--- Markus Schuster [EMAIL PROTECTED] wrote:
Could you please post some details (or even better:
write them in some sort
of Wiki) on the configuration you did on the Nokia?
I'm thinking about buying a Nokia E60 but after a
short web search there
seem to be some problems about the correct
Good morning,
Why still use mpg123?
Start using format_mp3 from asterisk-addons and your * will
play mp3 by itself...
good point - did that, and everything's working again. Thanks!
Marc Rohlfing
___
--Bandwidth and Colocation provided by
No , actually I am using Asterisk-1.2.9.1
I will try the q option though
Thanks and regards,
Amna
On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I assume you are using 1.0.x.Add the q option to the Meetmeextension.1.0.x has a known issue where enter/exit sounds cause
increasing
Hi!
I am using Asterisk-1.2.9.1
Zaptel 1.2.6
And my system has Linux Kernal 2.4
Best Regards,
Amna
On 6/14/06, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED]
,amna saleem [EMAIL PROTECTED] wrote: Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
wrote:
On Jun 13, 2006, at 7:52 PM, Josu頃onti wrote:
?? Doug,
If you it will not have hardware and if ztdummy will not have
installed
its moh will not function correctly
I believe this is no
Hey,
we're running an asterisk system connected to another telco system using
qsig. I'm currently trying to
use ChanIsAvailable to get the current phone status out of the foreign
system.
ChanIsAvailable always return 0 - UNKNOWN. The Qsig protocoll itself
supports the feature to
query the
I believe this is no longer be true with the new Native music on
hold...
Ah. I suspected as much, when my home server wasn't running any zaptel
for a bit I found that MoH was working fine, even though last time I
tried without ztdummy a while ago it was severely broken.
(The reason I
It seems that Microsoft RTC has some
problems with originated calls from Asterisk. If I execute Manager API
originate application, with SIP channel as parameter, the Microsoft RTC
softphone will start to ring after a couple of seconds delay, but nothing more
happens after when I answer
festival.i386 1.4.2-25
Too older.
And does anyone know if I can add Festival voices to Flite (slightly
off-topic... sorry...)?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Tuesday, June 13, 2006 5:56 PM
To: 'Asterisk
Santosh Rao a écrit :
asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER..
may be if someone good
Hello
How do I get Asterisk to receive DTMF from our Snom phones, when I use G.729?
Regards
Jon
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.394 / Virus Database: 268.8.4/363 - Release Date: 13-06-2006
___
Hi Marty,
thank you for your suggestion, but... just done and nothing happens.
The perfect RX gain for me is 6.0; if I change the TX nothing really happens,
but if I move it to a value less than 0.0, less than that I cannot ear
anything... strange but true.
As I said, on internal calls everything
Have you only one BN-Card? or more?
i have two cards, had compareable problems.
PCM was the magic word ...
from my misdn-init.conf:
card=1,0x8,pcm_slave,ignore_pcm_frameclock //important!
option=9,master_clock // 9
for port 9
pcm=1,1
Hi,
What is your setup? By MS
RTC do you mean Office Communicator?
If you are using MS OC,
do you use SER in between (to convert SIP UDP2TCP)? Please share some more
details J
Cheers,
Ohad
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Sent:
Hi,
Check if reinvites are
enabled, and that you dont use any parameter in the dial command that
forces asterisk to stay in the loop.
Ohad
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Baum
Sent: Wednesday, June 14, 2006
5:00 AM
To: Asterisk Users
I was just about to suggest the Powersense module that Cory mentioned. And no, the G models do not support 802.3af.
Cory, there was some discussion about just doing the cable only works on dumb poe injectors, not the ones that only send power if requested. I was under the impression the Linksys
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for
256k upstream you should be able to handle 8 calls but this is in ideal
conditions.
If you were to use IAX and enable trunking then you would use 30kbps for
the 1st call and 10kbps for each additional call.
See
Hi Friends,I am implementing Asterisk PBX in our office with 180 extensions. In our office, we will make 3 calls to USA daily. We have 1 MBPS bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts.1) How much bandwidth should we allocate for making VOIP calls? What can be the
Crazy Boy wrote:
Hi Friends,
I am implementing Asterisk PBX in our office with 180 extensions. In
our office, we will make 3 calls to USA daily. We have 1 MBPS
bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts.
1) How much bandwidth should we allocate for making VOIP
On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote:
Have you only one BN-Card? or more?
I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank.
i have two cards, had compareable problems.
PCM was the magic word ...
from my misdn-init.conf:
Crazy Boy wrote:
Hi Friends,
I am implementing Asterisk PBX in our office with 180 extensions. In
our office, we will make 3 calls to USA daily. We have 1 MBPS
bandwidth from ISP and 100 MBPS bandwidth in our LAN. I have two doubts.
1) How much bandwidth should we allocate for making VOIP
Hi,I have to build Asterisk server for about 30 user (30 concurrent calls). I decided to buy this box:-- motherboard Intel E7210 + Hence Rapids-- processor P4 3.0 GHz-- RAM 2x512 MB DDR ECC-- network interface Intel 82541 GI
Is this configuration enough to handle 30 users at the same time. I am
Hi Koen Van Impe
Thanks for the meridian config and asterisk. I will
defenitly try them
And let every one know.
Just a few words and correct me if I am wrong
There are two things
1
E1 : the 32 channels once both the equipment
see each other and the ccs/hdb3
Its overkill, go get some more employees :)
So yes, its just fine and there's room for expansion.
Zoa
Andrew Nowrot wrote:
Hi,
I have to build Asterisk server for about 30 user (30 concurrent
calls). I decided to buy this box:
-- motherboard Intel E7210 + Hence Rapids
-- processor P4 3.0
Andrew Nowrot wrote:
Hi,
I have to build Asterisk server for about 30 user (30 concurrent
calls). I decided to buy this box:
-- motherboard Intel E7210 + Hence Rapids
-- processor P4 3.0 GHz
-- RAM 2x512 MB DDR ECC
-- network interface Intel 82541 GI
Is this configuration enough to handle
Nope, it's just the
Microsoft RTC Core 1.3 library ... more or less a single DLL J. And I'm almost sure
there is no SER in between should there be one? It's pretty much a
straightforward thing I have a few SIP clients defined in my sip.conf,
like this:
[general]
context=default
Hi there sir
Thanks for ur
suggestion but the problem with us is that we are running the whole distributed
call center in three different cities of pakistan.
So we can not take risk on that behalf we
just want that our expansion need to be fulfilled by expanding throght asterisk
Hi,
I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu
6.06 - by downloading the v3 driver package from Melware and compiling
everything. Yet, after activating the necessary modules (divas and
divadidd) and interactively configuring the card
(/usr/lib/divas/Config), starting
I've got a strange situation with musiconhold.
It works if I dial my extension 6000:
From extensions.conf:
exten = 6000,1,Answer
exten = 6000,2,MusicOnHold()
Debug output if I call 6000:
-- Executing Answer(SIP/gs1-b6ee, ) in new stack
-- Executing MusicOnHold(SIP/gs1-b6ee, ) in new
On Wed, 14 Jun 2006, Marc Rohlfing wrote:
Hi,
I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu
6.06 - by downloading the v3 driver package from Melware and compiling
everything. Yet, after activating the necessary modules (divas and
divadidd) and interactively
Hi
I
can use now DISA settings like this one when I set E1 card connected directly
to Asterisk. In this way every call dialed with pass 29 will be accepted. I
have a billing which filters caller ID number and address calls to each account
with same caller ID number previously set
Hmm.. Interesting,
I didnt try to implement it this way... but, if its the same libraries
used for Office communicator, than it supports only SIP over TCP or TLS, since
asterisk doesnt support any of those its impossible to connect them
directly...
If udp works, maybe the registration
Hi,
The error /usr/lib/divas/divactrl load -c 1 -Debug produces
complains about the missing protocol image
A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0)
It is not necessary any more to download any firmware files
(they are incompatible anyway). All needed files are part
Hi,
With 30 users and NO transcoding, that is
certainly enough.
Even if you use real-time
configuration (that requires a SQL server)
Now, if you system will be accessible both
from inside (LAN) and outside (Internet), I would advice a second network card (10/100)
Regards,
T.
I have had 2 GXP-2000 for a while now and been slowly following the
firmware releases made by Grandstream and am now up to 1.1.0.13. This
version works really well on these 2 original phones (MAC's
00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's
00:0B:82:09:xx:xx).
When trying to figure out why something's not working, is there any way to
have the output specify which line of extensions.conf was being executed?
I mean, sure, I could pour a million NoOp()'s into it, but that's not
exactly scalable, nor easy. It would be really nice if, instead, along
with
Following on from a posting yesterday from Kevin, I have the following
in the dialplan:
exten = 709,1,AddQueueMember(SomeQueue|Local/[EMAIL PROTECTED])
I am on extension 706.
From the CLI:
SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s
holdtime), W:0, C:0, A:3,
The only issue with 1.1.0.13 which affects only certain versions of the
gxp-2000 is the display blanking issue on very early phones.
It sounds like you have a faulty phone and should return it for a
replacement.
On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote:
I have had 2 GXP-2000 for a
Howdy,
have working realtime queues using queue_members looking something like;
queuea|Local/[EMAIL PROTECTED]|0
queuea|Local/[EMAIL PROTECTED]|1
queuea|Local/[EMAIL PROTECTED]|10
Regardless of what strategy is used in the queues
(roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER
Thanks for all replies Now, if you system will be accessible both
from inside (LAN) and outside (Internet), I would advice a second network card (10/100)Actually the machine has two interfaces - 1000 and 100 Mbit/s
CheersAndrew
___
--Bandwidth and
On Wed, 14 Jun 2006, Marc Rohlfing wrote:
Hi,
The error /usr/lib/divas/divactrl load -c 1 -Debug produces
complains about the missing protocol image
A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0)
It is not necessary any more to download any firmware files
(they
Thats what I thought the problem might be, so I have just now upgraded the
other phone to 1.1.0.13 and its exactly the same, no speaker phone and
hangs from a soft reboot.
I also tried the audio loopback in the factory functions menu, this
loopback's fine with the older 1.1.0.13 phones but does
Can you help me, how to update the old sangoma driver?
I downloaded the last driver from sangoma's web.
kind regards,
Szolke
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works
well with 1.1.0.11 firmware.
I can send you this firmware, if you mail me off-list.
Bye
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
Piotr Chytla schrieb:
On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote:
Have you only one BN-Card? or more?
I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank.
i have two cards, had compareable problems.
PCM was the magic word ...
from my
The problem was fixed in 1.2.0
amna saleem wrote:
No , actually I am using Asterisk-1.2.9.1
I will try the q option though
Thanks and regards,
Amna
On 6/14/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I assume you are using 1.0.x. Add the q option to the Meetme
extension. 1.0.x
Thanks for the offer, but I have just tried 1.1.0.11, it is available
publicly and it has the same problems on these 2 phones.
On Wed, 14 Jun 2006, Mimmus wrote:
If can help, I have 80 00:0b:82:08 :xx:xx GXP-2000 phones and they works
well with 1.1.0.11 firmware.
I can send you this
Hi,
I have cable modems Arris with MGCP protocol. And I need PacketCable NCSpatch for Asterisk. http://asterisk.urtho.net/
doesn't work!-- Pagarbiai,Giedrius AugysSiauliu Universitetas, ISTIP telefonijos inzinieriusTel. 8 41 590408Mob. Tel. 8 678 05790el. pastas
[EMAIL PROTECTED]
On Tue, Jun 13, 2006 at 01:47:27PM +0200, Koen Van Impe wrote:
Why still use mpg123?
Start using format_mp3 from asterisk-addons and your * will play mp3 by
itself...
Not to mention that an mpg123 package is availble in Debian-nonfree .
http://packages.debian.org/mpg123
Hi,
i got my Grandstream GXP-2000 phone today and want to configure it
with TFTP. I downloaded the firmware 1.1.0.13 and put it into my
tftp-server directory.
Then I downloaded the template from:
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the web
interface regarding this feature. I just want callers to hear the busy
tone when the called party is at the phone. Probably I can accomplish
this by using the
Is new to me that using G729 codec is a problem when sending DTMF.
Could it be that you are a little bit confused? Usually the problems
with DTMF depend on how the phone is configured and how Asterisk is
configured (DTMF using SIP INFO, RFC2833 etc), check this out:
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote:
GSM
and what is the size in KB that gsm spent?
bp
On 6/9/06, Pablo Allietti [EMAIL PROTECTED] wrote:
hi all, i saw in digium that the codec g729 is not free. exist
another
codec with low
- Danny Froberg [EMAIL PROTECTED] wrote:
Regardless of what strategy is used in the queues
(roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER
That is not how penalties are supposed to work. Calls are delivered to the
lowest-penalty members that are considered available (i.e.
Hi,
I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099
GET cfgMAC
What does errorcode 4099 mean?
Best regards,
Matthias
--
Programming today is a race between software engineers striving to
I'm sure this a very common and easy thing to do with Asterisk, but for
the life of me I can't find the application that will allow me to open a
Zap channel.
Real world example: To be able to connect to an open Zap channel, so it
would allow me to say, join in on a call that was originally
Hi
Sorry for the long response time. I was away for a while and am now
going over the asterisk-users backlog.
On Sun, Jun 11, 2006 at 06:12:50PM +0200, Olivier Saulnier wrote:
Tzafrir Cohen a écrit :
I'm still not hapy with that as a default. It should provide you a basis
for manual editing
On Sun, Jun 11, 2006 at 07:07:12PM +0200, Olivier Saulnier wrote:
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN]
You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.
On Wed, 2006-06-14 at 14:05, Matthias Fechner wrote:
Hi,
i got my Grandstream GXP-2000 phone today and want to configure it
with TFTP. I downloaded the firmware
On Wed, 2006-06-14 at 15:46 +0200, Matthias Fechner wrote:
Hi,
I was now successful in getting syslog messages.
Syslog says the following:
Jun 14 15:43:57 192.168.0.117 GS_LOG: [MAC][708][FF71][0101000D] ERROR 4099
GET cfgMAC
What does errorcode 4099 mean?
I don't know but it looks
Hi I use Asterisk with some SIP phone (grandstrea), while with my
notebook when I'm out of home/office I use X-lite and all work.
Some days ago I try to install wengophone and I decided that I want
replace X-lite for use wengophone as client for my Asterisk.
One of the reasons is that
This will just pick up the line
exten = *01,1,Dial(ZAP/1/)
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Carey O'Shea [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, June 14, 2006 9:48 AM
Subject: [Asterisk-Users] Which
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 10:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote:
Agreed.
-Original Message-
From: Santosh Rao [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 13, 2006 11:19 PM
To: Martin Joseph
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
asterisk has a extremely cool documentation. The wiki has
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 1:47 AM
To: Santosh Rao; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
Santosh Rao a écrit :
asterisk has a
I should note that we are not running the Digium g729 implementation, but the
intel one.
Also, to not angry people, this ofcourse isn't used in our production
environment, only for testing if we want g.729.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På
99.999%
I suspect you will see this drop as traditional PBX'es start to use
commodity parts. My Mitel ICP 3300 has a Maxtor 10 gig hard drive in it
(same as an Xbox!)
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
You need to encode txt configuration file using tool provided on GS site.
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthias Fechner
Sent: Wednesday, June 14, 2006 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Hi
Please I want to make a schedule to make list of extensions in a conference,
automatically the system call them and put them in a conference mode
Can any one help me
Regards
*
No employee or agent is authorized to conclude any binding
Hi
Please I want to make a schedule to make list of extensions in a conference,
automatically the system call them and put them in a conference mode
Can any one help me
Regards
*
No employee or agent is authorized to conclude any binding
Hi,
calling a partner on the other side of a SIP trunk, call gets disconnected
immediately after answer. This is the content of log file:
Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel:
SIP/cerved-out-6eba
Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging
Hi
Please I want to make a schedule to make list of extensions in a conference,
automatically the system call them and put them in a conference mode
Can any one help me
Regards
*
No employee or agent is authorized to conclude any binding
I swear Dial(Zap/X) was the first thing I tried and it didn't work, but
now it works fine... hmmm maybe I forgot to reload my extensions or
something like that.
Thanks though.
On Wed, 2006-06-14 at 10:03 -0400, Mailing List wrote:
This will just pick up the line
exten = *01,1,Dial(ZAP/1/)
Hi Gareth,
Gareth Blades wrote:
You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.
thx that was the problem. Now it works fine.
Best regards,
Matthias
--
Programming today is a race between software engineers
I tried your suggestion
and found out that someone/something I don't know whether that is an MS
RTC or Asterisk is having problems if the same Windows application is
using Manager and SIP at the same time. At least for now, it has always worked,
if I tried to initiate Originate
I am running on 1.2.7.1 and have an intermittent problem when making outgoing
calls. Sometimes the calling party does not hear the ring tone in their
handset, but the call goes through. From my extension I have only had 3 calls
like this in the last couple of weeks, other people have had 3 or
It has a conceptual problem i have notified several times to
Cisco-Linksys. It could not be disabled, i have the same problem with my
queue extensions, and the way to resolve has been to use call-limit=1 in
extensions.
i hope this helps.
Tommaso Calosi escribió:
I'm trying to disable call
Does anyone know where I can find some good DUNDi docs?
The ones are dundi.org are absolutely horrible.
The examples in dundi.conf are pretty much useless.
I still can't figure out why Digium can't write some good documentation. It's
their 'baby' after all. This really drives me nuts and pisses
In fact the www.onsip.org documentation does include discussion about the
avpops. It even gives an example of call forwarding using these functions.
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: 14 June 2006 15:06
To:
Hello,
I have problems using dynamic features while an other person is doing
call waiting in a call.
I define a dynamic application mapping in features.conf as the
following:
testfeature = *9,caller,Playback,tt-monkeys
I also set DYNAMIC_FEATURES = testfeature. The mapping is working well.
On Wed, 14 Jun 2006, Douglas Garstang wrote:
The examples in dundi.conf are pretty much useless.
I still can't figure out why Digium can't write some good documentation. It's
their 'baby' after all. This really drives me nuts and pisses people off in
general. I've been dicking around with
Hi,
I'm stuck writing a
Web GUI because nothing out there is exactly what I need. I'm not writing
something as extensive as what _is_ out there, but just something that allows
users to change where their calls are forwarded and other small things like
that.
What I wanted to
know is what
Well, it does help, but it causes the announced transfer to fail,
because if you set call-limit=1 you cannot dial out to announce the
transfer...
Alberto Sagredo wrote:
It has a conceptual problem i have notified several times to
Cisco-Linksys. It could not be disabled, i have the same
Hi,
Check this document, it helped me to build our DUNDi Network.
http://leifmadsen.com/papers/dundi-intro.pdf
Frédéric Marti
==
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: mercredi,
Make sure you have /etc/asterisk/indications.conf set up.
People that don't know any better might tell you to use the r option
to Dial. Those people are confused. Don't do that until you have tried
everything else.
Tim Sharp wrote:
I am running on 1.2.7.1 and have an intermittent problem
Wow! 22Kbps of overhead? Are you sure? That sounds like way too much
overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any
other suggestion?
Thanks,
Daniel
On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote:
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so
Carey O'Shea wrote:
I swear Dial(Zap/X) was the first thing I tried and it didn't work, but
now it works fine... hmmm maybe I forgot to reload my extensions or
something like that.
Don't expect Dial(Zap/X) to work. Expect Dial(Zap/X/) to work.
--
Now accepting new clients in Birmingham,
On Wed, Jun 14, 2006 at 11:51:06AM -0400, Mike wrote:
Hi,
I'm stuck writing a Web GUI because nothing out there is exactly what I
need. I'm not writing something as extensive as what _is_ out there, but
just something that allows users to change where their calls are forwarded
and other
Hi
all,
I have to connect an
asterisk box to a legacy pbx using QSIG signalling : where could i find more
information or any sample ocnfiguration file?
Has anyone never
used it?
Thanks in
advance.
Giordano
Grandis
___
--Bandwidth and
Yes, what is it you attempting? I use DUNDi extensively, though you are
correct that the existing docs don't go very far in describing some
things.
Regards
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Wednesday, June 14,
Hi,
using Sangoma drivers:
- doing 'lsmod', I see:
zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
I'd like to avoid loading all these modules. What have I to do?
- do I need to have 'zaptel' startup script under /etc/init.d ?
Thanks
--
Domenico Viggiani
Welcome to the wonderful world of VoIP, where people are eager to move
from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in
voice quality, and then throw over 20kbps of RTP, IP and related
overhead on top of them. Isn't IP wonderful? :-)
Regards,
Steve
Daniel Salama wrote:
Well, with 16 phones, it might be worth putting a
'satellite' asterisk in their office, have it handle local
transfers, and act as a protocol converter, talking sip to the
phones and (trunked) IAX2 to the outside world.
An embedded low power system would do fine.
You might even get away with an
Is there a method for detecting return values of applications in the
dial plan?
Thanks
Mark Price
UNETA
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I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/3004-fcfb(256) to SIP/3003-c1c3(2)
Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586
I think each application returns it's own value in a variable defined by
the application.
Mark Price wrote:
Is there a method for detecting return values of applications in the
dial plan?
Thanks
Mark Price
UNETA
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Thanks for clearing that up Kevin.
Now on to figure out how to PauseQueueMember when enough NOANSWER's
has been detected so he don't fubar the entire queue.
Would be alot cleaner than sending callers to ever higher level queues
*sigh*
Kevin P. Fleming wrote:
Regardless of what strategy is
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net = dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
Contact Digium to purchase a G729 license.
Osama Kamal wrote:
I am having a problem with asterisk transcoding GSM and G729 codecs, the
error message is below:
Jun 14 09:38:12 WARNING[18292]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/3004-fcfb(256) to
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi Docs
On Wed, 14 Jun 2006, Douglas Garstang wrote:
The examples in dundi.conf
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