From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Mon, 16 Apr 2007 08:45:38 +0300
On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote:
(But if Zaptel and Hylafax can share an X100P driver ...)
Where can you find a modem driver for a X100P?
Kinda my question, too. Motorola used to have an
Steve Totaro wrote:
Stephen Bosch wrote:
Steve Totaro wrote:
You could try to get it working but it may never be 100%. If your needs
are 100% then I suggest using a standard fax and get an analog line and
do it the old fashioned way. If you need Hylafax type features then buy
a modem
My problem is that I already did what you proposed and I did not have much
sucess.
An Hylafax server with an external/internal modem using a driver from
linmodems worked in around 80% of the cases. To improve this rate you need a
modem like the ones offered by Mainpine which were way out of my
On Sun, 15 Apr 2007, Steve Prior wrote:
I've got a run of Shielded Twisted Pair (4 conductors) which used to be a
Token Ring Network drop and I'm not using it anymore. Would it be decent to
replace the ends with normal analog phone connectors and use it for a phone
extension, or is STP
Hi List,
I need to change my provider, at this time Asterisk box is on VOIP trunk.
I have two options, T1 or 15 analog lines.
I have some experience with analog and I have had two main issues with it.
first is echo (I have not tried HPEC yet) and second unpredictable volume.
The question is, if I
In general, you're going to have better luck with a PRI. A lot of echo
occurs at the point where analog lines are broken out of a digital
transport. Also, from an economic standpoint, you *really*, *really*
don't want to order 15 analog lines. Generally the rule of thumb in most
areas of
I apologise now if I have managed to completely misunderstand this whole
subject!
I've built a small PC and loaded Etch 4.0 from the netinst cd.
I did 'apt-get install asterisk-bristuff' which seemed to work
but, it doesn't seem to have installed any files/modules for zaptel?
ztcfg zaptel
I also dropped the quotes on the dnis=${IVR-Exten}.
That's only allowed if the dnis column doesn't contain a string.
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp
BUT if the agent have to go elsewhere for some minutes (coffe
break, go
to piss, and so on..), usually he press the 'hold' button on the
phone;
Does the phone have a DND (Do not disturb) button?
yes, the phone have this options; I have to check if that works
Are all the agents trained to
Just run down to your local Radio Shack...and KISS.
http://www.radioshack.com/product/index.jsp?productId=2062696
Mark C.
Klaverstyn, David C wrote:
This is what I want. Do you have any URLs to such a device as I cannot
find any.
-Original Message-
From: [EMAIL PROTECTED]
On Mon, Apr 16, 2007 at 08:49:18AM +0100, Simon Faulkner wrote:
I apologise now if I have managed to completely misunderstand this whole
subject!
I've built a small PC and loaded Etch 4.0 from the netinst cd.
I did 'apt-get install asterisk-bristuff' which seemed to work
but, it
Hi all,
I have a client that is setting up a premium phone service and is required
by the Telephony regulator to stream call cost announcements into the call
at certain periods during the call.
For instance when the call cost is €3 per minute there needs to be an
announcement after 10 minutes
On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote:
1) But how do I inject them into the SIP channel.
2) How do I time the injection so that the correct message is played at
the correct time.
take a look at the L() option to Dial().
--
Regards, /\_/\ All dogs
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the
richard Coco wrote:
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange
On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard Coco wrote:
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the
--- J. Oquendo [EMAIL PROTECTED] wrote:
richard Coco wrote:
Hi all,
i have asterisk 1.2.17 with sip tcp support and i
am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the
HG3540.
But when i try to place a call from Asterisk to
--- Dinesh Nair [EMAIL PROTECTED] wrote:
take a look at the L() option to Dial().
The original poster said he need to play different
messages at different call durations. In order to do
that you would need to dynamically alter
LIMIT_WARNING_FILE as the call progressed.
Regards
Jon
Jon
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:* 9722/asterisk
972 is the tie access code from Hiapth to Asterisk.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard
Coco wrote:
Hi all,
Hi everyone,
I'm in trouble with queue.
There are a little local radio station with one studio and we have to
switch queued callers to the live program. Everything works fine
(counting callers, periodic announcements), but while the announcement
is played for 'firs in line' caller, studio gets a
Hi guys,
I have an asterisk box with sip 20 internal extensions and 100 lines
registered on a external voip provider.
For most part of time, it work fine, but in few moments it act
ignoring sip packets becouse my ip phones can't register in asterisk
and asterisk can't register his 100 lines in
Thanks.
On 4/16/07, Dinesh Nair [EMAIL PROTECTED] wrote:
On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote:
1) But how do I inject them into the SIP channel.
2) How do I time the injection so that the correct message is played at
the correct time.
take a look at the L() option to
yeah, it would be good to have 2 different messages. But I guess I could
adapt the message to be generic enough to cover both
scenarios - your call has now cost over €X. This might scrape through with
legal requirements.
cheers for the feedback - I appreciate it.
On 4/16/07, Jon Farmer [EMAIL
Frederico Madeira wrote:
Hi guys,
I have an asterisk box with sip 20 internal extensions and 100 lines
registered on a external voip provider.
For most part of time, it work fine, but in few moments it act
ignoring sip packets becouse my ip phones can't register in asterisk
and asterisk can't
I'm wondering about the difference between Cisco Call Manager and
SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to
speak through a NAT to an asterisk box, without having to do a bunch of port
forwarding on the NAT device.
Without the nat, everything works fine.
If the
I am using a billion hfc card
apt-get install zaptel-source
m-a a-i zaptel
Precompiled zaptel drivers should hopefully be added soon to Unstable /
Testing .
Thank you :-)
___
--Bandwidth and Colocation provided by Easynews.com --
sorry, it works with upd... I am now able to make and
to receive calls.
thx...
--- richard Coco [EMAIL PROTECTED] wrote:
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:*
9722/asterisk
972 is the tie access code from Hiapth to
Hi,
While Dial rings can a caller press 0 (or other number) to leave a
voicemail? I found that with a # can transfer to different context. I
want to use that two features together.
--
Suich
___
--Bandwidth and Colocation provided by Easynews.com
Call setup/teardown is handled with the SIP protocol while the actual call
audio is handled with RTP I think. Check the config of your NAT devices
relative to RTP.
scd
On 4/16/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I'm wondering about the difference between Cisco Call Manager and
Luca Corti wrote:
On Fri, 2007-04-13 at 17:46 +0200, map wrote:
Linksys SPAs work well with Asterisk
I know, I use them and besides some initial nasty bugs and occasional
quirks they are quite nice. I also think they are not so ugly.
Luca, what sort of nasty bugs and quirks have you
On Sun, 2007-04-15 at 14:53 -0600, Greg Woods wrote:
when I recompiled zaptel with 1.4.1 and
installed that, the problem is gone. I don't know if this was due to
changes I made in the 1.4.0 zconfig.h file, or that there were fixes in
1.4.1.
I checked, and the zconfig.h file that is in my
If you want to be able to run accurate reporting, you should tell the
agents that they must log out whenever they are unavailable to answer
calls. If accurate reporting is something you may be interested in
doing in the future, then make this the rule now.
Autoanswering queues is great for
On Saturday 14 April 2007 00:52, [EMAIL PROTECTED]
wrote:
Can you tell me if this sounds sane? We are planning on using a Dell
933Mhz dual CPU server, with 1GB of ram for our Trixbox setup. We
will have 7-10 internal phones, and maybe 3-4 max outbound
connections at a time. We will have
Quoting [EMAIL PROTECTED]:
I have two options, T1 or 15 analog lines.
The question is, if I use TE100 with PRI , will I have same issues?
I would appreciate any comments and sample zaptel.conf and
zapata.conf
15 lines should be well beyond the cost justification point for a T1 and
you will
might be old now hehe :)
pre10 for libunicall, unicall, libsupertone, libmfcr2
0.0.3 for spandsp (final) and 0.0.4pre
they were already out from testing folder/branch
[EMAIL PROTECTED] wrote:
Hi Nivlekch,
Thanks for that, just a comment:
What do you mean by new packages? new for spandsp,
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote:
Hello,
Did you find anything while testing the LAN? Also, can you confirm that
switching the switch, cabling, etc. did NOT solve the problem?
It did not.
We finally changed the server itself and reinstalled from a
Hi guys,
i've installed asterisk to handle multiple voip accounts. I've looked
at CDR configs, and managed to have cdr-csv files growing after each
call. It would be easier to check my locak asterisk cdr's than logging
into each account and check them at the provider website.
i found that
Oquendo,
My provieder require sip digest authebtication:
Asterisk send register to sip provider
sip provider response with 401
asterisk send register again with authentication header
sip provider response ok
This is normal process, when problem happen, this process ocour until
401 message, and
An open source queue log analyzer that we find useful...
http://www.micpc.com/qloganalyzer/
in combination with CDR analysis...
http://www.areski.net/asterisk-stat-v2/about.php
regards,
Drew
Rilawich Ango wrote:
HI all,
I have a queue say 5000 and there are 10 member in the queue. When
Hello,
It's good to hear that you have success with VICIDIAL and FreeePBX,
would you be willing to do any documentation on the steps you took to
get this working reliably?
Even something minimal on the VICIDIAL Wiki would be very helpful to a
lot of people.
http://eflo.net/VICIDIALwiki
Thanks,
On Apr 12, 2007, at 1:49 PM, Kevin P. Fleming wrote:
Got off the phone with Polycom on this I have the same
problem with
my new 601 phone here (haven't seen the problem on the 650).
I am using an IP650 with the latest firmware (and the corresponding
sip.cfg file) and I have seen this
Hello,
I have just updated my Asterisk installation from 1.2x to 1.4 (on
FreeBSD) - mostly everything seem to work fine.
However, I use G.729 pass-thru - and I have before successfully used the
following setup:
http://www.voip-info.org/wiki/index.php?page=Asterisk%20G.729%20pass-thru
Sounds like your VoIP provider is incorrectly sending you an Answer before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement that
is only on your VoIP trunk. I would double check that, and then contact
your
Hmm - just received an email from these guys last week. I know
nothing about them.
On Apr 15, 2007, at 9:23 PM, cb wrote:
On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote:
When a call comes in I want to ring an extension that happens to
be loud speaker. The users can the press *8
While I don't use 1.4, it could be that the registration failure (you said
100 registration lines with your provider?!?) are blocking the phones from
registering. This is only a guess, I don't know for sure.
On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote:
Oquendo,
My provieder
I have a customer who wants their receptionist to input the users' long
distance PINs for the because they use each others pins. I am having
trouble coming up with a way to do this because of creating a channel
between the user and receptionist, dropping the channel and its variables
and
Looks okay to me. either the number you are testing with your VoIP provider
has an automated response which answers the call at the same sec you sent
the Invite request or the provider is sending False Answer Supervision...do
a sip debug and check while you make the call.
On 4/16/07, Adam KOSA
Suity Zsolt wrote:
Hi everyone,
I'm in trouble with queue.
There are a little local radio station with one studio and we have to
switch queued callers to the live program. Everything works fine
(counting callers, periodic announcements), but while the announcement
is played for 'firs in
Hello all,
I'm moving from Asterisk 1.2 to Asterisk 1.4.0-beta3. I'm just
configuring an extension associated to an external sip-based voip
service provider in order to be able to initiate/rcv pstn calls.
Is there any relevant issue when moving from v1.2 to v1.4? Maybe
something related to
Jovanny Saravia wrote:
Hello asteriskers, I hope someone could help me ... !!
I bought a TC400B, and I am testing doing calls with G729 and G723.
When I used G729 it works fine, but when I try to use G723 the RTP has
very low quality and is not possible to hear to the other person in
the
Hi,
First, sorry to repost, As I didn't get any replies, maybe this time, I will
get more lucky.
I was wondering if there was a way in Asterisk (agi script, asterisk-itself,
whatever ... ) to send a notification to the user (Mail, SMS like voicemail
application is doing) if the user has called,
On Sun, 15 Apr 2007 22:10:34 -0700, Yuan LIU wrote
From: Steve Totaro [EMAIL PROTECTED]
Date: Sun, 15 Apr 2007 22:36:15 -0400
Stephen Bosch wrote:
Steve Totaro wrote:
You could try to get it working but it may never be 100%. If your needs
are 100% then I suggest using a standard fax and
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement
that
is only on your VoIP
What's wrong with:
User calls receptionist gives her the number.
Receptionist hits XFER on her phone... punches in pin and dials the number,
then hits XFER to complete the transfer?
This could all be done outside a dial-plan... just use the phone's transfer
feature. If you MUST have
Correct, if I am understanding you correctly when an announcement is
playing to the caller the caller is in 'limbo' until the announcement
completes. Once the announcement completes, the caller will go through.
On 4/16/07, Suity Zsolt [EMAIL PROTECTED] wrote:
Hi everyone,
I'm in
If the voicemail portion is reached, but hungup on, the extapp portion
of the config file is still executed. So you could have an external
app which does any number of things (IM, etc).
Rob
Jean-Marc Salsa wrote:
Hi,
First, sorry to repost, As I didn't get any replies, maybe this time,
I
I just had this issue, and fixed it with the 501Presumably the 601 has
the same thing.
If you had the old firmware before, and you forced your phone to re-register
every x seconds, take that line out. The phone will become more responsive.
To handle NAT, use nat.keepalive.epxires instead
Playback automatically answers the call unless you tell it not to. See:
show application playback in the Asterisk CLI.
Adam KOSA wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call actually completes. I
On Mon, 16 Apr 2007, David Cook wrote:
Remember, you don't need to activate all 23 lines so if you just need 15
then you can activate only that number. You also can have potentially
hundreds of numbers that terminate on this group of lines. This makes
some of your coding a little different
The playback wait command may be what's doing it.
On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call actually completes. I would contact your VoIP provider.
I
Carlos Chavez wrote:
You can have Asterisk and Hylafax on the same machine when you use
IAXmodem. This is the way I fax in my office with 99% success rate. I am
I've just compiled stats for the last 30 days on our system for
management, info below:
Awesome, any chance you can share your resource specs?
Thanks
Miles
Asterisk works great with openvz. Ive run 4 VE's with combined average
around 32 simultaneous calls at any time and you wouldn't know the
difference.
___
--Bandwidth and
The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.
Joss.
On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call
Hi All -
Got off the phone with Polycom on this I have the same
problem with
my new 601 phone here (haven't seen the problem on the 650).
I am using an IP650 with the latest firmware (and the corresponding
sip.cfg file) and I have seen this behavior. It is most noticeable
when
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs. I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten =
Tom Lynn wrote:
You could also look at Oreka at sourceforge.
Tom,
We are moving in that direction, but we don't have it in production
yet. Since it is a packet sniffing solution, the limiting factor
becomes the point at which the kernel starts to drop an unacceptable
number of packets. A
Thanks for the answer,
I already use this extapp, to set on another server the MWI.
But how to know if user has not let a message ?
One could guess that 0 is the number of message to trigger such a
notification ... but 0 is the number of message as well when you erase all
your messages, so you
Hi Victor -
I'm moving from Asterisk 1.2 to Asterisk 1.4.0-beta3. I'm just
configuring an extension associated to an external sip-based voip
service provider in order to be able to initiate/rcv pstn calls.
Is there any relevant issue when moving from v1.2 to v1.4? Maybe
something related to
Well, it _was_ up again Friday, and now it's down again Monday! :(
Moises Silva wrote:
Hum, I know Stefan, he is an asterisk-java dev, but he is not online
right now, I will let him know ASAP. Thanks!
On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote:
Does anyone know who maintains the
And by the way, I forgot,
If I remember carefully, there is not so much info passed to this script (VM
Number, context Number of messages) ...
So for example, how do you get the caller ID info ?
Thanks again,
JM
On 4/16/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Thanks for the answer,
I
Steve Edwards wrote:
On Mon, 16 Apr 2007, David Cook wrote:
Remember, you don't need to activate all 23 lines so if you just need 15
then you can activate only that number. You also can have potentially
hundreds of numbers that terminate on this group of lines. This makes
some of your coding a
There is no absolute way to verify if the user left a new message, since
it only tells you how many messages are currently in the box. If not
many messages are sent you could do a stat on the newest voicemail
file in their new folder. Then see if its more than a few seconds old.
If its only a
This is interesting to me.. I'm a newbie, so please forgive a dumb
question, but what use is it to play a message if you don't pick up the
phone first?? Who's hearing it?
-Original Message-
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs.
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is being set prior to
the recording, I need to pass this variable to the slave
hello, is there any equivalent, that is currently usefull, if I have
some iax connections with jitter spikes and another with minimal jitter?
for my jittery connections, I don't like to shrink jitter buffer too
fast, because another jitter spike can occur again and small jb can't
cover it.
as I
Has anyone figured out the way of getting the caller id for BSNL on Asterisk
1.4.2
I have tried following link
http://bugs.digium.com/view.php?id=6683nbn=24
but was not able to get it, although did not ge any error too.
I always get the caller id as asterisk.
Can someone please help.
Regards,
Steve Jones wrote:
This is interesting to me.. I'm a newbie, so please forgive a dumb
question, but what use is it to play a message if you don't pick up the
phone first?? Who's hearing it?
Many types of connections allow you to do early audio or on hook
audio. A perfect example of this is
Without using Dundi or SER, any thoughts on the following anyone?
Has something similar been implemented anywhere so as to me not
having to horribly butcher code...
4 servers SIP1-4
User1 -- -- SIP1 --
\ /\
User2 -- Go to
Steve Edwards wrote:
On Mon, 16 Apr 2007, David Cook wrote:
Remember, you don't need to activate all 23 lines so if you just need 15
then you can activate only that number. You also can have potentially
hundreds of numbers that terminate on this group of lines. This makes
some of your
Looks like a PolycomSoundPointIP bug to me. The Via header, Contact both
has 66.38.177.611:5060
Thanks,
Neel
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike
Sent: Friday, April 13, 2007 7:28 AM
To: 'Asterisk Users Mailing List
Doug Garstang wrote:
Well, it _was_ up again Friday, and now it's down again Monday! :(
sorry, there seem to be problem with the nameservers.
I'll hava a look at it asap.
=Stefan
signature.asc
Description: OpenPGP digital signature
___
--Bandwidth
I've seen this before, in an ISDN card (can't recall which one) that defaults
the incoming language to german. Since you don't have german, it defaults to
english files but voicemail still runs through the german logic (e.g. 1F for
femail). I reported a bug against this, it was silently killing
Hello Mr. Sanjay,
I tried a lot to get caller ID in India. But, It doesn't work. I came to know
that Its not possible to get caller ID in India (Not only in India, don't get
caller ID in some countrys).
Thank you.
Regards,
Chandra.
Sanjay Rajdev [EMAIL PROTECTED] wrote: Has anyone figured
Use round robin on DNS with a replicated DB on each server
On 4/16/07, J. Oquendo [EMAIL PROTECTED] wrote:
Without using Dundi or SER, any thoughts on the following anyone?
Has something similar been implemented anywhere so as to me not
having to horribly butcher code...
4 servers SIP1-4
People, I've setup Asterisk in a basic mode with SIP protocol. In the
future I wanna connect several offices each one with an own Asterisk
server, using IAX because I read it has no firewalling problems using
just one UDP port for control and data -aming other advantages- . SIP
has NAT problems I
Yossi Ben Hagai wrote:
The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.
thanks a lot to everyone who answered, this, of course solved this
issue, it's also in the doc, i just didn't have the idea to look at
playback's manual :(
Matt,
It's make no sense. Asterisk should process messages in diferents
threds, not in queue.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
2007/4/16, Matt [EMAIL PROTECTED]:
While I don't use 1.4, it could be that the registration failure (you said
100 registration lines with
On Mon, 16 Apr 2007, Stephen Bosch said something to this effect:
There are some parts of the world where you can't get partial PRI
anymore, and there's an ugly unserved gap in between analog lines and a
full PRI :(
Even where you can get a fractional PRI, there's not really a lot of
It's possible, have the SIP clients use SRV records for server location and
use asterisk ARA to store SIP peers and extension.conf on DB. if the users
are not behind NAT it should work.
(open)SER is much better solution for high traffic / availability setups.
On 4/16/07, J. Oquendo [EMAIL
Philippe Lindheimer wrote:
I've seen this before, in an ISDN card (can't recall which one) that
defaults the incoming language to german. Since you don't have german,
it defaults to english files but voicemail still runs through the german
logic (e.g. 1F for femail). I reported a bug against
What I'm retrying to do is update mysql field with the new message ID
that was just recorded. Ideally, I'd like to specify
the field to update using a variable ${BINID} and use ${NEWPHRASENAME}
for the value - I'm not sure asterisk will allow
using a variable for the field name and if not, I'll
I understand... but I know, at least in 1.2 if there was a DNS failure
for some reason asterisk stopped doing anything else.
That is... if I restart asterisk and it goes to register with , say, my 6
SIP upstream peers... but they are timing out for some reason asterisk
won't
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
From: Jesus Mogollon [EMAIL PROTECTED]
Date: Mon, 16 Apr 2007 13:33:16 -0400
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is
On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote:
Has anyone figured out the way of getting the caller id for BSNL on Asterisk
1.4.2
I have tried following link
http://bugs.digium.com/view.php?id=6683nbn=24
but was not able to get it, although did not ge any error too.
I
${CALLERID(num)}
or
${CALLERID(name)}
Sanjay Rajdev wrote:
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
___
--Bandwidth and
I don't think that this problem is DNS, becouse asterisk can send
register to my provider and he can replay to asterisk, so, DNS is
working fine.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
2007/4/16, Matt [EMAIL PROTECTED]:
I understand... but I know, at least in 1.2 if
Sanjay Rajdev wrote:
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
That information is in UPGRADE.txt -- part of the Asterisk source.
___
--Bandwidth
${CALLERID(num)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sanjay
Rajdev
Sent: Monday, April 16, 2007 13:39
To: asterisk-users
Cc: asterisk-dev
Subject: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERIDNUM} is DEPRECATED in 1.4.2 what
Hey All,
I've been using Asterisk for a couple years now, but have always had
some unsolvable audio problems. I get audio stuttering and popping
quite often. Even if I have just one call up! The server is a Dual
Duo-Core 3.0ghz Xeon Processor PC with 4 GB or ram. It just seems to me
that
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