Displaying 20 results from an estimated 4000 matches similar to: "Meetme codec translation and callerID library."
2005 Mar 21
2
CallerID Name with IAX Providers
I am pretty sure that there are no IAX providers that offer CallerID name
but wanted to double check with the list in case something has changed
recently. Is anyone aware of an IAX provider that offers incoming CallerID
name?
Is there a technical limitation within IAX which is preventing IAX providers
from offering CallerID Name? Why is no one offering this?
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2005 Mar 17
2
Getting caller-name - cid_rewrite 1.0.0
Hi folks, I think my little agi script is ready for the big one-oh-oh.
Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This
agi-script does the following:
- Standardize incoming caller-id numbers to adhere to US dialing code;
NANPA numbers are reformatted to 1+10, international numbers become
011<country-code><number> (this is customizable with a little PHP
knowledge).
2008 Feb 21
1
cid_rewrite.php -- Caller ID Name lookup
For those folks who are still using it --
I updated the cid_rewrite script. I noticed that two of the providers
were "iffy" and one had changed format a little while ago. It's working
again.
http://muware.com/asterisk has the latest (1.2.0)
Enjoy,
-- JM
2005 Mar 24
3
Outlook contacts -> Asterisk database (LookupCIDName)
Is it possible in any way to use an Outlook contacts database as the
source for the internal Asterisk database that is used for callerid
lookups?
Thanks!
2005 Apr 09
3
CallerID name lookup AGI script
Hi all,
My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote
an AGI script that does the following:
1) If it's a toll free number (800|888|877|866), set the CallerID name to
"TollFree Caller"
2) Use curl to look up the number in Google phonebook
3) If a business listing, set the CallerID name to business name, as is.
4) If it's a residential
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2010 Oct 11
1
Unable to find a codec translation path from ulaw|h261 to slin
I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've
encountered a problem playing back a .wav file to an Ekiga client:
My dialplan looks like:
exten => 730,1,answer
exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
exten => 730,n,hangup
Sovereign.wav is a .wav file that plays nicely on my 1.4 server.
Here is what the console displays:
2015 Feb 17
4
Callfile problem - Unable to find codec translation path from (nothing)
Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using:
[outbound-swift]
exten => _[a-zA-Z].,1,Answer
exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure)
;exten => _[a-zA-Z].,1,Swift("${EXTEN}")
exten => _[a-zA-Z].,n,Goto(1)
[mis-phone]
exten =>
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
2007 May 04
2
Asterisk Codec Translation Table
Hello list,
I have always though codec translation table is dircetly connected to system speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw
2011 Sep 30
1
Core show translation > 4000ms
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the Lenny server
Translation times between formats (in microseconds) for one
second of data
2010 Sep 06
1
MeetMe errorhandling
Hi Group,
i have a MeetMe Question.
I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist Asterisk play (conf-invalid.slin)
If i use "MeetMe(${room},Ms)" (value from DTMF Read) and the Conference Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the Call.
there is a solution for the kind my problem?
Thanx and
2013 Jun 10
3
no silk translation ?
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c
[Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032
dial_exec_full: Had to drop call because I couldn't make
2005 Mar 16
3
NuFone and CallerID
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows "Toll Free
Call" and will not give me the calling party's caller ID info.
Is this just something I have to live with using NuFOne, or did I miss
some type of config in * that will grab the callerID other than the
inbound 866 number...?
2006 Mar 09
1
Mutable models based on dbfunctions or stored procs?
Is there any way that a model can be based on a stored procedure or
function that so that it can return a set of data based on a parameter
provided in either the controller, or s a variable in the model? That
way you could:
1) Set the variable
2) Call find_all
3) Get a mutable set of results that takes better advantage of some
of the advanced database features available?
2008 Jan 15
3
Meetme recording
Hello,
Is there a way to change the format from the default?
'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format
${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. -
requires chan_zap.so
Many thanks
********************************************************************
This email and any attachments
2015 Sep 30
3
Change Asterisk MulticastRTP codec
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data
2007 Sep 29
3
meetme conference using g729?
Hi,
is there a way to use g729 in meetme?
Thanks!
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2014 Jun 04
1
Renegotiate SIP audio codec after call is up
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2012 Nov 21
1
core show translation - difference in Asterisk Versions
Hello All,
I was wondering if somebody could elaborate the change in
translation of codecs specifically the amount of time increased in Asterisk
11. For example
*Asterisk 11*
* **alaw **speex *
*gsm **15000 **15000 *
*ulaw 9150 15000*
* *
*Asterisk 1.6.x*
* **alaw **speex *
*gsm **2 12002 *
*ulaw 1 12002*
I did recalculate the