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Hello
On 17/07/2004, at 4:17 PM, Mark Spencer wrote:
ftp://ftp.digium.com/pub/asterisk
Can someone grab a copy and put it on a mirror server?
Jean-Yves
- ---
Jean-Yves Avenard
Hydrix Pty Ltd - Embedding the net
www.hydrix.com | fax +61 3 95093717 | phon
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Hello Mark.
thank you heaps for this.
I also just updated our CVS source copy and it doesn't compile anymore
it stops in the pbx directory with hundreds of GTK errors.
I'm running RedHat 9.0 Linux (with 2.4.26 kernel)
Now, will download the files from
Nicolas Gudino wrote:
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
chan_sip2 support
We have officially made the first release candidate of Zaptel, Libpri,
Asterisk and Gastman available. While there are still open major bugs,
they are relatively limited, and it was time to go ahead and get the 1.0
ball rolling in earnest.
ftp://ftp.digium.com/pub/asterisk
Enjoy the code. Spe
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You should read the manual of your snom then (available from the
internal website on the snom phone)
Basically you press the a->1 soft-menu button at the top.
Jean-Yves
On 17/07/2004, at 1:23 PM, Rana Dutt wrote:
My SNOM 200 phone got into a funny mod
Has anyone had any success in getting the Cisco 7905/7912 phones working
with a managed switch and VLAN's?
I programmed a voice VLAN and changed the parameter in the telephones
for the new VLAN. The individual ports are set up for tagging. It
seems to work fine with the Cisco 7960/7940 telephone
My SNOM 200 phone got into a funny mode where if I dial any digit, a letter
gets displayed and sent, so dialing no longer works. For example, if I dial
"9", the letter "w" gets displayed and sent when I press OK. How do I get it
out of this mode?
___
Ast
Just shoved my 7960 onto one of my clients networks. I'm talking from
outside the office to them (they are using a GS BT101). I'm getting the
following message repeatedly in the log, whenever someone talks.
Jul 17 12:29:40 NOTICE[96402352]: frame.c:120 ast_smoother_feed:
Dropping extra frame of
I'm trying to access feature codes remotely over a channelised T1 between a
Norstar MICS (rev 4) and Asterisk. The timeslots are configured E&M and have
been working fine under most circumstances except this one. There is mention
of accessing the facility by calling Flash() from within extensions.c
Hello,
What version of the astguiclient suite are you using?
What version of PHP are you using?
Do you have GLOBAL_VARS turned on or off?
It's very strange that being a POST all of the variables seem to be showing
up on the URL like a GET would. also it doesn't sem to be submitting to the
admin
Hello,
I’m very new with * and I would
really appreciate some help to implement a SIP to PSTN
Gateway.
My current scenario includes an *
box with a TE405P board. I have a 1.5Mb connection to the outside world (using a
router with firewall capabilities) and channel banks that allow me
> It's a way to specify a DNS via config file which has
> priority over whatever is handed out from DHCP.
>
> (Optional) IP address of a new dynamic DNS server. If a new
> DNS server address is specified, it is used for any further
> DNS requests after the phone uses the initial DNS address
>
Quoting Marty Mastera <[EMAIL PROTECTED]>:
> Hello everyone
>
> Searching the archives and google always comes up with entries regarding
> the "dyn" dns option in the 7960, but I can't find answers to my
> specific question
It's a way to specify a DNS via config file which has priority o
Line 1 is always the center two pins. Line2 (on a two line analog jack) are
the pins just to the outside of the center two pins.
Now you know why the center two pins are pair 1 and the pins to the outside
of them is pair 2. The rj11/12 jack will plug into the wider rj45 just
fine. You just have
Please disregard, I have 'solved' the issue.
Thank you,
Chris
Christopher L. Wade wrote:
Hi,
I'm am currently in the process of trying to integrate an * box with an
NEC Electra Elite IPK.
Currently, we have 7 POTS lines coming into our building. These lines
are plugged into our NEC using the ap
Hello everyone
Searching the archives and google always comes up with entries regarding
the "dyn" dns option in the 7960, but I can't find answers to my
specific question
My 7960 is connected via cable modem and is NAT'ed (everything is
working fine). On the 7960 under SIP configuration\
On Fri, 16 Jul 2004 16:36:34 +0200, Roger Schreiter
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have to develop a phone application using asterisk's
> chan_oss.
>
> When the phone is idle, i.e. the last command was a hangup,
> one hears a "toot, toot, toot, ..."
> But unforuntaly its use is in Germany
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
> After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds "transfering" I'm sorry that was an invalade
selection. Sometim
On 17/07/2004, at 7:13 AM, James H. Thompson wrote:
Looking for WLAN - WiSIP - WiFi phone recommendations and experiences.
What works, what doesn't.
I'm using a ZyXel Prestige 2000W. It works, and sounds OK.
Call hold, transfer, forward (and anything else that makes the phone
worth using as a SIP
The "SPA User Guide" now covers configuring the 3000.
http://www.sipura.com/support/index.htm
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.
Just use a standard phone cable. It will seat properly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florin
Andrei
Sent: Friday, July 16, 2004 5:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue
I just received a
I just received a Wildcard TDM400P by FedEx yesterday. I noticed that
the FXO/FXS modules use connectors similar to Ethernet.
Now, i want to connect the TDM400P to the PSTN connector in the wall,
and also to a regular analog phone. Both the PSTN conn and the phone use
smaller connectors, typical f
Hi Matthias,
On Fri, 2004-07-16 at 18:28, Matthias Endler wrote:
> Hi all,
>
> is it possible to receive SIP/IAX register and unregister events via the
> manager API (like in CLI)? I do receive all kinds of call events
> (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
chan_s
Hi all,
is it possible to receive SIP/IAX register and unregister events via the
manager API (like in CLI)? I do receive all kinds of call events
(Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink).
My manager.conf looks like this:
[general]
enabled = yes
port = 5038
bindaddr = 0
Looking for WLAN - WiSIP - WiFi phone recommendations and
experiences.
What works, what doesn't.
Thanks.
Jim
James H. Thompson[EMAIL PROTECTED]
I have installed the Asterisk gui client that is available off of sourceforge.net. I
was curious if anybody here has used it and what experiences they have had with it.
I am having a problem with it, I am able to use the admin page except when I try to
submit information to the server to add ph
you may want to contact your state's PUC. sounds like you want to be a
long distance carrier and each state will have different
regulations/requirements in addition to federal rules. from everything
i have read this all is still a very "grey" area
- hcir
On Jul 16, 2004, at 3:09 AM, John G
Hi,
I'm am currently in the process of trying to integrate an * box with an
NEC Electra Elite IPK.
Currently, we have 7 POTS lines coming into our building. These lines
are plugged into our NEC using the appropriate analog line interface
card from NEC. The NEC effectively has NO configuration
They will disappear in 15 seconds... they are waiting around just in case we
get a late response on register.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of M3 Freak
> Sent: Friday, July 16, 2004 12:58 PM
> To: [EMAIL PROTECTED]
Rich Adamson wrote:
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but
I have a hard time to forward all my outgoing calls to my SIP
connection.
For incoming calls I set up the registry and it is working perfectly
How can I tell asterisk to forward all calls beginning with 9
to
Gate.Sipserver.com
User : userxx
Password : pwxx
Best regards,
Ha
> On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
>
> > No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
> > or x100p running any Head cvs after June 23rd (totally stock install).
> >
> > Wouldn't necessarily recommend this box for any commercial production
> > use, bu
[EMAIL PROTECTED] wrote:
From the CLI and during a call I want to be able to:
*** Pulse the outgoing line and record at least 50 ms of the incoming line.
The pulse waveform must be specifiable as a series of amplitudes
for each 1/8000 sec time slot. It would be best of these values
http://bugs.digium.com/bug_view_page.php?bug_id=0001188
This patch unifies the code that decides on the location of a mailbox and
stores voicemail in a tree-like structure, to be prepared for very large
volumes of voicemailboxes in one file system.
It's disputed whether this affects performance on
I cannot get Festival to work with asterisk. I have
the following:
exten => 555,1,Answerexten =>
555,2,Festival(mary has a little lamb)exten => 555,3,Hangup
I get the following from asterisk: "Festival returned ER" and the festival logs shows the
following:
client(1) Fri Jul 16 15:35:54
Hello All,
I have a large customer with close to 30 offices worldwide who want a VOIP
solution. We have already implemented
it in Singapore and India, but as usual the Corporate IT department which
is in London is pushing for the Cisco call
manager for all their locations.
I dont know a
Thanks for all the suggestions, unfortunately none of them worked. I
could not for the life of me get phonegaim to compile. Had to change the
automake calls in the 20 makefiles and it was still hosed up. I ended up
just running xlite under wine. Works perfectly. Kind of a shame though.
chris
On T
I'm getting red alarms with my T100P card that last from 4-15 seconds.
They seem to happen randomly, every couple of days.
Has anyone seen this behavior, or does anyone have any ideas regarding
what would be causing it?
Thanks,
Glen
___
Asterisk-
On Fri, 2004-07-16 at 13:28, Eric Wieling wrote:
> I'm having an oddball issue with a Polycom SoundPoint IP 500. As you
> can see below Asterisk thinks there are 2 SIP channels active, but show
> channels tells me there are no calls active. Anyone have any idea why
> this is happening? The Po
http://bugs.digium.com/bug_view_page.php?bug_id=0001693
This patch adds a lot of options for AgentLogin/AgentCallbackLogin
Please test and respond in the bug tracker!
/O
-
"This patch adds quite a few new features i
Hi,
maybe my fault was, that I named the gatekeeper
in h323.conf.
When I comment those lines, just tell in gnugk about
the gateway ip address (asterisk), it's fine:
| alias=myname
| ;gatekeeper = 123.456.789.12
| ;AllowGKRouted = yes
|
| [myname]
| type=h323
| prefix=0,1,2,3,4,5,6,7,8,9
Roger.
Rog
I am presuming your Asterisk box is behind your ISP. You don't actually
need user/pw to send somebody email in the outside world, but your ISP
has prevented you from _directly_ sending email to anybody and make you
go through their SMTP server which forces you to authenticate with it
like a Mail Us
On Friday 16 July 2004 12:43, W. Kevin Hunt wrote:
> First let me say that normal cheapy PC hardware couldn't be made to
> function with out echo. We tried on both the single port Digium T1 card
> and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/
> hardware scsi raid we had ech
On Jul 16, 2004, at 11:07 AM, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but...
What's common and not so c
Hi
there,
I have been looking
for information about radius authentication for SIP module on registration
process, any help will be appreciated.
Regards
Conexion GroupGildo J.
Cáceres M.Developer[EMAIL PROTECTED]www.conexiongroup
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
> No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
> or x100p running any Head cvs after June 23rd (totally stock install).
>
> Wouldn't necessarily recommend this box for any commercial production
> use, but...
>
> What
We're using the "Supermicro X5DPE-G2" in both of our asterisk boxes. They
have 3 (I think) PCI-X (133mhz) 64 bit slots. We're only using 2 of them
(slots) and their not sharing IRQs. We have one pstn gateway that talks to
the pri and the channel bank. The other machine is the pbx (and vm) where
I'm having an oddball issue with a Polycom SoundPoint IP 500. As you
can see below Asterisk thinks there are 2 SIP channels active, but show
channels tells me there are no calls active. Anyone have any idea why
this is happening? The Polycom occasionally stops accepting calls and
requires a
Tony Nichols wrote:
> After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds "transfering" I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digi
> After speaking with several people, and even participating in a forum of
> several other people with echo issues, I thought I'd share what we've
> done (well actually what our chief R&D engineer, Brett Bourn has
> done...)
>
> First let me say that normal cheapy PC hardware couldn't be made to
>
Title: Message
http://www.voip-info.org/wiki-Asterisk+dimensioning
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
NocettiSent: 16 July 2004 14:43To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Astersik
with g729 and 120 acti
I'm new to asterisk, I have some questions regarding the hardware required. For a
complete E1 and something like 30 phones, what hardware would be required ? What is
the bottleneck ? Memory, processor speed ? Any experiences of how much scalability
does the asterisk have ?
Can Asterisk have "vi
After speaking with several people, and even participating in a forum of
several other people with echo issues, I thought I'd share what we've
done (well actually what our chief R&D engineer, Brett Bourn has
done...)
First let me say that normal cheapy PC hardware couldn't be made to
function with
Hello,
Who can help me I am trying to setup the sendmail so that I can
mail the voicemail’s to an internet SMTP mail server.
I know that I have to setup the sendmail.cf and configured a
relay to my normal SMTP server.
I am running RedHat 9 and my internet provider has a SMTP mail
s
Yes, this is working for me.
The remote side must send *** for this to work. If asterisk
does not detect the first *, then it prints the message you are seeing.
On Fri, 2004-07-16 at 07:18, Joseph wrote:
> Is anyone using Feature Group D with a digium 405P card
> and get the ANI spill to work (ca
Hello,
I’m very new with * and I would really appreciate some
help to implement a SIP to PSTN Gateway.
My current scenario includes an * box with a TE405P board. I
have a 1.5Mb connection to the outside world (using a router with firewall
capabilities) and channel banks that allow me t
http://bugs.digium.com/bug_view_page.php?bug_id=0002055
This patch adds the ability to send text and HTML messages as
voicemail notificiations.
Please test and respond to the bug tracker!
/O
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.d
On Jul 16, 2004, at 12:22 AM, Randy Bush wrote:
please be specific about dialplan hack and how you pointed to
it in the config. thanks!
Set Admin-Advanced->Line1->DialPlan->
([2-9]xx<:@gw0>|[3469]11|0|00|[2-9]xx|1xxx[2
-9]xxS0|.)
"[2-9]xx<:@gw0>" will send 211,311,411...911
http://bugs.digium.com/bug_view_page.php?bug_id=0002013
If you use the Czech language, please test this and add your
opinion, good or bad, to the bug tracker.
/O
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http://lists.digium.com/mailman/listinfo/as
http://bugs.digium.com/bug_view_page.php?bug_id=0001858
Constfilin writes:
"The attached patch allows dynamic configuration of asterisk queues.
Queue information is re-read from the configurable database in real time.
Additional Information Right now implemented only for postgres, no mysql"
Please
On Fri, Jul 16, 2004 at 10:49:57AM -0500, Chris Luke wrote:
[stuff snipped]
> http://www.btwholesale.com/content/binaries/service_and_support/pricing_information/carrier_price_list_browsable/b1_08.rtf
> Tells you how much geographic number portability costs with BT, though
> you may also need to
Chris Bond [EMAIL PROTECTED] wrote:
> What I want to know is why you can port mobile numbers from network to
> network but say you have a local std code and you wanted to use that with
> a VoIP provider in the UK - in most instances you cant. Some are
> available to offer STD codes in certain town
Mark Turner [EMAIL PROTECTED] wrote:
> Dameon D. Welch-Abernathy wrote:
> >
> > There are probably others.
> >
> Such as www.intervivo.net.
>
> p.s. I work there.
>
You might want to change the FAQ:
Q. Are you planning on offering non 0845 numbers?
A. The industry is looking into various altern
On Jul 16, 2004, at 7:03 AM, <[EMAIL PROTECTED]> wrote:
Yeah...that's believable. Mike, wake upthey made changes, the
broke
things. While they obviously "tried" to give everyone a month's
notice,
it just didn't work out that way. While the old config still works
(now), I find it difficult
[EMAIL PROTECTED] wrote (on Jul 16):
> On Fri, 16 Jul 2004, Chris Bond wrote:
> > What I want to know is why you can port mobile numbers from network to
> > network but say you have a local std code and you wanted to use that with a
> > VoIP provider in the UK - in most instances you cant. Some ar
Thanks for the quick reply. That should do it.
-Galt
On Fri, 16 Jul 2004 17:17:29 +0200, Thomas Niesel <[EMAIL PROTECTED]> wrote:
> Hallo John Galt
> On Fri, 16 Jul 2004 07:48:23 -0700 you wrote:
>
> > Could someone please point me to the proper url to register for a fwd
> > acount and get a fw
Go here: http://account.freeworlddialup.com/index_new.php?section_id=94
or just press the "GET FWD"-Button on the page...
At 16:48 16.07.2004, you wrote:
Could someone please point me to the proper url to register for a fwd
acount and get a fwd number. I couldn't find it at
www.freeworlddialup.com
Can anyone point me in the right direction to learn how to make the 3COM
3102 phone work with Asterisk. I have looked everywhere that I can think
of and I am not able to turn anything up. Do I need to have a login to
3COM's site to obtain a SIP image?
Any help would be GREATLY appreciated.
Than
On Fri, Jul 16, 2004 at 03:50:40PM +0100, Chris Bond wrote:
> What I want to know is why you can port mobile numbers from network to
> network but say you have a local std code and you wanted to use that with a
> VoIP provider in the UK - in most instances you cant. Some are available to
> offer
I have a macro that handles my outbound dialing... First it does an enum
lookup, and if that fails it hands the call off to voicepulse for connection.
The problem is... EnumLookup() requires that international numbers *NOT* be
prefixed with 011, however VoicePulse does... Can someone provid
On Fri, 16 Jul 2004, Chris Bond wrote:
> What I want to know is why you can port mobile numbers from network to
> network but say you have a local std code and you wanted to use that with a
> VoIP provider in the UK - in most instances you cant. Some are available to
> offer STD codes in certai
Holger Schurig wrote:
I keep replying to myself quite often.
As it turned out, this is a problem with incrementing CSEQ on the
Grandstream. I don't have the clue if the SIP specification says that you
have to increment it, but the GS sometimes sends a different SIP message
with the same CSEQ. He
Maybe another nice feature might be for the other end of this problem...
Menu options while you are on hold to change the crappy music or mute it.
Also an option to punch in a callback number and have the company ring
your phone when it is your turn to have the call answered.
- Jason
> I have no
I'm getting down to the last of my * issues ...
After calling a bank, or cc processing center; you have to enter your
social security number, or the cc number - followed by the # key.
The lovely * voice responds "transfering" I'm sorry that was an invalade
selection. Sometimes the IVR on the oth
I keep replying to myself quite often.
As it turned out, this is a problem with incrementing CSEQ on the
Grandstream. I don't have the clue if the SIP specification says that you
have to increment it, but the GS sometimes sends a different SIP message
with the same CSEQ. Here's an example, only
What I want to know is why you can port mobile numbers from network to
network but say you have a local std code and you wanted to use that with a
VoIP provider in the UK - in most instances you cant. Some are available to
offer STD codes in certain towns but not all.
Surely someone in the UK nee
Could someone please point me to the proper url to register for a fwd
acount and get a fwd number. I couldn't find it at
www.freeworlddialup.com or fwd.pulver.com
Thanks,
-Galt
--
"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Dameon D. Welch-Abernathy wrote:
There are probably others.
Such as www.intervivo.net.
Cheers,
Mark.
p.s. I work there.
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To UNSUBSCRIBE or update opti
On July 16, 2004 09:02 am, Jan Goericke wrote:
> hello,
>
> yesterday the uniden uip200 phone was recommended to someone. i am looking
> for an alternative to grandstream bt-100 because i can not do a supervised
> tranfer with it. here my questions:
>
> 1) does the uip200 support supervised transfe
Hi,
I have to develop a phone application using asterisk's
chan_oss.
When the phone is idle, i.e. the last command was a hangup,
one hears a "toot, toot, toot, ..."
But unforuntaly its use is in Germany, where one expects
a continous "too ..."
before dialing.
Is ther
As far as I can tell I have this resolved. For those of you that may
encounter this error I send this out to help.
I have determined that the problem lay with my mail setup. I was attaching
voicemail messages to email alerts. The problem was my server decided to
take the backup MX instead of the
On Fri, 2004-07-16 at 02:03, Shawn Lawrence wrote:
> Is there a way to flag a bad or noisy PRI channel in Asterisk so it will
> be skipped over?
How can a DIGITAL channel be noisy?
your big problem will come from the fact that your provider will still
attempt to ship calls down the channel you d
Is anyone using Feature Group D with a digium 405P card
and get the ANI spill to work (callerid)?
I can get the dnis to work fine.
But not the caller id.
Here is the error I get:
chan_zap.c:4628 ss_thread: Got a non-Feature Group D input on
channel 3. Assuming E&M Wink instead
zapata.conf: ***
On 16/07/2004, at 11:57 PM, Dennis Cartier wrote:
Sorry if this has been asked before, but does anyone have any pointers
on getting the Message Waiting Indicator working on a GS ATA-286?
I have tried both settings on the 286's web page for sending a
SUBSCRIBE to the SIP server. I was expecting the
I have a recent version installed. I am having problems with hangup
detection on my zap channels.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
> -Original Message-
> From: [
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> Randy Bush
> > Sent: Thursday, July 15, 2004 3:35 PM
> > To: [EMAIL PROTECTED]
> > Cc: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Re: VoicePulse changes
> >
> > the message arrived here so
>The fact how your Server/LAN is connected to the 'outside' world has
>little to do with the way Asterisk connects to the PST, imho.
>
>What kind of voice 'streams' pass on this fiber? How is this fiber
>connected?
The fiber is connected through a hag, in which voice stream is digitized
in IP tra
Sorry if this has been asked before, but does anyone have any pointers
on getting the Message Waiting Indicator working on a GS ATA-286?
I have tried both settings on the 286's web page for sending a
SUBSCRIBE to the SIP server. I was expecting the LED on the 286 to
flash when there are messages w
Hi everybody,
I have a problem using zaphfc. When I start asterisk after 8-10 seconds I
get the message "Primary D-Channel on span 1 down" and my isdn modem stops
to work.
If I place or receive a call before this message all works really fine (even
if the call is very long), but when I hang up, aft
Hiya,
I've been seeing this lately with our Asterisk PBX as well. With
Asterisk CVS HEAD 05-21, the ringing was normal. After our last
upgrade, CVS HEAD 07-03, the first ring is almost always one continuous
ring that lasts about 2 to 3 seconds.
We're noticing other problems too, such as hangup d
Hello, I want to
know what kind of equipment I need to handle 120 simultaneous calls with a
Digium 4E1 card... and using 120 G.729 licences some
help?
thanks
Sebastian.
Jan Goericke wrote:
Think nobody would like to use call parking via #700 etc if he is "spoilt"
by the features of an old Siemens Hicom and my fellows are spoilt. Or is
there a more elegant way to make a BT100 doing this feature. I read nearly
all threads in archive concerning transfer and sip. Se
Ok, this seems to be a bug in channel_sip.
When I do put all debug message AND I dial 0-0914115, then I only see:
$ grep DTMF /var/log/asterisk/messages
Jul 16 13:13:19 VERBOSE[6150]: Receiving DTMF!
Jul 16 13:13:19 VERBOSE[6150]: * DTMF received: '9'
Jul 16 13:13:20 VERBOSE[6150
What phone do you have?
On Fri, 16 Jul 2004 11:59:39 +0500, atif <[EMAIL PROTECTED]> wrote:
> I am configuring a sip-phone, receing calls, excellent voice quality. but it does
> not place calls, please, can some one sort out.
>
> here is my debug output, and below that is sip-debug,
>
> Jul 16
On Fri, 16 Jul 2004, Jean-Yves Avenard wrote:
> With the BT100 you can always use call parking to achieve something
> similar, sure it's not the most elegant way..
>
Think nobody would like to use call parking via #700 etc if he is "spoilt"
by the features of an old Siemens Hicom and my fel
Chris Smales - Magenta Solutions wrote:
I've configured voicemail.conf to delete voice mails after they are
emailed. The email work ok but the message don't delete. The config is
as follows:
[default]
3000 >= ,CS,[EMAIL PROTECTED],,delete=yes
Thanks,
Chris
It should be
3000 => ,CS,[EMAIL P
Sorry, my mistake, I thought you'd said you had assigned priorities, let me
go back and search for your original posting
Steve
-Original Message-
From: Robert Jackson [mailto:[EMAIL PROTECTED]
Sent: 16 July 2004 14:10
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ACD Issues
T
On Fri, 16 Jul 2004 11:56:32 +0200, Holger Schurig
<[EMAIL PROTECTED]> wrote:
> In newer Grandstream Firmware you can have different dialtones for
> different incoming caller id's. Asterisk can set the callerid to
> anything, based on the incoming line, so you can hear by the ring from
> which line
That would certainly make sense, but I am not sure how to set an Agent's
priority. The only information that I have been able to find is setting
a QUEUE_PRIO value when queuing the calls (New as of July 2004).
Thanks,
Robert Jackson
> -Original Message-
> From: Steve Hanselman [mailto:[
I've configured voicemail.conf to delete voice mails after they are
emailed. The email work ok but the message don't delete. The config is
as follows:
[default]
3000 >= ,CS,[EMAIL PROTECTED],,delete=yes
Thanks,
Chris
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