware you're using.
I can say that if you're using chan_capi (for CAPI compatible ISDN hardware)
caller ID works perfectly.
You'll find getting it working is highly dependant on which hardware and
therefore channel driver you're using.
Regards,
Kimble
A couple things:
In zapata.conf, the channels line should be:
channel = 1-15,17-31
Thanks, now asterisk loads without error.
If you are connected to the PSTN, the signalling should be pri_cpe (customer
premise equipment). But your setting would be correct for connection to a
channel
Hi All,
As a company, we are looking to rationalize our phone system
infrastructure and have come across using a digium quad port E1 PRI
cards in conjunction with the Asterisk PBX software. I'm hoping you'll
be able to answer the following questions and maybe give me a few
configuration
On Mon, 2004-08-02 at 20:32, Bartosz Wegrzyn wrote:
I joined this group 2 weeks ago, because I was having problems with my
asterisk box and broadvoice. I found many discussions regarding similar
issue. I belive that this is the group where we can share our problems and
help each other.
On Tue, 3 Aug 2004, Andrew Kohlsmith wrote:
On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote:
From what I have heard, Asterisk does not allow for iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the jitter
processing.
Can you
The only change I believe I had to make was under
/usr/src/asterisk/channels/chan_zap.c
#define DEFAULT_CIDRINGS 2
The default is 1
Google search if you want some of the previous threads...
http://www.google.com.au/search?q=asterisk+callerid+patch+australiaie=UTF-8
hl=enmeta=
-Original
On Tuesday 03 August 2004 07:18, [EMAIL PROTECTED] wrote:
I am the source for that statement. Is that a problem? ;-)
Not at all. :-) But I do thank you for taking the time to write a few
paragraphs explaining what's going on in the current code. It's certainly
something I didn't know
What an ACTIVE newsgroup!
I'm in the early stages of researching Asterisk. My current environment
is a small college (~1000 sets/~400 student sets), Avaya Definity
G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance,
licensing, and equipment costs are HEFTY.
So.. are there any
On Tue, 3 Aug 2004 20:24:50 +1000, Robert Barnes
[EMAIL PROTECTED] wrote:
Hello All,
Can any Australians who have any info or current patches relating to
Caller ID in Australia please drop me a line? There is little or no
info on the Wiki regarding this topic, although I am aware of a
On Tue, 2004-08-03 at 08:21, Brian Hudson wrote:
What an ACTIVE newsgroup!
I'm in the early stages of researching Asterisk. My current environment
is a small college (~1000 sets/~400 student sets), Avaya Definity
G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
i fixed wrong capi.conf (there was [controller1] after [interfaces])
now capi.conf is:
;
; CAPI config
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=855285,859609
incomingmsn=*
controller=1,2,3,4
A thought occurred to me to on how to further quantify the impact of glare
on a properly dimensioned trunk group and debunk the ground start glare
concern. A cursory traffic analysis clarifies:
1. Assume you have a two-way trunk group, dimensioned for average busy
hour, average busy season for
Just opened my July/August VON Magazine, and as usual started reading
from the back.
SIP at Risk and Asterisk caught my eye, gives */IAX a nice plug.
--
Dave Cotton [EMAIL PROTECTED]
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On Tue, 2004-08-03 at 21:30, Christopher Lee wrote:
The only change I believe I had to make was under
/usr/src/asterisk/channels/chan_zap.c
#define DEFAULT_CIDRINGS 2
The default is 1
This is one of 2 patches I make to asterisk every time I download. It is
needed to make the callerid
Hi guys,
ive heard that the latest version of asterisk can be compiled to run
with the old iax1 protocol as a default.
Any ideas ?
Thanks
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To
Is features.conf included in the cvs as of 8-1-04? I have updated, but am
not seeing it?
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To UNSUBSCRIBE or update options visit:
-Original Message-
From: AJ Grinnell [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 10:28 AM
To: Asterisk
Subject: [Asterisk-Users] features.conf
Is features.conf included in the cvs as of 8-1-04? I have
updated, but am not seeing it?
I think that it should be in
Hi,
I just installed astguiclient, following the SCRATCH_INSTALL, without errors.
But when I try to enter the administration page (http://127.0.0.1/astguiclient/
admin.php), it's blank. The browser shows me the following page source:
htmlbody/body/html
The same happens with
There could be several causes for this. First, check your php.ini file to
see that Globals are turned on.
Did you do a full install from scratch and follow the instructions from the
beginning?
We can continue this off-list as to not annoy everyone with troubleshooting.
MATT---
-Original
I'm working on putting together some Ideas about using Asterisk in our
environment, one of the things I want to consider is DID trunks
(analog), what hardware do I need to terminate these trunks? I'm
looking at the voicetronix openswitch6 or openswitch12.
On the openswitch, I'd like to use some
Where do you set the outgoing mail server for use with asterisks mail
system? I have entered the info in the voicemail.conf file correctly,
but I am still unable to get the voicemail messages via email. I ran a
tcpdump on the system while calling in and leaving a voicemail and I
don't even see
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
My qualification is having worked on the IAX2 jitter buffer,
consequently having studied how audio flows from the received frames
through the jitter buffer and then via ast_translate() into the codec.
Hmm... having
Not in configs or /etc/asterisk/. Asterisk is still running, just curious
why I am not seeing that file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Jackson
Sent: Tuesday, August 03, 2004 10:36 AM
To: [EMAIL PROTECTED]
Subject: RE:
I have a PRI comming into each of 2 buildings. How do I redirect an incomming
call on PRI_A of particular DIDs to arrive at PRI_B instead?
Thanks,
John
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Finally, can I turn off the '#' to transfer, since we're using the
hook-flash (albeit manually) instead? ISTR an option to do this but have
spent the morning trying to find it again unsucessfully...
I think you might want to look at the 'T' and 't' options on the Dial
application, documented
On Tue, 2004-08-03 at 11:22, Sean Garland wrote:
Where do you set the outgoing mail server for use with asterisks mail
system?
It uses the command '/usr/sbin/sendmail -t' by default. You can use the
mailcmd parameter in voicemail.conf to override that. From the wiki:
Mailcmd allows the
Hi,
We are having a problem with asterisk detecting that an analog ext has been
put down. This seems only to happen after a number of calls have been made.
We have an FXO port (TDM400P with FXO module) connected to our PBX and are
using this to test asterisk prior to rolling our for our small
- Original Message -
From: AJ Grinnell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 8:02 AM
Subject: RE: [Asterisk-Users] features.conf
Not in configs or /etc/asterisk/. Asterisk is still running, just curious
why I am not seeing that file.
Hmmm.. Maybe try
Adam Hart wrote:
Steve Underwood wrote:
Adam Hart wrote:
Daniel Niasoff wrote:
Is G729 more sensitive to packet loss or delays due to its higher
compression. If Ive generally got the bandwidth available, am I
best sticking to ulaw.
G.729 has lost packet concealment, G.711 doesn't. G.711 will
Hi Everyone,
After a fair amount of faffing ive managed to get the 2000w working with
asterisk for IP - PSTN calls (i.e. get the phone to make and receive calls
over our BT line). The final solution is to set up outgoing VoIP calls but
I now know that without a SIP aware router I can think
If this works, wouldn't it fix the problem using silence supression as
well
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh
Sent: Tuesday, August 03, 2004 11:22
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] G729 Codec+packet loss
-BEGIN PGP SIGNED MESSAGE-
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Hi,
A thought occurred to me;
Background;
In the early days of cable, the cable people seemed clueless to things like
over selling bandwidth. But as time went along they got it better and better
under control.
Of course their natural
[EMAIL PROTECTED] wrote:
On Tue, 3 Aug 2004, Steve Underwood wrote:
Eh? G.729 has no particular features to allow more effective packet loss
concealment. iLBC has, but at the cost of a substantially higher bit
rate. In fact G.711 is a little ahead of G.729 in the regard, since
packets are
If you want complete control from the Asterisk side you'd probably need a 2B
transfer facility enabled on the PRI to allow Asterisk to tell the central
office to shunt the call elsewhere. However, that functionality isn't in
Asterisk yet:
Andres wrote:
[EMAIL PROTECTED] wrote:
On Tue, 3 Aug 2004, Steve Underwood wrote:
Eh? G.729 has no particular features to allow more effective packet
loss concealment. iLBC has, but at the cost of a substantially
higher bit rate. In fact G.711 is a little ahead of G.729 in the
regard, since
Well, we currently use Rodopi and are trying to find away to use it for
our VOIP billing. However, because it's based on radius I'm unsure if it
will be suitable for Asterisk.
I was just curious if anyone else has used the 2 together before.
- Darren
On Sat, 2004-07-31 at 13:47, [EMAIL
On Tue, 3 Aug 2004 05:47:59 -0400
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED]
wrote:
From what I have heard, Asterisk does not allow for
iLBC to
take advantage of the lost packet concealment.
I understand this has something to do with the
Chris Shaw [EMAIL PROTECTED] wrote:
Not in configs or /etc/asterisk/. Asterisk is still running, just
curious why I am not seeing that file.
Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's
been in there for over a week now, I just checked out a new copy and it's
in
On Fri, 30 Jul 2004, Darren Bentley wrote:
Hello,
Has anyone used Asterisk in conjunction with a billing system like
Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used?
I suffered with Rodopi for three years in a previous life. Avoid it like
the plague.
On Mon, 02 Aug 2004 12:08:34 +1200, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
be it. Most of the unstable behavior has been in GUI based parts:
Gnome in particular. Since no sane person runs * on a machine that is
also running X, it's a non-issue.
Is this always going to be the case?
Kevin Walsh wrote:
Chris Shaw [EMAIL PROTECTED] wrote:
Not in configs or /etc/asterisk/. Asterisk is still running, just
curious why I am not seeing that file.
Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's
been in there for over a week now, I just checked out a
This certainly explains why we get terrible audio at 10% packet loss
between Asterisk servers between 2 end points using iLBC, but if we
use 2 SPA2000s using G.729 to commincate directly with each other
(and having the same 10% packet loss), they sound pretty good. We
had been trying to
Hello,
I am attempting to get the asttapi driver working on Windows XP
Professional, and am running into some strange problems. I've combed the
Web and the Wiki for information on debugging the application to see if I
can solve my issue, but nothing is helping me. I have tried the
On Fri, 30 Jul 2004, Darren Bentley wrote:
Hello,
Has anyone used Asterisk in conjunction with a billing system like
Rodopi? Is the Rodopi VOIP module worth getting, or can radius be
used?
I suffered with Rodopi for three years in a previous life. Avoid it
like
the plague.
OMG.. I had
Josh Roberson [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Or simply rename musiconhold.conf as features.com and restart Asterisk.
no.. WRONG. rename parking.conf, as parking.conf is what features.conf
Oops. I knew it was one of them. At least I didn't say sip.conf :-)
--
_/ _/
Uhm... So ATT pays you back WHAT for the time they're done? So if they
go down all-day Monday, I'll get back... A dollar? Heck, my cellular
provider does better than that. That's not an SLA, that's a simple
refund-agreement... Nobody makes you pay for service you don't receive.
It's also
Ok,
I may have spoken to early, I have * compiled and running on Sparc64/Linux,
tried to configure sip softphones etc., everything works till here.
Yesterday I tried to place a call to the demo but right after the call
is bridged with the
demo sounds it receives a SIGBUS and terminates with Bus
Kevin Walsh wrote:
Josh Roberson [EMAIL PROTECTED] wrote:
no.. WRONG. rename parking.conf, as parking.conf is what features.conf
Oops. I knew it was one of them. At least I didn't say sip.conf :-)
True that. This is another reminder that everyone needs to make sure
that when they
I just got my copy of 'VON Magazine' and there is a 1 page article about
Asterisk titled, SIP at RISK and Asterisk. Here is a small quote:
NAT is the place where SIP messes up the worst-an IP address in the payload
of a SIP signaling packet, generated on one side of a NAT, is likely to be
I would like to integrate * with an existing Altigen PBX. I want to spend
as little money as possible to make it happen. My main goal is to
inexpensively connect a branch office to the phone system. Eventually I
would like to replace the Altigen system with an Asterisk server so I don't
want to
Well, can anyone recommend a full featured ISP billing system that would
handle VOIP/Asterisk?
- Darren
On Tue, 2004-08-03 at 11:09, Brian D'Arcy wrote:
On Fri, 30 Jul 2004, Darren Bentley wrote:
Hello,
Has anyone used Asterisk in conjunction with a billing system like
Rodopi? Is the
Under what circumstances? If the first T1 is down, for example?
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Has anyone integrated asterisk with current version of rt. I followed
the Wiki but I only get as far as hold on while i create a ticket then
it hangs up. I don't see it connect to the rt-soap-server.pl script
running on the console of my rt machine. any help would be greatly
appreciated.
Darren Bentley wrote:
Well, can anyone recommend a full featured ISP billing system that would
handle VOIP/Asterisk?
There is not one solution.
Canned billing solutions never work.
Write your own.
If you cannot code, hire someone that can. (not me)
Jeremy McNamara
While we have not integrated the asterisk CDRs yet it should not be a
problem to do. We our building a billing system for ISP/CLECs that will do
what you want. If you want more information you can contact via email to
[EMAIL PROTECTED] or by calling 910.402.5010
Regards,
Gary Carr
Rich Adamson wrote:
So, for those that don't have any interest in the broadvoice interface
topic, find your delete key. Its not all that hard, really.
Every night when I go to bed I say my prayers. And in those prayers I
include a request that those on the list who are so quick to snap at
- Original Message -
Message: 15
From: Geoff Nordli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Tue, 3 Aug 2004 11:36:05 -0700
Subject: [Asterisk-Users] Integration with Altigen
Reply-To: [EMAIL PROTECTED]
I would like to integrate * with an existing Altigen PBX. I want to
On Tue, 3 Aug 2004, Danial Subhani wrote:
What we see using 'zap show channel 10' is as follows:
debian*CLI zap show channel 10
Echo Cancellation: 128 taps, currently OFF
Actual Hookstate: Offhook
Yep - I get this too, but incoming calls still seem to arrive OK. I don't
use the line
On Tue, 3 Aug 2004, Steve Underwood wrote:
It would probably help if you understood what that table means. It is
very misleading. G.729 has features to mitigate the awfulness of a lost
packet. It has nothing to help conceal lost packets really well. What I
said is correct. If you fudge
Is it possible to play and audio file into a meetme conference for both
parties to hear? I thought I remembered reading something about it, but I
can't find it now. Any help would be greatly appreciated.
Paul
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Wayde Nie wrote:
I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810
comes with a built in Ethernet port and I believe it does SIP too...
Will this mean that I won't need a T1 card and dedicated channel bank? ie.
Asterisk connected over Ethernet with the MC3810 and the
Has anyone used this feature successfully? I 'think' I have a .wav file
that it wants.
Here is what 'file' says:
sf-george.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
bit, mono 8000 Hz
I see the logs on my web server as it tries to access it, but all I get
is a screech out
Well one of the _big_ problems I see right now, is that the cisco ubr924 is
reporting it's
MGCP version as 0.1 and asterisk errors with incompatible version.
Not sure if that is a cisco bug and really should be 1.0 I will upgrade the
IOS and see.
Maybe they do run version 0.1 but I've never seen
Hallo everyone,
at first I would like to say hello to anyone as I am new to this list
and new to asterisk which I find very fascinating.
I am currently using asterisk with the German SIP-Provider sipgate and
with my little ISDN-Line using the Modem-Driver vor I4L.
I upgraded my source tree
Hi
Il mar, 2004-08-03 alle 22:09, Paul Egger ha scritto:
Is it possible to play and audio file into a meetme conference for both
parties to hear? I thought I remembered reading something about it, but I
can't find it now. Any help would be greatly appreciated.
sure. use the call spooling
I just upgraded to RC1 from a two-three month old CVS , and noticed that
during IAX2 calls to my service provider there are periods (usually less
than 10 seconds long, minutes apart) during which the caller can not hear
me, but I can hear the caller fine.
Inter-office calls (SIP-to-SIP) does not
My SPA-3000 finally arrived and I'm trying to get the FXO port on it to
work as if it was a X100P card as far as Asterisk is concerned.
I have Asterisk dialing out over the SPA-3000 FXO port no problem.
The issue I'm having problems with is having the SPA-3000 automatically
forward all incoming
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 03 August 2004 05:14 pm, [EMAIL PROTECTED] wrote:
I just upgraded to RC1 from a two-three month old CVS , and noticed that
during IAX2 calls to my service provider there are periods (usually less
than 10 seconds long, minutes apart)
The issue I'm having problems with is having the SPA-3000 automatically
forward all incoming PSTN calls to the Asterisk mainmenu context (or
ext I guess).
Configure an auto-dial number in the SPA to that it corresponds to
something in the mainmenu context. Like:
PSTN_Caller_Default_DP[2]
Works like a charm Andres. Much appreciated.
On Tue, 2004-08-03 at 16:44 -0500, Andres wrote:
The issue I'm having problems with is having the SPA-3000 automatically
forward all incoming PSTN calls to the Asterisk mainmenu context (or
ext I guess).
Configure an auto-dial number in
I have the need for a slightly odd * configuration for testing purposes. I
have a working * setup with SIP softphones, VoIP trunks and a single X100P
clone for PSTN access.
The PSTN line I'm using for testing is also in use by other folks. For
incoming calls, I'd like to set is up so that *
Dr. Chudobiak,
I do not believe it is possible (yet). I know it is implemented in the snom
4s product but I am pretty sure asterisk cannot handle line appearances. I
will do some further research.
Please use my personal email address [EMAIL PROTECTED] since I
get all the list emails to this
I'm looking for recommendations for UK-based VoIP-PSTN gateways.
They should ideally offer:
- IAX connection
- Multiple simultaneous calls on a single account
- Lower call rates than BT Business
- Auto-top up or invoicing in arrears
I can find several that offer one of
Hi..
Has anybody been experiencing any problems with transfers
using # to transfer after taking a call off of hold?
Transfers using the # and music on hold work fine by
themselves. However, when we place somebody on hold we can no longer use the #
to transfer. This is a problem since
- Original Message -
From: Stephen Hon
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 3:48 PM
Subject: [Asterisk-Users] problems with'#' transfer after hold...
Hi..
Has anybody been experiencing any problems with transfers using # to
transfer
after taking a call off of hold?
On Tuesday 03 August 2004 12:07, Steve Szmidt wrote:
But with VoIP it has to go both ways and things like latency can easily
become a big issue. (I have cable and it seems that I get sound
degradations much easier than I'm comfortable with, yes it's a shared
connection with occational POP
Hi David-
You may want to post this in the asterisk-biz section, you'll probably get
more leads there..
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 4:05 PM
Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL
On Tuesday 03 August 2004 12:07, Steve Szmidt wrote:
But with VoIP it has to go both ways and
Thanks for the vote of confidence guys. We just bought an
ISP that uses rodopi exclusively for Accounting and Billing.
...sigh...
-e
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
[EMAIL PROTECTED]
Sent: Tuesday, August 03, 2004 12:15 PM
To:
That sigh will turn to cursing after a couple of months. We currently use
Rodopi, have for 10 years but the inflexability is too much to deal with
anymore so we are moving away from it.
Gary
- Original Message -
From: Ejay Hire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday,
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