I got the same problem, look in the thread
ztdummy / timer problem with kernel 2.6.14.
But when I compile a new kernel back to
2.4.31 I managed to play the music for some more secs, and the shoutcast music
is working fine. If you do a zttest which results do you get? And what kernel
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
How could we arrange this problem ? We want to use a sip soft and have
the possibility to do attended transfer
Register only one line on softphone.
--
Tomislav Parcina
[EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf
globals section and uncommenting automon=*1 in features.conf, nothing
happens when pressing *1
Solved that...
When I change blinsxfer in features.conf to
I'm using kernel 2.6.9-22.0.1.EL, my asterisk has no zap card.
but my another E1 box suffers same no music problem
using 1.2.1. with kernel 2.4. zttest result
worst 99.86, average 99.95. the same E1 box running 1.0.9 version do have music
played fine.
Best Regards
matt
- Original
As soon as they port it to Gentoo I'll try it out... ;-)
Evert
Kerry Garrison wrote:
Everyone should simply uninstall Skype and switch to the Gizmo project
because it interfaces quite nicely with Asterisk.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949)
Just wondering...
Has someone ever contacted Skype/Ebay and asked them about their point of
view/opinion on interfacing with SIP / Asterisk? 8-)
Regards,
Evert
[EMAIL PROTECTED] wrote:
I sincerely believe that it's completely non-sense to make a channel for
Skype.
Skype is a
We have found the reason for the memory consumption problem:
It was caused by the ODBC applications: ODBCget and ODBCput. These applications
are not a part of the standard Asterisk package. Every time they were used, the
memory footprint was increased by 12 kB.
Using the MYSQL
On Mon, Dec 19, 2005 at 02:56:30PM -0800, Kerry Garrison wrote:
Yes you can send and receive calls via Asterisk.
http://voipspeak.net/index.php?/content/view/19/28/
Which demostrates how to connect to sipphone.com . This is very simple,
indeed. But what about text chats with gizmo-project
Hello,
Polycom phones send SUBSCRIBE message for login .
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
On 12/18/05, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello,
Polycom ip soundpoint support ACD login/logout .
Can we configure asterisk with polycom ACD
support?
Like the
Hello Douglas,
You can enable ACD in sip.cfg and phone.cfg of the
polycom phones.
Polycom send SUBSCRIBE to server .
Harry
--- Douglas Garstang [EMAIL PROTECTED] a écrit
:
You can configure Asterisk to log an agent in/out of
an ACD Queue with AgentCallBacklogin, AddAgent etc,
just like with
http://www.gizmoproject.com/
from the website, it quite looks like skype - no network setup, IM
integration, you may call POTS phones by paying the company who did it.
not very useful, in the end, if the purpouse is asterisk-skype
interoperability - I doubt that every one of the millions
Hi,
I have this problem. When I call a GSM number with the IDSN line if the
GSM phone I not hear the messages of operator but I hear the ring.
I suppose that the problem is that asterisk waits a response that is yet
arrived.
Any idea?
This is my extension:
exten = _0XX.,1,Wait,1
exten =
On Tue, Dec 20, 2005 at 12:31:01PM +0100, Lenz wrote:
http://www.gizmoproject.com/
from the website, it quite looks like skype - no network setup, IM
integration, you may call POTS phones by paying the company who did it.
not very useful, in the end, if the purpouse is asterisk-skype
On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote:
...
And this is bad for us. With Gizmo we can talk. With google talk we have
stand a chance of talking. But we're blocked from Skype.
since you cite it, what compatibility is there with google talk ?
any pointer to descriptions of
On 12/16/05, Jason Kim [EMAIL PROTECTED] wrote:
To solve echo problems, I'm considering 2
alternatives.
1 Sangoma A104d
- I can't find support for asterisk 1.2.1
This is an interesting possibilty too - Even if Asterisk 1.2.1 is not
yet supported. I currently make heavy use of the
Tim Connolly [EMAIL PROTECTED] uttered the following thing:
I'm looking for ideas on how to implement voicemail notification on
Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone
would be perfect. Even maybe go so far as a quick ring to the extension
every 15 minutes
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote:
On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote:
...
And this is bad for us. With Gizmo we can talk. With google talk we have
stand a chance of talking. But we're blocked from Skype.
since you cite it, what
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote:
On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote:
...
And this is bad for us. With Gizmo we can talk. With google talk we have
stand a chance of talking. But we're blocked from Skype.
since you cite it, what
Hi all,
Is there a way (via AMP ) to route a call based on the DID , or better
based on the inbound channel number
I had non problems doing that with AMP and with digium PRI Cards, but now
for the first time I am trying to use a TDM400 card
It seems that DID is non populated.
I also tryed
Hi All
I have some remote users signed in tot he asterisk from various locations. Most
of them are fine but a couple of users every now and again will become
unreachable
Name/username HostDyn Nat ACL Port Status
100101692/100101692 xx.xx.xx.xxD N
On 12/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello Douglas,
You can enable ACD in sip.cfg and phone.cfg of the
polycom phones.
Polycom send SUBSCRIBE to server .
Harry
--- Douglas Garstang [EMAIL PROTECTED] a écrit
:
You can configure Asterisk to log an agent in/out of
an
Is there a way (via AMP ) to route a call based on the DID , or better
based on the inbound channel number
I had non problems doing that with AMP and with digium PRI Cards, but now
for the first time I am trying to use a TDM400 card
It seems that DID is non populated.
There is no such
Il 12:38, martedì 20 dicembre 2005, Matteo Piazza ha scritto:
Hi,
I have this problem. When I call a GSM number with the IDSN line if the
GSM phone I not hear the messages of operator but I hear the ring.
I suppose that the problem is that asterisk waits a response that is yet
arrived.
Any
I have some remote users signed in tot he asterisk from various locations.
Most of
them are fine but a couple of users every now and again will become unreachable
Name/username HostDyn Nat ACL Port Status
100101692/100101692 xx.xx.xx.xxD N
So, in theory, if the agent channel had a hint
extension and
provided device state updates based on whether the
agent was logged in
or out, this would probably work then as the buddy
watch feature does
now with the Polycoms.
This is probably worth some further investigation.
What
Buenas mi nombre es Will y queria pedirle donde puedo descargar el Sistema
Operativo Debian 1:3.3.5-8ubuntu2
Gracias
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Hi,
I need to develop a project in which the user can phone a number, say
something and the voice will be output to a speaker, if the user want to
select other actions, he could just press a number on the keypad, e.g.:
press 1.
I did it with the following:
1. make a incoming context, looks like:
Cory Andrews wrote:
I am trying to compile and document and list of server makes and models
that are physically compatible with the new Digium TDM2400 series cards,
and which meet both the unique power requirements as well as physical
chassis space requirements necessary to successfully
Existe un Debian 3.1 y un Ubuntu 5.1. Si hay un Debian 1:3.3.5-8ubuntu2 debe ser algo muy viejo.
Ademas, te comento que esta lista es en ingles. Trata de postear en ingles.
This list is in english. Please try to post in that language so more people can help you.
On 12/20/05, Will Velez [EMAIL
Hello,
I am currently testing the Sangoma a104d echo-can card and comparing
it to the Digium TE406P. I should be ready to release my testing
results next week, I was a bit sidetracked by some T1 problems which
have delayed my review.
Even though the a104d is not supported on 1.2. I did have it
Hi,
I want to be able to receive incoming calls via H323 to Asterisk for SIP Conversion and then send the Call to a seperate machine running SER to route the call to the end user CPE. However if the call is not answered, I want to be able to send that call back to the machine with Asteriskon it
I had MOH working with 1.0.9, but now it keeps
showing the following log message
Dec 20 11:45:05 WARNING[30548]: interface.c:215
decodeMP3: Junk at the beginning of frame 54414700
And no moh is being played!
- Original Message -
From:
Fredrik Emil
Jensen
To:
I some have problems with TE205P E1 PRI card.
The Asterisk installation is as follows:
Zaptel-1.2 drivers with Asterisk CVS HEAD from
November First
1 TE205P E1 card with span 1 connected to the PSTN and
span 2 connected to
the digital E1 interface card of a Panasonic super
hybrid PBX.
1
I have an Asterisk system for a small office with 12
extensions. For parts of the incoming dialplan that go to support/sales
we have phones ring various people in an additive fashion. Example:
- snip --
exten = s,2,Dial(${E25}|18)
exten = s,3,Dial(${E25}${E24}|12)
exten =
i want to forward a
call after the dial is not succesfull.
But the problem is
when the phone is not registered i get this error:
Dec 20 15:01:45
VERBOSE[15092] logger.c: -- Executing
Set("SCCP/1000131-000b", "LANGUAGE()=de")Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing
On Tue, December 20, 2005 9:13, Tomislav Parcina said:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf
globals section and uncommenting automon=*1 in features.conf, nothing
happens when pressing *1
Solved
Hints and queues are different things.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20, 2005 6:03 AM
To: BJ Weschke
Cc: asterisk-users@lists.digium.com; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ACD with polycom ip phones
So, in
The hint extension, in conjunction with Polycom's buddy feature, allows you to
have a busy lamp field against appearances. As far as I know, it isn't
connected with Asterisk ACD queues in any way. It would be nice if there was
some way to get a graphical icon showing when an agent was logged in
Harry,
This will log an appearance in permanently. ie while the phone is active. There
is no way to log an appearance in or out that has the acd feature enabled
against it (that I know of).
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20,
Hi Alberto. I never have used FastAGI, usually i use MAGI(), but AFIAK, just use a pipe to pass the arguments:
Agi(script_source|firstargument|nextone|andnextone|etc)
if youre curiose enough, you can check this on the asterisk source
res/res_agi.c function static int agi_exec_full(). It does the
Hi Johnathan, this will help.
Exten =
611000,1,Answer()
Exten =
611000,2,DigitTimeout(3)
Exten =
611000,3,Background(/var/lib/asterisk/sounds/Welcome)
Exten = _XX,1,SetVar(entereddigits=${EXTEN})
Exten = _XX,2,Authenticate(${entereddigits},a)
Exten =
Is there an option to specify a stun server in those IP phones?
On 12/19/05, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings all,A couple of clients have recently decided they'd like extensions to theiroffice PBXs at their homes, so they've duly been provided with preconfiguredphones which
On Tue, 2005-12-20 at 09:05 -0400, Will Velez wrote:
Buenas mi nombre es Will y queria pedirle donde puedo descargar el Sistema
Operativo Debian 1:3.3.5-8ubuntu2
Esta es una lista de Asterisk. No obstante puedes mirar en la seccion
download de www.debian.org o de www.ubuntulinux.org
Try to
Would
someone be so kind as to point out what stupid little mistake I have made.
I thought I did everything according to the setup page but I fail to
register.
HOSTS
file contains
147.135.8.128 sip.broadvoice.com
SIP.CONF
[general]context=iaxclients; Default context for incoming
Is it possible from within the dialplan to determine if an Agent channel is
already a member of
a queue? Would like to use this as part of a check that will play a message if
the agent is the
last person to log off the queue.
I can sorta do it by using AddQueueMember and checking
I try posting again... unfortunately, this time I'm not able to solve by myself!!!
I've checked zttool and I've got no alarm just OK.
No other news.
Please HELP!
Thanks!-- .:FaberK:.
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- Original Message -
From: Guillermo Salas M [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 20, 2005 10:55 AM
Subject: Re: [Asterisk-Users] Soporte
On Tue, 2005-12-20 at 09:05
Try
register = 7723821447:[EMAIL PROTECTED]/7723821447
That works for me.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 20 Dec 2005, Shawn Porter wrote:
Would someone be so kind as to point out what stupid little mistake I have
made. I thought I did
What password are you using? This is the special one they created for you
correct? This should not the one that you created on your own (that you use
to log in).
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=7723821447
Hey Rich,
As it turns out, this wasn't a firewall configuration problem at all.
I had the firewall cofiguration nailed from the outset. This was just
me not being used to how AMP works ( new to AMP ).
Thanks though, I apprecaite the offer for help!
Sean
Rich Adamson wrote:
I've got
Can I use asterisk as a SIP or IAX to PSTN gateaway like Cirpack ? if
not does anyone know of an opensource tool that would do it?
Thanx in advance
Patcito
-- Why thou shall not use Skype:http://fossvoip.blogspot.com/2005/12/why-thou-shall-not-use-skype.html
I was just on the phone with BroadVoice support, when the engineer
placed me hold. Low and behold, my own MOH was engaged seconds later.
I've never experienced this until 1.2.1. Has anyone else experienced
such an oddity?-- Ronald LewisDenver, Coloradoron (at) ronaldlewis.comwww.ronaldlewis.com
Thanks Steven
Works great.
They should put a little more detail in the setup page as to where you get
that password!!
very difficult to figure to that out in the wee hours of the morning.
Shawn
-Original Message-
From: Steven Job [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20, 2005
Are
queues in 1.2.1 completely screwed up or what?
The
'announce' command doesn't do anything. I'm getting this on the console which
may be related:
Dec 20
10:32:00 WARNING[3467]: file.c:508 ast_openstream_full: File 0 does not exist in
any formatDec 20 10:32:00 WARNING[3467]: file.c:820
Ce fel de ajutor ai nevoie. Trimite-mi un
email.
Best regards,
Chris HARIGA
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FaberK
Sent: Tuesday, December 20, 2005
11:07 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
very difficult to figure to that out in the wee hours of the morning.
Yes it is!!!
I made the same mistake... I think everyone does at first since it is hidden
way deep in a tab of some screen.
But I'm glad I was able to help. :-)
- Original Message -
From: Shawn Porter [EMAIL
Confirmed.
If I
place a file called 0.gsm in /var/lib/asterisk/sounds it plays the file. Why the
HELL when I have 'announce = inside-sales' in queues.conf, is Asterisk trying to
play a file called 0.gsm??? My god... Does digium have any sort of QA process
for Asterisk???
Doug.
I am going to step out on a limb and guess that you need to hangup the call
when the digits are received which would move you on to the next priority where
you could then enter your loop. This is an autoattendant I made at some point
when I was playing around with how to do it.
Maybe it will
- Original Message -
From: John Reynolds
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Tuesday, December 20, 2005 4:13 AM
Subject: Re: [Asterisk-Users] Toll Free Providers
I have nufone.net for 800 and all seems fine... although my useage is very
Can anyone confirm that when using the G729 codec from http://kvin.lv/
pub/Linux/Asterisk/ and a Cisco gateway running IOS 12.4, codec
negotiation fails? When I configure the dial-peer in the router with
g729r8, it fails. If I use g729br8 (which uses a built-in VAD), it
works. This
Is anybody running * on a SunFire X4100? If so, any issues?
Peder
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Does _Digium_ have any sort of QA process? The hardware makers? I
thought Asterisk was open source ;)
Douglas Garstang wrote:
Confirmed.
If I place a file called 0.gsm in /var/lib/asterisk/sounds it plays the
file. Why the HELL when I have 'announce = inside-sales' in queues.conf,
is
Hi,
Is there a CLI or manager command that allow me to
know whether a meet me room is locked or unlocked?
lv09*CLI meetme list 3User #: 02
herbertarauj Herbert Araujo Channel:
SIP/herbert.araujo-0929 (unmonitored)1 users in that
conference.
Thank youDov
If you look at the changelist, most of the contributors are Digium employees.
-Original Message-
From: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20, 2005 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi,
I have a more general question.
Our group over 5000 employees world wide wants to do
a survey for all employees asking them if they are happy with the job, salary,
environment etc
Can I use an * where people can call a certain phonenumber,
go through voice menues and entering
Hi all,
Is there a solution to run Asterisk on Compact PCI
platform ?
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Messenger
! Découvez les tarifs exceptionnels pour
Sorry, I was actually referring to the fact that since it's open source,
it's not in Digium's court to be QA necessarily. While they may be
major backers and contributors, I feel it's up to bug testers and the
public in general to be QA. There're just so many more of them than
Digium
Does anyone have distinctive ring working with Asterisk? Could you
share your zapata.conf and relevent extensions.conf?
Thanks.
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Mojo with Horan Company, LLC wrote:
Does _Digium_ have any sort of QA process? The hardware makers? I
thought Asterisk was open source ;)
Douglas Garstang wrote:
Confirmed.
If I place a file called 0.gsm in /var/lib/asterisk/sounds it plays
the file. Why the HELL when I have 'announce
I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic
PBX to Asterisk. Can anyone recommend a stable and reliable one?
Thanks,
Waldo
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Tom Vile wrote:
Looking for a good toll free DID provider. Any suggestions?
All ready tried Sellvoip and Gafachi and the experience was not desirable.
Thanks,
Tom Vile
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Hello,
I've been looking for a way to merge two existing asterisk channels
manually through the manager interface, but have been unable to find
any support for this. Does anyone know if it exist or if there is
something out there that might accomplish this?
Thanks,
~Senyo
Serge Schumacher a écrit :
Hi,
I have a more general question.
Our group over 5000 employees’ world wide wants to do a survey for all
employees asking them if they are happy with the job, salary,
environment etc…
Can I use an * where people can call a certain phonenumber, go through
What is the proper way to email to multiple email addresses.
I have been intending to also email my cell phone when there is a message, but
have yet to try different options like comma,
semicolon, etc.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
Thnx for the fast reply,
What does tanscoding mean ?
I also think that the calls are not the problem but playing about 120 voice
files on 240 channels simoultanously ??
Cluster ? DNS ? Anyhow the voices playback can only be done on the maschine
having the E1 cards connected I suppose ?
That's probably because I've never received any from you Brian.
Just because it's open source doesn't mean it has to be crap.
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 20, 2005 12:17 PM
To: Asterisk Users Mailing List - Non-Commercial
I have been wondering the same thing. I would like to be able to link 2
channels inside an AGI script.
Also, a way to send variables back-and-forth.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Senyo
Sent: Tuesday, December 20, 2005 2:33 PM
I have seen in several places where the analog adapters you can use for faxes
and modems have intermittent problems. Is this an issue with asterisk and are
there work-arounds?
We currently have side-stepped the issue by sharing the analog line between the
tdm card and the fax machine (I
Robert,
This configuration is working fine for me (In ontario with Bell Canada)
dring1 is the 2nd ring pattern on our line, it is a double-ring
dring3 is the regular ring, which I wanted to ignore but since you cant do
that I just send it to a wait loop
ZAPATA.CONF
[channels]
usercallerid=yes
On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote:
That's probably because I've never received any from you Brian.
Just because it's open source doesn't mean it has to be crap.
Will someone please give this guy a refund so he can go and spend his
money somewhere else.
--
Peter Bowyer
LOL, I agree... if someone doesn't like it, write some code and make it
better, or use something else :)
Aaron
Peter Bowyer wrote:
On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote:
That's probably because I've never received any from you Brian.
Just because it's open source doesn't
Hi,
I have been battling with the following problems for a while and was
hoping someone could shed some light on the subject.
I am using AT320 402 IAX2 phones with 1.49 firmware (latest) connected
to an Asterisk server running [EMAIL PROTECTED] 2.1 (includes Asterisk 1.2 and AMP
1.10).
I am
I know it's not the same thing, but put the phones
into a queue, and call the queuemaybe...
PaulH
- Original Message -
From:
Ryan Booz
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, December 21, 2005 1:03
AM
Subject:
I wrote a patch to do just this quite a while ago. Have been using it
in production since Asterisk 1.0.6. Here's the bug tracker link:
http://bugs.digium.com/view.php?id=4297
action_bridge-updated-10-12.txt is newer/better not written by me :)
Kevin promised it would go in 1.2.0 release
I don't know about you, but my option 4 says change password yet when
I press it, it does give me the option to remove the temp
greeting. A bit confusing - I agree.On 12/12/05, Matt [EMAIL PROTECTED] wrote:
Thank you that will do it.Wow that was slightly not intuitive :)On 12/12/05, Steve Blair
Hi,
Do you think * could play around 300 voicemenu messages simoultanously?
Regs,
Serge
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As mentioned before, the new Sangoma card coming out over New Year is a
reliable and very good quality card. You can start with a 2-port module
and add on more as you need, up to 16 ports (mix of dual FXO/FXS
modules).
Best of all, it fits in a 2U server!
Thanks
Bjorn
-Original
I would recommend the new Sangoma analog card coming out over New Year.
It's built on the same quality standard and technical solution as the T1
cards.
You can mix and match FXO/FXS cards.
The other analog cards tends to have a lot of echo problems, but this
one seems a lot better.
To get fax
We've had over 500 active users of the IVR on a server at one time.
That server was pretty full, but our current servers are beefier. You
really need to look at the hardware more than what Asterisk can do.
On Tue, 2005-12-20 at 22:22 +0100, Serge Schumacher wrote:
Hi,
Do you think *
Does Asterisk have SIP support for PRACK/100rel?
Asterisk does also seem to filter out some SIP header fields. Is there
a way I can force Asterisk to pass on ALL SIP header fields?
Thanks,
trond
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I'm trying not to use a server. It's more of an appliance to connect
to a Panasonic PBX. I saw some from Grandstream and Sipura. There are
others but I'm just trying to get feedback as to who has successfully
used them with Asterisk. I only need 2 FXO ports.
Thanks,
Waldo
On Dec 20, 2005,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of Brian Capouch
Sent: Tuesday, 20 December 2005 06:06
Mark Hulber wrote:
The paper is definitely interesting and I commend them for
their effort but it doesn't represent a complete
Serge,
How are you going to be building this server? I am not going to claim to
be any sort of expert on
sizing, but I do have some experience as an IVR
designer/developer.
In one
of your previous posts you mention E1 cards. In order to get 300 msgs at
once you would need to
be
So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all
your SIP subscriptions. Nice. Basically that means the use of hints and
subscriptions in a production environment is a completely impossible. Awesome.
Considering traditional phone users have come to expect this
my own
criticism. I just talked with a friend about erlang tables.
completely blows away all the stuff I just wrote below...
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Shawn
PorterSent: Tuesday, December 20, 2005 4:47 PMTo:
Serge Schumacher a écrit :
Thnx for the fast reply,
What does tanscoding mean ?
It means translating from one codec to another, typically g.729 -
g.711. You mostly need to worry about it if you're doing VoIP.
I also think that the calls are not the problem but playing about 120 voice
On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote:
So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all
your SIP subscriptions. Nice. Basically that means the use of hints and
subscriptions in a production environment is a completely impossible.
Awesome.
Shawn Porter a écrit :
my own criticism. I just talked with a friend about erlang tables.
completely blows away all the stuff I just wrote below...
Well, the bloke says they have 5,000 employees /worldwide/. Clearly the
BHT should be significantly lower than if they were all in the same
With 2 quad E1 I can handle 240 channels
at the same time, so those 240 channels have to playback voicemenus on the same
time.
The survey will have about 30 questions
and the complete survey will take about 20-30 minutes
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: mardi 20 décembre 2005 23:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IVR and db
Serge Schumacher a écrit :
Thnx for the fast
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