Hi,
Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call to 700. No problem. Later, when it's picked up from
701, can it NOT be transferred again?
Moj
No I don't park the call.
try setting
transfer=no
or
notransfer=yes
in iax.conf
Depending on the age of your asterisk version.
Tim.
- Original Message -
From: randulo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 20 February 2008 18:12:01
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
For the brave: use modules.conf without 'autoload = yes'.
This promises you many hours of interesting dialplan debugging. Enjoy.
Is there any method of automatically parsing a dialplan and generating a
list of
Hi All
Agter a bit of logging to a syslog server, I found a peculiar entry
today, ironically right after a call failed to transfer. They key
sequence and call path used until it gets transferred is as follows
* Phone rings on Asterisk
* Asterisk transferres to the receptionists phone
Dear list,
We have an weird problem with our FXO card (TDM01B). When I made a call
using this channel, all goes well, the called phone rings but when the
called phone answers the call. In me handset I can hear an weird sound like
a Clack. I tryed diferents TDM cards and modules, and my
On Fri, Feb 22, 2008 at 02:33:01AM -0500, Matt Florell wrote:
Hello,
I was never able to get the TE407P card running on a 2.4 Linux kernel.
Using a 2.6 kernel I was able to get it working.
What error(s) do you get? On what platform / kernel exactly?
Not really surprising since a lot of
On Fri, Feb 22, 2008 at 01:28:58AM -0800, Steve Langstaff wrote:
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
For the brave: use modules.conf without 'autoload = yes'.
This promises you many hours of interesting dialplan debugging. Enjoy.
Is there
I forgot to say that I'm using bristuff-0.4.0 with zaptel 1.4.4, libpri
1.4.1 and asterisk 1.4.9
Thanks.
2008/2/22, voip crazy [EMAIL PROTECTED]:
Dear list,
We have an weird problem with our FXO card (TDM01B). When I made a call
using this channel, all goes well, the called phone rings but
Anyone from South Africa out there that has gotten SMS over Telkom lines
right?
Ive found the SMSC but I dont have the foggiest how to go about it
___
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asterisk-users mailing
Linksys SPA942. Tried most of available phones on the market.
These phones sits on companies tables for more then a year.
No problem at all, easy to use, nice(!) to use. I recommend to everybody.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
Hello,
Junghanns and BeroNet offer 8 BRI ports cards.
Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports
subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out
of 2 B410P ?
I know you could (theorically) do this with Junghanns and BeroNet cards.
On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
For the brave: use modules.conf without 'autoload = yes'. This promises
you many hours of interesting dialplan debugging. Enjoy.
Yup, that's what I anticipated, which is why I was asking which
modules I can _safely_ remove
Hi All,
if i do this setup:
|---[ext 100]
|--[router/nat gw]--|
| |---[ext 101]
|
[asterisk]--[internet]---|
|
On Thu, 21 Feb 2008 08:33:20 -0500, C F [EMAIL PROTECTED] wrote:
first off I anwered you to use vi and you complained showing me cat.
There's some misunderstanding. I didn't complain. I just didn't know
if Asterisk only looked for stuff in modules.conf because there was so
little there and so
In article [EMAIL PROTECTED], Eric Wieling [EMAIL PROTECTED] wrote:
No that will not work. You would want three exten = lines, one for
each area code.
And if you have a lot of common dialplan that you don't want to duplicate
between the three extension patterns, put the common stuff up at a
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002
On Fri, 22 Feb 2008 18:50:16 +0800, Ron [EMAIL PROTECTED] wrote:
If i set, canreinvite=yes on all ext, assuming all ip phones have the
same codec, if 100 calls 101, or vice versa will rtp still go thru
asterisk? and same scenario for 200 to 202 or vice versa.
... and I'd like to add to this
On Fri, Feb 22, 2008 at 11:42 AM, Vieri [EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call 4002 then I
would like to do something like:
- lookup 4002 on *1, try to establish a call if it's
REGISTERED here
- if it's
They have their ups and downs. If you live outside the US, localising
your tones is a pain in the proverbial since you have to define every
tone by frequency combination and intervals, although I guess you do
only need to do it once.
One other shortcoming of the 942 is the lack of any usable
That's what I figured, and the way I've always done it. I was just
*hoping* someone knew of a better way that I didn't know about.
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Hello List,
After seeing a few positive responses for the Linksys SPA-942 phones I
was hoping to get some answers on the following questions:
* How do the phones handling system wide paging? Is it similar to
the Polycom phones?
* Can a corporate directory be configured with the phones
Hello,
This was about a year ago when we abandoned putting new systems on the
2.4.33 kernel. About that time we started having other vendors stop
supporting it very well also so it wasn't only that card that we were
having issues with. The problems were related to the CRC Linux modules
and zaptel
If one does a dundi lookup, shouldn't the ${NUMBER}
variable be replaced with the current value?
ie. if I run dundi lookup [EMAIL PROTECTED] shouldn't I get
an answer string like
IAX2/priv:[EMAIL PROTECTED]/4065 (EXISTS)?
The *CLI does not show me the dst extension:
*CLI dundi lookup [EMAIL
I need to add a few phones to an existing installation. They have a
dozen IP430 at the moment. Does anyone feel that there are advantages
to the IP330? Cost is not the major consideration as long as they're in
the same range. (under $175)
Michael
--
Michael Graves
mgravesatmstvp.com
Let me add another variable into the mix...what about the Linksys
SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a
deal at $80 street price.
Michael
On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote:
Michael Graves wrote:
I need to add a few phones to an existing
This is my first time setting up Asterisk in production and we are
buying the Digium TE121-card for use with an ISDN-30 connection. We
are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html
- for handling
More than enough.
Julian.
harry wrote:
This is my first time setting up Asterisk in production and we are
buying the Digium TE121-card for use with an ISDN-30 connection. We
are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
I'm finding the volume of the calls on my wildcard 100XP (clone) is too
low. I understand I can muck with rxgain and/or txgain (which one in
fact will increase the volume of the other party as far as I'm hearing
it?) to deal with this but right now I have them both at 0.00 and I am
concerned
I verified that qualify=no. I am getting this CPU load at 10% without any
peers even registered which is very strange, but it doesn't happen when I run
on 64-bit CentOS 5 kernel. Remi
- Original Message -
From: Jared Smith
Date: Thursday, February 21, 2008 5:55 pm
Subject: Re:
Harry,
I think this system will suffice for your needs. I have a similar setup
working great with 2 Dual Core Xeon @ 2GHz
On Sat, Feb 23, 2008 at 9:21 AM, harry [EMAIL PROTECTED] wrote:
This is my first time setting up Asterisk in production and we are
buying the Digium TE121-card for use
Two questions:
1. Does anyone have a good way to transfer a call from inside
comedian mail to the current extension? The problem is: let's say the
phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd
ring. I come running into my office but miss it by a split second.
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to
On Fri, 2008-02-22 at 14:51 +0100, harry wrote:
This is my first time setting up Asterisk in production and we are
buying the Digium TE121-card for use with an ISDN-30 connection. We
are considering buying a Fujitsu-Siemens Primergy TX200 S4 -
On Fri, 2008-02-22 at 07:43 -0600, Michael Graves wrote:
Let me add another variable into the mix...what about the Linksys
SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a
deal at $80 street price.
I'm quite happy with my SPA-962 + sidecar... I tend to use it more than
the
On Thu, 2008-02-21 at 13:57 +0800, zhao_x_q wrote:
HI, Friends,
Now I have 20 polycom’s SS2 phones. Can Asterisk support 20
users conference meeting?
Yes.
And I want to build HD audio conference by using polycom’s 650 ip phone.
Can asterisk support G722 HD audio conference?
Atis Lezdins wrote:
BTW, we have 512kbs over the iax connection.
G711 needs about 80Kb/sec each way to work. (It's 64Kb/sec plus IP
overhead). GSM needs about 32Kb/sec (13Kb/sec plus IP overhead).
So with DSL 512kbs up and 3mbs down, plenty of room for G711.
Take the weakest
Olivier wrote:
Hello,
Junghanns and BeroNet offer 8 BRI ports cards.
Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI
ports subsystem (2 PCI slots used, but 1 one set of interrupts) could
be made out of 2 B410P ?
You can also use the Sangoma A500 for up to 24 BRI
On Fri, Feb 22, 2008 at 03:22:39PM +0100, Patrick wrote:
And I want to build HD audio conference by using polycom’s 650 ip phone.
Can asterisk support G722 HD audio conference?
Afaik Asterisk only supports it in 1.6beta. If you need a working
solution *now* then have a look at FreeSWITCH
On Fri, Feb 22, 2008 at 07:59:14AM -0500, Matt Florell wrote:
Hello,
This was about a year ago when we abandoned putting new systems on the
2.4.33 kernel. About that time we started having other vendors stop
supporting it very well also so it wasn't only that card that we were
having issues
On Friday 22 February 2008 04:55:13 Vincent wrote:
On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote:
For the brave: use modules.conf without 'autoload = yes'. This promises
you many hours of interesting dialplan debugging. Enjoy.
Yup, that's what I anticipated, which is why I was
Michael Munger wrote:
Two questions:
1. Does anyone have a good way to transfer a call from inside comedian
mail to the current extension? The problem is: let’s say the phone
goes to voicemail after 4 rings, yet I don’t hear it until the 3^rd
ring. I come running into my office but miss
On Fri, Feb 22, 2008 at 09:05:17AM -0500, Michael Munger wrote:
Two questions:
1. Does anyone have a good way to transfer a call from inside
comedian mail to the current extension? The problem is: let's say the
phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd
Did you file a bug report?
http://bugs.digium.com
-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 21, 2008 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] High CPU load after upgrading to 1.4
On
With some mods it surely did the trick
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan
Company, LLC
Sent: miércoles, 20 de febrero de 2008 01:49 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I guess someone has to say it.
Have you considered Aastra?
You can argue about quality/features/functionality but I have set up both
and the Aastra are definitely easier to configure and they reboot quicker.
Nobody ever complains about the quality of sound or speakerphone on them
either.
vi /etc/asterisk/extensions.conf
On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote:
hi,
how to write a advanced dial plan
for example:
dial to a extension(123).if the user didnot pick the call, caller should get
a ivr script(Enter 1 to to dial operator and 2 to go to
Are you using buttons on your phone to effect the transfer, or are you
using codes defined in features.conf?
Moj
Ian wrote:
Hi,
Mojo with Horan Company, LLC said the following on 20-Feb-08 09:31 PM:
Is it AFTER you have parked a call? Meaning, for example, you transfer
an incoming call
On Fri, Feb 22, 2008 at 7:56 AM, Anciso, Roy [EMAIL PROTECTED] wrote:
Hello List,
After seeing a few positive responses for the Linksys SPA-942 phones I was
hoping to get some answers on the following questions:
· How do the phones handling system wide paging? Is it similar to the
What I would like to do is have two identical *
servers which accept registrations of sip extensions
4000-4999.
If I define a rrDNS or LinuxHA then I should have
load-balanced registrations.
However, say ext. 4001 is registered on *1 and 4002 is
registered on *2, if 4001 tries to call
--- Andres Jimenez [EMAIL PROTECTED] wrote:
On Fri, Feb 22, 2008 at 11:42 AM, Vieri
[EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and
4002 is
registered on *2, if 4001 tries to call 4002 then
I
would like to do something like:
- lookup 4002 on *1, try to
Sorry, I jut got your other message stating the steps your boss'
secretary uses to transfer calls, so this question's time is past.
I'm curious if the 'flash' button is the only way those phones can do a
transfer. Do they have any other transfer keys, or could you try the
featuremap codes?
Hi,
Could some one let me know if a fax is received through a FXO card
connected to * and fax machine is connected to a Linksys FXS device
which support T38, is T38 going to be used for faxes which comes from
PSTN or go through PSTN ? or because of Asterisk T38 passthrough support
it is not
Hi
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
client so i can make VOIP calls thru that phone. Aslo that Phone easly able
to register with Asterisk Pbx to recive inter-office calls.
i try to search from web also from Nokia site but they only mention this
features
Hello All,
I have my own AGI script running and I am trying to push the call to
voice mail when Busy, Unavailable and Not Answered.
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only Temp Greetings
voice mail
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote:
--- Andres Jimenez [EMAIL PROTECTED] wrote:
On Fri, Feb 22, 2008 at 11:42 AM, Vieri
[EMAIL PROTECTED] wrote:
However, say ext. 4001 is registered on *1 and
4002 is
registered on *2, if 4001 tries to call 4002
On 2/22/08, amit salunkhe [EMAIL PROTECTED] wrote:
Hi
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
client so i can make VOIP calls thru that phone. Aslo that Phone easly able
to register with Asterisk Pbx to recive inter-office calls.
i try to search from web
On Fri, Feb 22, 2008 at 5:49 PM, Vieri [EMAIL PROTECTED] wrote:
Thanks. I'll try that although I hope it won't go into
an infinite loop between the 2 servers.
You are right. That could happen if the phone is not registered anywhere
You can put some security in the dialplan.
if calls
Nitesh Divecha wrote:
([EMAIL PROTECTED]|b)
Any suggestions... By the way I am running Asterisk 1.2.18
I believe under 1.2.x it would be [EMAIL PROTECTED]
One of my older dial plans lists:
s-BUSY,1,Voicemail([EMAIL PROTECTED])
Doug
--
Ben Franklin quote:
Those who would give up
On Sat, 2008-02-23 at 00:03 +0530, amit salunkhe wrote:
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP
Calling client so i can make VOIP calls thru that phone. Aslo that
Phone easly able to register with Asterisk Pbx to recive inter-office
calls. i try to search from
On Sat, 23 Feb 2008, amit salunkhe wrote:
Hi
i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling
client so i can make VOIP calls thru that phone. Aslo that Phone easly able
to register with Asterisk Pbx to recive inter-office calls.
i try to search from web also from
All,
I am setting up asterisk on a nslu2 (Linksys) using unslug.
Everything is working great except that I have a polycom 301 and I
cannot get the message indicator to work. I have created the users and
mailbox in users.conf and I can manually dial the mailbox (*986000).
Last thing is I
Trying to figure out how to prefer voip traffic on a dsl line.
Found a great howto:
http://www.howtoforge.com/voip_qos_traffic_shaping_iproute2_asterisk
but I'm trying to figure out the relationship between the tos of
iax.conf and tos of tc from Iproute2. my traffic goes from my linux
router
* How do the phones handling system wide paging? Is it similar to the
Polycom phones?
No idea I'm afraid, none of our clients use paging functionality.
* Can a corporate directory be configured with the phones using
Asterisk?
Yes and no. You can set up a directory on a
Cavanna, Richard wrote:
All,
I am setting up asterisk on a nslu2 (Linksys) using unslug.
Everything is working great except that I have a polycom 301 and I
cannot get the message indicator to work. I have created the users and
mailbox in users.conf and I can manually dial the mailbox
I know how to remap a key on a polycom 301 and 501
But does anyone know of a list of mapping keys?
For example, the Do Not Disturb on a 301 is #23. I got that one by just
guessing though.
Thanks,
Rob
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Jay Milk wrote:
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)
Updated to 1.2.1 to fix an
Olivier wrote:
Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI
ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be
made out of 2 B410P ?
No, the card does not support that mode.
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The
It's time to ask this question again. Maybe I will get a reply one day. :)
Asterisk
1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip
time, etc.
Since there's only one set of variables, and calls will
have
Gordon Henderson wrote:
What about the need for 1.4 at all sites? Is it sufficient to just have it
in the man in the middle site?
It uses new IAX2 commands, so it requires that all three endpoints
understand those commands. At this time the only IAX2 implementations
that I am aware of that
Fons van der Beek wrote:
Because i want a ringing signal while people are in a waiting queue i've
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but
when i make an external call, no signal is heard.
everything
I tried that, its gives me the same problem.
Kevin P. Fleming schreef:
Fons van der Beek wrote:
Because i want a ringing signal while people are in a waiting queue i've
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is
Edwin,
I feel your pain. I struggled getting fax to work reliably with both
1.2 and 1.4 versions. Any combination I tried, usually caused a crash.
I recently upgraded to 1.6.beta4 and installed the app_fax from the
addons installation and it worked first time out of the box :-)
I've
I have absolutely no idea since I was not even aware of it. However,
this may give you some hints as to where you can find more information:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg27124.html
- Waldo
On Feb 22, 2008, at 5:08 PM, Douglas Garstang wrote:
It's time to ask this
That can be found in the monstrous admin guide for the phone, seemly in
Section 3.1.7 in my ancient version 1.5.0 document. It shows me that on
the 501, that button is 9 instead of 23.
http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip301.html
There's a link to the
Hi,
I am looking for a did from Saltillo Mexico.
Any pointers?
robert
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To UNSUBSCRIBE or update options visit:
So far I've never run into anything that's even /close/ to the
speakerphone quality of the Polycoms. There's no comparison on the
speakerphone between the Linksys phones and the Polycoms - it's chalk
and cheese, but by the same token that holds true for just about every
other phone too.
This problem would happen if you did not have /etc/asterisk/indications.conf
Fons van der Beek wrote:
I tried that, its gives me the same problem.
Kevin P. Fleming schreef:
Fons van der Beek wrote:
Because i want a ringing signal while people are in a waiting queue i've
created a wav
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding a fax call to T.38.
The only real option available at the moment
Tnx.
I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options
So i am still puzzled
my extensions.conf in respect to incomming calls (as basic as possible)
exten = s,1,Answer
exten = s,2,queue(receptie|r)
exten = s,3,Voicemail(201)
everything else works
I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of
voicemail messages into MySQL. It is my understanding that I can only do this
via ODBC. I installed per
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
unixODBC unixODBC-devel
On Friday 22 February 2008 18:28:56 Mike Hammett wrote:
--snip--
[asterisk]
enabled = no
dsn = asterisk
;username = myuser
;password = mypass
pre-connect = yes
--snip--
WARNING[21068]: app_voicemail.c:2233 inboxcount: Failed to obtain database
object for 'asterisk'!
What does enabled
Don't answer the line. Also try using the US indications, just in case
something odd is in the NL setup.
Fons van der Beek wrote:
Tnx.
I checked /etc/asterisk/indications.conf and my default location nl
is listed in the options
So i am still puzzled
my extensions.conf in respect to
Replying to my own post. Asterisk uses indications.conf when it has to
provide tones AFTER the line is answered. You might get a message on
the console like Unable to handle indication 15 or something like that.
Eric Wieling wrote:
Don't answer the line. Also try using the US indications,
it's very odd
-I just upgraded to 1.4.18 (from 1.4.17)
-removed answer
-changed to several other options, still no luck
(restarted also)
Eric Wieling schreef:
Don't answer the line. Also try using the US indications, just in case
something odd is in the NL setup.
Fons van der Beek wrote:
NOT answering did the trick!
Tnx a lot! now it works like it should work!
Eric Wieling schreef:
Replying to my own post. Asterisk uses indications.conf when it has to
provide tones AFTER the line is answered. You might get a message on
the console like Unable to handle indication 15 or
It was my understanding that voicemail.conf referenced MySQL and not
asterisk.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
If not answering fixes the problem then the issue is indications.conf.
Try using the indications.conf.sample file included with the Asterisk
source code, then stop Asterisk and starting it again. I do not know if
indications.conf is reloaded on a reload.
Fons van der Beek wrote:
NOT
I tried to use DUNDi on my local servers but I can't
seem to make it work. Most howtos out there explain
the use of DUNDi when the extension ranges do not
overlap.
The following doc describes using the same extensions across multiple *
servers. It requires using realtime, but seems to do
Thanks Doug,
I tried that but it didn't work either... As per Wiki
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it
has a statement that starting from 1.4-trunk FLAG must be pass using a
pipe sign '|'.
I have other Asterisk 1.2 running with FreePBX and I went over the
I have an E61i and it works great with my Asterisk. No
extra software needed, everything is built into those phones.
Rajeev.
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Hello,
I've one nokia E65 that works very well with my asterisk box.
The people here don't let me even try it as they are afraid it will consume the
battery more than when it is used the usual way. Is this true?
Thanks, __Yehavi:
Hi, I have some experience with Asterisk. What I would like to know is, are
there any programmable APIs that we can use to get the information monitored
by asterisk.
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Nitesh Divecha wrote:
Everything is working fine but the only problem is voice mail greetings
for Busy and Unavailable is not played. By default only Temp Greetings
voice mail greetings is played. I am passing the correct parameters for
Busy = 'b', Unavailable = 'u' and default goes to Not
Rob Hillis wrote:
Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk
has had passthrough support for T.38 for a while (somewhere in 1.4 it
became available IIRC) but is currently completely incapable of
terminating or encoding a fax call to T.38.
I thought * was
I've overwritten the indications.conf with the one from the sourcecode,
stil no luck
Perhaps somebody knows what the correct value for indications.conf is
when using the dutch xs4all as sip carrier??
and even with verbose set to 114 (quite big) there are no errormessages
indicating that
Hi Robert,
DID World Wide has coverage for Saltillo (please see
http://www.didww.com/virtual_numbers/Mexico), with flat-rate forwarding to
PSTN, VoIP, SIP, H.323, IAX, Skype, MSN or Google Talk.
Regards,
Gideon
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri,
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