Oliver,
SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway
I get the following error:
Unable to find a codec translation path from ilbc to ulaw
Does your phone support ilbc as a codec ?
Is the codec_ilbc loaded on the * box ?
Usually you get this kind of error when the codec is not
On 4/27/07, Steve Murphy [EMAIL PROTECTED] wrote:
I'm the guilty party. I've been trying to fix several CDR bugs,
involving stuff like missing times, missing changes in state (like
NO_ANSWER when the call was ANSWERED), etc.
A-HA! Don't get me wrong, I am not opposed to progress as there have
Steve Finkelstein wrote on 4/28/07 12:21 AM:
my musiconhold.conf:
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
and finally in my extensions.conf:
asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf
exten = 100,1,MusicOnHold(30)
exten = 100,2,Hangup
When I dial
On Sat, Apr 28, 2007 at 01:47:15PM +1200, CSB wrote:
On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:
[snip]
As suggested earlier I replaced this with:
/etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1
[snip]
dmesg
Zapata Telephony
Ok this is my first post and I will try to keep it short.
I have searched everywhere and haven't found an answer to my question
I have two Trixbox servers that are connected over the Internet via an IAX2
connection. We are experiencing very poor sound quality. I have tried many
different
Hi Matt,
you didn't mention what type/bw of each site Internet connection, i suggest
that you try to split the scenario into smaller pieces:
- run long term pings between the server while you make a call and check for
packet loss.
- make internal calls between extensions on the same branch and
Yep it's possible though why not just use a handset with a microbrowser
that states on the display logged in out or?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Try SIP if at all possible. I have had mixed results with IAX that SIP
made go away. If you try SIP, you can at least rule out IAX as the
cause.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yossi Ben
Do you guys have an ISO install CD yet?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, April 27, 2007 2:00 PM
To: Asterisk Users Mailing List -
Not yet, but it is something we are working on. There are a few people
that have made some special-hardware ISOs for VICIDIAL but they are by
no means universal, more for quick install on specific high-end
servers.
Right now we are just concentrating on making VICIDIAL as solid and
Hi Dave!
Thank you very much for replying!
what gateway provider are you referring to?doesn't your sip phone
webcalldirect (it does not seam to support iLBC directly)
connect directly to * as your diagram indicated?
Yes, my sipphone ist connected directly to * and also the gateway
Sorry if I´m not clear.
I´m using zap channels. I need to limit the number of calls that dial one
extension. No more than 3 calls using an IVR service (eagi) at the same
time.
May be It can be resolve using GROUP() and GROUP_COUNT()
exten = 99,1,Set(GROUP(99) = G99)
exten =
Hi Stelios!
Thank you very much for you reply!
Does your phone support ilbc as a codec ?
Definately. By using to phones and forcing them to use iLBC I can make
calls from one phone to the other. The gateway provider does not support
iLBC and so * has to do the conversion to ulaw. I've also put
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:
SIP-phone --iLBC-- Asterisk ---ulaw PSTN-Gateway
I get the following error:
Unable to
Just to follow up on my previous comment,
/usr/share/doc/asterisk/copyright contains the following on my Debian
system:
* The iLBC codec library code has been removed from the Debian asterisk
package as it does not conform with the DFSG.
James
___
If you are going to have clusters of phones like a cubicle setup, you
could buy one of the Linksys routers like the WRT54G and setup WDS.
Then plug four phones into it.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-Original Message-
From: [EMAIL PROTECTED]
Nabeel Jafferali wrote:
You can look for headsets made for Motorola cell phones. Also,
Plantronics has some compatible models - I can dig up part numbers if
you're interested.
Yes, please - Plantronics is in my regular suppliers catalog, but still
only with 3.5mm jacks. If you've got
Per Jessen wrote:
Nabeel Jafferali wrote:
You can look for headsets made for Motorola cell phones. Also,
Plantronics has some compatible models - I can dig up part numbers if
you're interested.
Yes, please - Plantronics is in my regular suppliers catalog, but
still only with 3.5mm
On 4/27/07, Per Jessen [EMAIL PROTECTED] wrote:
Try your local mobile phone supplier. I used a headset that came with
one of my cell phones, and it worked great w/ my SPA-941.
Not a bad idea - which make was this for? None of my phones (Ericsson,
Nokia) have a 2.5mm socket, they're all
Greetings list,
Thanks to all who replied to my thread a few days ago SIP devices with packet
loss tolerance. One of the suggestions that came out of that thread was to
replace routers at users' premises with ones that support QoS.
I've used m0n0wall's QoS in the past with reasonable success,
Hi Matt -
I have two Trixbox servers that are connected over the Internet via an IAX2
connection. We are experiencing very poor sound quality. I have tried many
different codecs gsm, ilbc, g729, g711 and all seem to have the same
problem. (All though g729 seems to work the best but still
Hi Joseph -
Thanks, I think you are on the right track.
When no Sip adapters were connected to asterisk it took me over one
minute from the time I typed reload to the time I've seen anything on
the screen.
When, I connected the all the sip devices and eliminated some entries in
sip.conf and
Interesting, that works David.
I got the example directly out of the published VoIP Hacks book and
followed instructions step by step.
Either way, thanks much. :-)
- sf
Dave Miller wrote:
Steve Finkelstein wrote on 4/28/07 12:21 AM:
my musiconhold.conf:
[default]
mode=quietmp3
Hi Forest -
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one server. Is the
best way to do this just through a dial app?
Can anyone think of draw backs to this? One I can think of
Noah Miller wrote:
Hi Joseph -
Thanks, I think you are on the right track.
When no Sip adapters were connected to asterisk it took me over one
minute from the time I typed reload to the time I've seen anything on
the screen.
When, I connected the all the sip devices and eliminated some entries
Noah Miller wrote:
Hi Forest -
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one server. Is the
best way to do this just through a dial app?
Can anyone think of draw backs to this?
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:
Thanks to all who replied to my thread a few days ago SIP devices with
packet loss tolerance. One of the suggestions that came out of that thread
was to replace routers at users' premises with ones that support QoS.
Sangoma S518
Hello,
Installed Trixbox with a digium card and it is taking 2 rings for it to pick
up. Any suggestions how to have the system pickup immediately?
Thanks,
Neal
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hi,
You need to post this on the trixbox forums.but as a fellow trixbox
user I'll give you the answer.
Turn off fax detection.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
Hi,
Write down your problem clearly.
Thanks
[EMAIL PROTECTED] wrote: Hello,
Installed Trixbox with a digium card and it is taking 2 rings for it to pick
up. Any suggestions how to have the system pickup immediately?
Thanks,
Neal
___
Andrew wrote:
On Saturday 28 April 2007 11:22 am, Chris Bagnall wrote:
Thanks to all who replied to my thread a few days ago SIP devices
with
packet loss tolerance. One of the suggestions that came out of that
thread
was to replace routers at users' premises with ones that support QoS.
One thing I would suggest trying, just from experience, Is the load on the
boxes.
Unless you have REALLY poor latency, calls do not cut out for just 3-4, but
they very well
could if the box load is getting very high.
Keep a look at top (though not reliable) and the call count when the
Hi Steve -
Can you elaborate on this, I changed to storing the voicemail via ODBC
on MySQL. Each server had it's own local storage, and then MySQL
replicated the databases between the sites. This setup was terribly
finicky and unstable. It was much worse than the NFS mount. I quickly
gave
I snipped all of the previous data, as I'm trying to boil down
this problem to its essence...
I turned off the firewall for a few seconds, and still got no
audio. For those that will be suspicious, the commands were:
shorewall stop
shorewall clear
tested connection, no audio
shorewall
Sorry bringing it up again
Meanwhile switched to asterisk 1.4.3 on fbsd-6.2 but still
no luck getting my 7970G to run via skinny...
It registers fine with *:
Adding button: 9, 1
Device capability set to '268'
asterisk*CLI skinny show devices
Name DeviceId IP
for it to pick
up. Any suggestions how to have the system pickup immediately?
Thanks,
Neal
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A little with skinny debug set to on shows during register:
Device SEP00175A872053 is attempting to register
Requesting capabilities
Buttontemplate requested
Adding button: 9, 1
Sending 30006 template to cisco
Received SoftKey Template Request
Received SoftKeySetReq
RECEIVED UNKNOWN MESSAGE
Hello All,
We have been doing Asterisk and CME implementations recently but we
almost always exlusively bring in analog lines and or PRI for PSTN
access to our systems. I have known about providers providing SIP
based lines and SIP trunks to end users for PSTN access. I am curious
Apparently while it was a simple question it was either not a simple answer
or no one found it interesting..
I guess i'll give an example:
Here is a hard coded queue.conf queue configuration that i would like to put
into real time config
[CAIS]
musicclass = default
announce = queue-markq
Hi,
I'm wondering what the best option to obtain a high availability
asterisk server is.
I currently use a TE410P (4 x E1) card.
I'm thinking of 2 different solutions:
- 2 servers configured with Heartbeat + DRBD (drbd mainly for
voicemail) and the E1 span plugged to the 2 servers (with a
pickup immediately?
Thanks,
Neal
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___
--Bandwidth and Colocation provided
Okay i think that real time does work as expected... my issue was actually
poor documentation... it seems that everywhere you look call_limit is the
configuration option for sip.conf however the REAL option is call-limit not
call_limit... the underscore is listed in the initial bug report
Hi Laurent -
Is it technically good to connect an E1 span to 2 cards at the same
time (with only one accepting the calls).
Since it is possible with BRI cards, i'm wondering if it could be done
with PRI.
Nope. You can use a device like the Redfone fonebridge to convert the
PRI to TDMoE.
On Sunday 29 April 2007 01:06, Noah Miller wrote:
I've heard of a device that acts as a failover for a PRI line so you
can plug a PRI into two different devices and have the PRI failover if
one device fails. Unfortunately nothing like this is commercially
available today.
Sounds like the
On Sat, 2007-04-28 at 23:22 +0200, Laurent CARON wrote:
Hi,
I'm wondering what the best option to obtain a high availability
asterisk server is.
I currently use a TE410P (4 x E1) card.
I'm thinking of 2 different solutions:
- 2 servers configured with Heartbeat + DRBD (drbd mainly
How do you handle transfering vmail from one user to another when they're on
separate servers?
I'm using the single vmail server, mounted NFS partition for this right now.
I'd love to be able to have them standalone so they're survivable when the
WAN collapses, but I haven't figured out transfer.
oliver,
ugh, it is too obvious... why did it take me so long to figure it
out...
both phones have to have to negotiate the same codec for audio... as
far as I know, * is supposed to do automatic translation and your
gateway should be doing translations only on the below codecs. I
haven't had
Has no one else experienced the problem I mentioned a few days ago with
app_dictate? Or maybe no one is using that app. We're having a problem
with choppy audio and failure of the accelerated playback feature which
seems to be consistent on a couple of installs, failing with some SIP
carriers
On Sat, Apr 28, 2007 at 01:47:15PM +1200, CSB wrote:
On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:
[snip]
As suggested earlier I replaced this with:
/etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND honormode=1 boostringer=1
fastringer=1
[snip]
dmesg
Zapata Telephony
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