Does anybody know what steps are required to configure a Sipura 1001 ATA to use [EMAIL PROTECTED]. Thanks,Olu
Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries for just 2¢/min with Yahoo! Messenger with Voice.___
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How many times do u need to repeat it?. You could change this info via
web list manager.
I think you need to read how to do that before sending 20 emails with
same subject.
[EMAIL PROTECTED] escribió:
Please change the email address
of [EMAIL PROTECTED] to [EMAIL PROTECTED]
Thanks
I also have an 8700g. Have you managed to figure out how to play .wav voicemails?On 5/13/06, Kerry Garrison
[EMAIL PROTECTED] wrote:Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without anyproblem. My problem
For voicemail you can fix this easily.
Simply nfs export the voicemail spool on the voicemailbox
and mount them on the other servers.
That way all systems will know which voicemail boxes have
new voicemail.
good luck
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key:
I think it's most likely that it's a mail loop caused by a brain dead 'change
of address' script.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alberto Sagredo
Sent: Saturday, 13 May 2006 17:00
To: asterisk-users@lists.digium.com
I use res_snmp.so with asterisk do you provide mib
--- David Yat Sin [EMAIL PROTECTED] a écrit :
Hi Harry,
The Sangoma Card when used for TDM Voice will work
under zaptel, so you
would need to perform the SNMP through Asterisk.
Regards,
David Yat Sin
Sangoma Technologies
(905) 474 1990
Hello,
Can we park a call or put on hold a caller in a queue
?
I have sip polycom phone but when i press hold key or
#800 i can't neither park call nor hold this call .
Is it possible ?
Harry
Can you explain me why I 'm obnoxious cretin ?
I 've been asking for monitoring the digium cards via
snmp.
What's the problem ?
I post to asterisk-users and asterisk-dev to get
informations why some people of these list insult me ?
What are yours problems ?
Plesae to send me back my cretin
Can you explain me why I 'm obnoxious cretin ?
I 've been asking for monitoring the digium cards via
snmp.
What's the problem ?
I post to asterisk-users and asterisk-dev to get
informations why some people of these list insult me ?
What are yours problems ?
Plesae to send me back my cretin
Tzafrir,
cross-posting to asterisk-users and to asterisk-dev
is not a good idea.
You should know that by now, as you have been told
that numerous time.
I cross-posting because nobody answer !
I think some people are able to answer this question ?
Please, don't tell me people here are not
Tzafrir,
cross-posting to asterisk-users and to asterisk-dev
is not a good idea.
You should know that by now, as you have been told
that numerous time.
I cross-posting because nobody answer !
I think some people are able to answer this question ?
Please, don't tell me people here are not
Earl Terwilliger wrote:
On Friday 12 May 2006 18:30, Rich Adamson wrote:
Carlos Alperin wrote:
Rich,
Check what is the content of /lib/modules/2.6.15-1.2054_FC5/build?
I see:
[EMAIL PROTECTED] build]# ls
arch crypto initMAINTAINERSREADME usr
block
On Saturday 13 May 2006 09:02, Rich Adamson wrote:
Earl Terwilliger wrote:
On Friday 12 May 2006 18:30, Rich Adamson wrote:
Carlos Alperin wrote:
Rich,
Check what is the content of /lib/modules/2.6.15-1.2054_FC5/build?
I see:
[EMAIL PROTECTED] build]# ls
arch crypto
Hello list,
I'd like to share something u all , so that i could understand whats
going on into my Asterisk box.
i have a setup like this
client(ip phone) -ip network--- [Asterisk]ip network
---[Service provider]
i have configured A2biling in my Asterisk box. so when client
Gareth Blades wrote:
I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.
Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being
Darrick Hartman wrote:
Tom Vile wrote:
Same problem with audio quality. Got rid of them. Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.
Which audio codecs were you
What makes you say that?
Well for one thing even the Sangoma cards aren't listed there, and
theres many more from Germany and China not listed.
To the original poster, as far as I know (my info is quite outdated)
brooktrout didnt have any
open source linux drivers and they haven't personally
Unless I did something wrong when we did that originally, if the voicemail
server's NFS share dropped out, the main call servers froze while trying
to access the voicemail share.
On Sat, 13 May 2006, Michiel van Baak wrote:
For voicemail you can fix this easily.
Simply nfs export the
On 11:29, Sat 13 May 06, Aaron Daniel wrote:
Unless I did something wrong when we did that originally, if the voicemail
server's NFS share dropped out, the main call servers froze while trying
to access the voicemail share.
Hi,
Possible, I dont know.
I always make sure my nfs servers are
On Sat, 13 May 2006, Michiel van Baak wrote:
On 11:29, Sat 13 May 06, Aaron Daniel wrote:
Unless I did something wrong when we did that originally, if the voicemail
server's NFS share dropped out, the main call servers froze while trying
to access the voicemail share.
Hi,
Possible, I dont
Why not installing Asterisk and zaptel from the atrpms repo. It works
great even if you need to recompile...
Earl Terwilliger a écrit :
On Saturday 13 May 2006 09:02, Rich Adamson wrote:
Earl Terwilliger wrote:
On Friday 12 May 2006 18:30, Rich Adamson wrote:
Carlos Alperin wrote:
Rich,
Hi all,
I have 2 of these with me and from day 1 im facing issues with the
background noice i have done all sorts of testing but all in vain. the
changinf of poer supply reduced noise to certain extend but still its
very clearly audible.
A good fix would be really helpful to me
Thanks
On
Install iptraf, that will allow you to check incoming and outgoing traffic
(or trafshow what do that on /host basis, but not so detailed info)
If you choose ulaw, that should take about 90kbps fullduplex traffic.
I'd like to share something u all , so that i could understand whats
going on
Are dependencies resolved when using yum to install form the ATrpms
http://www.voip-info.org/wiki/view/ATrpms
--Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phi Fou
Sent: Saturday, May 13, 2006 11:55 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Unless reinviting works, wouldn't that add up to what he's experiencing ?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them would give 800k or so.
What I can't understand is: if only g723 is allowed, and Asterisk only
allows it as passthrough, how's the
The fix I found was to not use them. They are not ready yet for production use.
On 5/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi all,
I have 2 of these with me and from day 1 im facing issues with the
background noice i have done all sorts of testing but all in vain. the
changinf of
AR Tarzi wrote:
Unless reinviting works, wouldn't that add up to what he's experiencing ?
client - asterisk - service provider.. makes that 180k each connection
so 4 of them would give 800k or so.
What I can't understand is: if only g723 is allowed, and Asterisk only
allows it as passthrough,
I did not get this back from the list so I'm not sure if
thishit the list last week or not so I'm sending it again. Sorry if this
is a duplicate post!
---
Has anyone had problems with a Cisco 7970 running sip image
Hall, Eric M. wrote:
I did not get this back from the list so I'm not sure if this hit the
list last week or not so I'm sending it again. Sorry if this is a
duplicate post!
---
Has anyone had problems with a
On 11:29, Sat 13 May 06, Aaron Daniel wrote:
Unless I did something wrong when we did that originally, if the
voicemail
server's NFS share dropped out, the main call servers froze while
trying
to access the voicemail share.
Hi,
Possible, I dont know.
I always make sure my nfs
I don't see that anywhere. Here is my zapata.conf This is only happing
on my 7970 all other phone are working without trouble.
[channels]
context=pri
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
Must be able to pass Caller ID number. Email me with your terms.
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AlexOn 5/13/06, Bob's Leaky News Service [EMAIL PROTECTED] wrote:
Must be able to pass Caller ID number. Email
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