On Wed, 2003-10-01 at 07:46, Steven Critchfield wrote:
> Or if you wish to write closed source apps, you purchase licenses.
> Nothing in life is free. Open source software has strings attached, only
> these strings don't go to your wallet.
I've never heard it expressed so beautifully.
> Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as
> the 102D. And to make matters worse he starts the bid at $90.00 Beware.
There's no need to beware -- anyone who doesn't shop around deserves to get
suckered.
Regards,
Andrew
You can create a "web browser" like app with Visual Basic and the
internet explorer rendering engine. Fool them Johnny
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of CW_ASN
> Sent: Wednesday, October 01, 2003 10:16 PM
> To: [EM
You mentioned that Asterisk is programmed using C. Can apps be written
in C++ also or only C.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Posted At: Wednesday, October 01, 2003 10:23 PM
Posted To: Asterisk User Group
Convers
On Wed, 2003-10-01 at 21:08, PBX wrote:
> I have a general question. Can the voicemail app be rewritten? If so,
> is it written in C / C++ now?
Cudos for starting a new thread. Bummer you used HTML.
Yes voicemail could be rewritten, this is the beauty of open source
software. It is written in C l
> Jamie, I'm sorry if you took that as a insult about VB programmers. I
> was getting ahead of an argument that didn't materialize. I have coded
> in VB also. But the comment stands that if the language was chosen
> specifically to ease getting developers, then the developer pool was
> also shrunk
>
> If this is the case, obviously you need more experience in
> creating user friendly web frontends. Some that I have
> seen are brilliant and look great on any OS. PHP is best
> suited to what we are trying to do, because it's powerful,
> it has all the feature we want to use and it runs on th
Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D.
And to make matters worse he starts the bid at $90.00 Beware.
http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch=
--
Costas Menico
Meezon Software Corp
201-224-81
On Wed, 2003-10-01 at 18:58, Jamie Carl wrote:
> Steven, i resent the discrimination of VB programmers,
> especially the comment: "those types of programmers are
> unlikely to use linux in any fashion". I've been writing
> in VB for 8 years and it was the first language I learnt.
> I still us
Ok, so this seems to be in several threads:
http://lists.digium.com/pipermail/asterisk-users/2003-October/022328.html
I've not trimmed this message on purpose to have a collection of
thoughts, further in this thread, please trim and refer to the thread.
I think it's safe to say that a PHP based sy
I have a general question.
Can the voicemail app be rewritten?
If so, is it written in C / C++ now?
Thank you
Geoff
Its not fixed. I still have the issues with fwd using the latest cvs
On Wed, 1 Oct 2003, John Todd wrote:
> >Can anyone confirm that the SIP updates in CVS have fixed the channel
> >leakage and the codec negotiation problem that was happening a few days
> >ago?
> >
> >Thanks
> >dave
> >
> >
> >-
Ok, is it time for my comments on all of this?
On Wed, 01 Oct 2003 16:56:22 -0500
Steven Critchfield <[EMAIL PROTECTED]> wrote:
On Wed, 2003-10-01 at 16:29, CW_ASN wrote:
Yes, I know... but some people uses Windows and hates
web frontends (some
customers, for example)...
I hate Windows platfor
> Then please reread the whole of my argument. I was arguing only against
> a front end that couldn't be used on other platforms also. This is why I
> was so down on VB, it is only usable on Windows. I was mildly down on
> .net stuff only because of the potential to have the mono rug ripped out
> f
Should have, yes.
Mark
On Wed, 1 Oct 2003, Dave Weis wrote:
>
> Can anyone confirm that the SIP updates in CVS have fixed the channel
> leakage and the codec negotiation problem that was happening a few days
> ago?
>
> Thanks
> dave
>
>
> --
> Dave Weis "I believe there are more inst
> On Wed, 2003-10-01 at 16:40, James Sharp wrote:
>> I've got a handful of T1s going into a TE410. When I place calls into
>> the
>> system over these T1s, the system either doesn't decode all of the DTMF
>> digits or it decodes ones that aren't there.
>>
>> When the system places calls out, there
Can anyone confirm that the SIP updates in CVS have fixed the channel
leakage and the codec negotiation problem that was happening a few days
ago?
Thanks
dave
--
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual a
On Tue, 30 Sep 2003, WipeOut wrote:
Whats the default SPEEX bitrate set to in Asterisk?
The default bitrate for speex (at this time determined by the speex lib
because we don't explicitly set it) is 15k
I'm still looking for a good way to implement options for codecs so we can
modify these setti
On Wed, 2003-10-01 at 16:40, James Sharp wrote:
> I've got a handful of T1s going into a TE410. When I place calls into the
> system over these T1s, the system either doesn't decode all of the DTMF
> digits or it decodes ones that aren't there.
>
> When the system places calls out, there is no pr
On Wed, 2003-10-01 at 16:29, CW_ASN wrote:
> Yes, I know... but some people uses Windows and hates web frontends (some
> customers, for example)...
> I hate Windows platforms, but I'm only a technician...
Then please reread the whole of my argument. I was arguing only against
a front end that coul
Can anyone confirm that the SIP updates in CVS have fixed the channel
leakage and the codec negotiation problem that was happening a few days
ago?
Thanks
dave
--
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gr
I've got a handful of T1s going into a TE410. When I place calls into the
system over these T1s, the system either doesn't decode all of the DTMF
digits or it decodes ones that aren't there.
When the system places calls out, there is no problem doing the DTMF
detection. Everything works great.
Yes, I know... but some people uses Windows and hates web frontends (some
customers, for example)...
I hate Windows platforms, but I'm only a technician...
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 01, 2003 5:56 PM
> ;
> ; Music on hold class definitions
> ;
> [classes]
> default => quietmp3:/var/lib/asterisk/mohmp3
> ;loud => mp3:/var/lib/asterisk/mohmp3
> ;random => quietmp3:/var/lib/asterisk/mohmp3,-z
Hi, Consider me the most embarrased newbie _ever_. It was as simple as
whitespace: I had 'mohmp3, -z',
I'm in the middle of that, using VB6.
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 01, 2003 5:38 PM
Subject: Re: [Asterisk-Users] CDR Web Search Frontend
> Do we have any Visual Basic or .NET programmers out there? The reason I
> say th
How about doing it in Java with Java web services(for conf file generation
and user authentication,e t c) so that way other developers can write in
other languages such as PHP or perl different front-ends but consume the web
services over http.
Well this is the way I plan on doing it so the code
On Wed, 2003-10-01 at 15:38, [EMAIL PROTECTED] wrote:
> Do we have any Visual Basic or .NET programmers out there? The reason I
> say that is this. I would suggest that a program or application be put
No reason to put it in a dead end language like VB. At least you might
be able to use something i
[EMAIL PROTECTED] wrote:
Do we have any Visual Basic or .NET programmers out there? The reason I
say that is this. I would suggest that a program or application be put
together with Visual basic or .NET that will generate the config files for
you by translating what you want to do in telephony ter
Ernest,
There is a beta load that you can get from the Audiocodes dealer which
is working for us.
We are using their 4-port MP-104 SIP gateway and the only problems we
have with it
are:
1. Outgoing calls go out to the lines in a round-robin fashion. You can
put any number of the
lines in
Do we have any Visual Basic or .NET programmers out there? The reason I
say that is this. I would suggest that a program or application be put
together with Visual basic or .NET that will generate the config files for
you by translating what you want to do in telephony terms/jargon to
asterisk cofi
Hi,
Anybody seen this error?
Getting an odd error on the console when I place a call from console to
a SIP station. As soon as the station answers, here is the error.
SIP/172.16.10.24-527b answered OSS/dsp
WARNING[1298960704]: File chan_oss.c, Line 679 (oss_indicate): Don't
know how to display
Please send your musiconhold.conf
Or try this:
;
; Music on hold class definitions
;
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
;loud => mp3:/var/lib/asterisk/mohmp3
;random => quietmp3:/var/lib/asterisk/mohmp3,-z
- Original Message -
From: "Toby Seaman" <[EMAIL PROTECTED]>
Just the sort of newbie question we all hate ;-)
I'm a bit stuck with MOH. I think all is done right and I've read
everyhing I can find, but whenever * tries to do MOH, all that happens
is
'-z: No such file or directory'
Yes, I am on redHat. Yes I have installed real mpg123. Yes, it does
se
Please check cvs. See into libr2.
The README file says that only China and Argentina are available (I'm a
lucky guy, I'm in Argentina :) ).
Regards,
Gus
- Original Message -
From: "Musaluke AK" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 01, 2003 12:36 PM
Subje
Is anyone on the list using an Audiocodes gateway with asterisk and SIP?
I'm looking at that platform, but I have a couple of issues:
1) Echo cancellation. The echo that I'm hearing with an X100P is
unacceptable. Does the Audiocodes do better?
2) Line signalling. I'm using Kewlstart with the X10
And what about # tarnsfer on asterisk. I still
cannot make it work.
But If i make it work, then can I make attended
transfer. (I am not talking about call parking)
-- Bart
- Original Message -
From:
T Aksoy
To: [EMAIL PROTECTED]
Sent: Wednesday, October 01, 2003 4:
I have Cisco 79xx phones on the desktop here, which are capable of
alpha input through the numeric keypad. I'd like to place calls to
fully-qualified SIP addresses ([EMAIL PROTECTED]) but it seems that
Asterisk is somehow stripping the @foo.edu part of the request off
when the entry hits my di
An update on this issue:
It seems to be an echo cancellation problem. My setup is as following:
FXO -> IAX2 server 1 - IAX2 server 2 <- FXO
1st call has a terrible echo. If I start server 2 and immediately issue zap
destroy channel X (one of the 4 fxos), than the call goes through without a
pr
Only
blind transfer I think. Attended transfer can be sort of be done using asterisk
call parking feature.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Bartosz
JozwiakSent: 01 October 2003 20:04To: ASTERISK
USERSSubject: [Asterisk-Users
Hello,
I am thinking about buying grandstream phones and I
have following question.
Does granstream ip phone support attendend
transer and blind transfer?
-- Bart
Hi George -- I think the answer is going to depend upon your extention
situation -- how many extensions do you plan on having?
- Jeff Dodge
- Original Message -
From: "George L. Carden III" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 01, 2003 4:2
Dave Weis wrote:
On Wed, 1 Oct 2003, Lists wrote:
Does anyone out there use Asterisk with voip(sip or iax) long distance
provider?
Care to share about your experiances doing this?
I'm using nufone.net. I just paypal'ed him some money, he sent me the
configuration changes and password info, and
We can do that in the UK and in Poland and soon in more countries, but since
you are US based I would recommend rather to talk to Jeremy at Nufone.
On Wednesday 01 October 2003 8:30 pm, David Harris wrote:
> Are there any sip providers out there providing full business telephone
> service. Not
Are there any sip providers out there providing full
business telephone service. Not
just single line/residential service like I have seen with vonage etc.
For example take a company currently using a legacy pbx
connected to the PSTN with a PRI. I
would like to replace this setup wi
Hello all
I new to this list and would like to ask a few questions with the following
assumtions.
1. im iggy to astrick
2. i havent made any dec as to what softswitch i will use
I am curious as to the quality of this asterisk softswitch and what success
or failure stories on deploying asterisk,
Take a look at the "switch" app..
Also search the archives for switch..
Thank you, I've missed that function.
Found an example in the archives and in the sample extensions.conf in the distribution.
Updated the wiki ;-)
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
I am trying to optimize echo cancellation. Originally (with only one
phone attached), echo cancellation worked well. Echo was only obvious
in the first second or so of a call. Now, with multiple phones on the
system, the echo does not go away. It is quiet, but audible and somewhat
annoying.
I was looking at some fixes in the replies to the chan_sip.c problems and
I am wondering if I am seeing the same thing in the earlier file version. I
just checked to see that my chan_sip.c is version 1.179 when I did my
checkout so I never had the later versions. The problem that I am seeing
is t
Here's a good example: wiki? there is a wiki? :-) I saw a broken link
once but heard no more...does it still exist?
Oh yes, there is a Wiki.
And there's a lot of how to's and software documentation up there.
If you search the archives, I've mentioned the correct URL so often
that people propably a
Thank you Greg,
I have fogotten to say that I can call my MGCP phone from my SIP phone
but mys MGCP Phone can't place a call.
Here is some parts of my extensions.conf:
[general]
static=yes
writeprotect=yes
[globals]
dandre => sip/p-dan.phone.iris-tech.fr
swiss1 => mgcp/aaln/[EMAIL PROTECTED
You need to specify what port and address its listening on and please
include your extensions.conf exten lines to use mgcp
/etc/asterisk/mgcp.conf
[general]
port = 2427
binaddr=192.168.0.1 ; if this is your asterisk box' ip
-Greg
- Original Message -
From: "Daniel ANDRE" <[EMAIL PROTE
Rich Adamson wrote:
Does anyone know if there is a zapata.conf option to tell * to
listen for a dialtone before dialing?
I've got a couple of analog phones on a pstn line shared with a
x100p * fx line. If someone is on the analog phone and another
person initiates a call through * to use the sam
On Wed, 2003-10-01 at 10:23, Paulo H. Mannheimer wrote:
> No interrupts are being shared ;-(
>
> take a look
>
[snip]
My guess is that four Digium cards is probably too many for one box to
handle... they are probably overloading your system with interrupts.
(I'm just guessing at this point, b
Does anyone know if there is a zapata.conf option to tell * to
listen for a dialtone before dialing?
I've got a couple of analog phones on a pstn line shared with a
x100p * fx line. If someone is on the analog phone and another
person initiates a call through * to use the same line, * dials
over
No interrupts are being shared ;-(
take a look
CPU0
0: 155590 XT-PIC timer
1: 3 XT-PIC keyboard
2: 0 XT-PIC cascade
3:1534436 XT-PIC wcfxo
4: 49926 XT-PIC serial
7:1534528 XT-PIC
On Wed, 2003-10-01 at 06:32, Paulo Mannheimer wrote:
> Hi folks,
>
> I'm still having the following problem, maybe someone can help me out of
> it.
>
> Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
> communicate through IAX2. Everything works ok on machine 1. On machine
> 2, if I
On Wed, 2003-10-01 at 07:55, Andreas Anderson wrote:
> Hi *,
>
> does someone has a directory that works with the Cisco 7960
> and astdb or mysql/ldap?
>
I have one that connects to mysql... contact me off-list (or try to find
me on the IRC channel as jsmith) and I'll pass it along.
Jared Smith
On Wed, 2003-10-01 at 10:41, rjrae wrote:
> Could use some assistance please
>
> I have failed to get past the cisco gauntlet and am prevented
> from accessing the 7940 firmware required to get SIP on the phones I
> purchased on ebay. I would be happy to buy the cisco support but
Could use some assistance please
I have failed to get past the cisco gauntlet and am prevented
from accessing the 7940 firmware required to get SIP on the phones I
purchased on ebay. I would be happy to buy the cisco support but they tell
me I have to be IP-tel certified first, th
hello,
I have been searching the mailing list for info on R2 signalling support
in *
The best I came up with was a post by Steve as far back as 18 Jul 2003.
This post indicates that some work was being done on it. Has anyone got
any info if the current cvs does have support for R2 on E100P as
I read in the message archives that call Park may not be supported using
sip phones. Can this be true?
I can not transfer a call to the call park extension from a SIP phone.
I can from the Zaptel FXS. Both phones are in the same context?
Any Ideas?
Kevin
__
In reading through the message archives, I read that only Nufone allows
for setting the caller ID for the outgoing SIP call. The problem I have
is when calling a number that has anonymous call blocking in effect, I
hear ring back tone but in the call never connects.
Has anyone else experience thi
On Wed, 1 Oct 2003, Nick Knight wrote:
> Hello all,
>
> I am sure that this is possible - for helpdesk envioroments i.e. when
> you hear an announcment "Your call may be recorded for quality purposes"
> can asterisk record all calls onto disk or similar - hopefully as MP3?
The Monitor app will re
yes, that is correct.
I just realised that myself.
Thanks
Senad
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Hello,
Sorry for posting again my question about MGCP Phone and Asterisk But I
can't use it.
I'd like to know weather it is a pb of my confiuration (mgcp.conf), My
IP Phone device or asterisk.
I include my mgcp.conf file and may send some debug trace.
Thank you for any feedback.
Best regards,
Senad Jordanovic wrote:
* has application called "Record".
type "show appliaction record" for more info at CLI prompt.
Senad
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"Record" is not t
* has application called "Record".
type "show appliaction record" for more info at CLI prompt.
Senad
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great, thanks for that.
What if interface triggers CLI commands?
Senad
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Nick Knight wrote:
Hello all,
I am sure that this is possible - for helpdesk envioroments i.e. when
you hear an announcment "Your call may be recorded for quality purposes"
can asterisk record all calls onto disk or similar - hopefully as MP3?
Thanks
Nick
Matt,
It's done by using the switch keyword in extensions.conf
Thus if you fill in the stuff below correctly and make
the appropriate settings in iax.conf:
switch => IAX/username:[EMAIL PROTECTED]/context
Will send all extensions which cannot be resolved in the local dialplan,
over IAX to the
> If a web interface (similar to vonage account management) gets produced
> using PHP/MYSQL to administer
> *, does that require licence from Digium if the code is not open source.
If it merely manipulates Asterisk's config files and in no way links to
Asterisk itself, then the choice of licenses
Hello all,
I am sure that this is possible - for helpdesk envioroments i.e. when
you hear an announcment "Your call may be recorded for quality purposes"
can asterisk record all calls onto disk or similar - hopefully as MP3?
Thanks
Nick
___
Asterisk-Us
mattf wrote:
I have heard it mentioned several times by different people but can anyone
explain to me how you can set up a single dialplan for 2 or more than
asterisk boxes located on the same local network?
MATT---
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[EMAIL
> >The way i see it the webtool needs to be something that puts the power of
> >asterisk config to non config file 'freaks' :)
I totally agree, while I don't know any PHP it seems like you're
starting down a track much more like what I was thinking, a menu system,
not a condensed text editor.
>
I can not transfer a call to the call park extension from a SIP phone.
I can from the Zaptel FXS. Both phones are in the same context?
Any Ideas?
Kevin
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If anyone is integrating Dialogic hardware with Asterisk, we have (22)
Dialogic D/240SC-T1 REV2 voice boards available for immediate sale. These
are UDD Tested pulls in perfect condition with (1) Year Warranty. Asking
$800/ea
Cory Andrews
*
b2 Technologies
454 Sonwill Drive
Buffa
I have heard it mentioned several times by different people but can anyone
explain to me how you can set up a single dialplan for 2 or more than
asterisk boxes located on the same local network?
MATT---
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h
Hi *,
does someone has a directory that works with the Cisco 7960
and astdb or mysql/ldap?
Regards,
Andreas
_
Gaming galore at http://xtramsn.co.nz/gaming !
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On Wed, 1 Oct 2003, Lists wrote:
> Does anyone out there use Asterisk with voip(sip or iax) long distance
> provider?
> Care to share about your experiances doing this?
I'm using nufone.net. I just paypal'ed him some money, he sent me the
configuration changes and password info, and it worked.
When I place a call on hold after dialing with a SIP LD provider, the
called party hears no music. When I originate the call with the ZAP FXO
the music works. Any suggestions?
Kevin
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Does anyone out there use Asterisk with voip(sip or iax) long distance
provider?
Care to share about your experiances doing this?
Michael
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On Wed, 2003-10-01 at 07:56, costas wrote:
> >From the discussion thread, you will notice that the only real answer
> is to get a lawyers opinion. And even there the final answer is the
> judge. So hope for the best and be prepared for the worst.
Or if you wish to write closed source apps, you pur
Subject pretty much hits it.
Can someone explain the difference?
Is global var / db similar? They effect all channels.
db lasts through restarts and vars are gone?
And Var is per channel?
John
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Is anyone using nikotel with asterisk? When I attempt to place a call,
I get "Everyone is busy at this time".
Thanks,
Kevin
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I agree on the aspects of documentation.
However, my inset is that most of the basic functions (e.g. an example
dialplan) can be found on the net.
Since I am interested in building a powerful and usable VoIP gateway, I
would like to see more of the subtler and not so often exposed functions
buil
- Original Message -
From: Roderick Montgomery <[EMAIL PROTECTED]>
Date: Wednesday, October 1, 2003 10:46 pm
Subject: Re: [Asterisk-Users] Asterisk Documentation
> ... ...
>
> integration with basic scripting languages, I'm not focusing at all on
> documenting Asterisk code development. I
>From the discussion thread, you will notice that the only real answer is to get a
>lawyers opinion. And even there the final answer is the judge. So hope for the best
>and be prepared for the worst.
It seems no one can agree on this. And even a lawyer would probably be confused. Why?
Because i
According to [EMAIL PROTECTED]:
>
> Does anybody have any thoughts on this plan, or better ideas?
> Negative/positive thoughts about Doxygen?
> Most importantly, is anyone else working on something along these lines
> already?
Daniel,
Documentation is much-needed, and I'm glad to hear that there
Hi folks,
I'm still having the following problem, maybe someone can help me out of
it.
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
communicate through IAX2. Everything works ok on machine 1. On machine
2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I
manual
Hello,
Could somebody tell me what I should change in
iax.conf file to be able to receive calls from iaxtel.
I am already registered and I can make calls to
IAXtel users but what I should do in iax.conf
to be able to receive call also.
-- Bart
I have setup Asterisk to work with a SIP gateway, some SIP phones
and the Digium FXS/FXO development card combo on another *
box with pretty good results so far. Here are a couple of questions
that I have that wasn't obvious from the documentation:
Voicemail vs Voicemail2 - What is the major diffe
I've spent the last couple of days learning Doxygen and getting at least
basically familiar with the Asterisk source code. I'm starting to write
up comments for Doxygen to generate API docs from, and I've also started
looking at ways to use Doxygen to generate a configuration reference with
config
Hi, I thought you were using * and was wondering which kind of PC server
you used to compress 120 voice channels.
Yes I have a working * (1xE1 PRI + analog)
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul Hakeem
Sent: Wednesday, October 01, 2
Tjardick van der Kraan wrote:
While it seems this discussion is taking on here, i'll just jump in myself.
A couple of weeks ago i started playing with the idea of a webadmin, but one
that is bit different then the phpconfig there is now. Allthough i think
it's a good thing, don't get me wrong, i
While it seems this discussion is taking on here, i'll just jump in myself.
A couple of weeks ago i started playing with the idea of a webadmin, but one
that is bit different then the phpconfig there is now. Allthough i think
it's a good thing, don't get me wrong, i myself had something else in mi
I am not a coder hence this question:
If a web interface (similar to vonage account management) gets produced
using PHP/MYSQL to administer
*, does that require licence from Digium if the code is not open source.
Thanks...
Senad
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Asterisk-Users mai
Hi,
I used Cisco 3640 with 2xNM-HDV-2E1 cards.
The default GW router has RTP and TCP/UDP header compressions.
There is also a Linux solution for this. You can run RTP compression on
your asterisk box, and or run UDP/TCP header compression on the default
GW router.
Do you have a working * box at the
Thank you for example, wasim.
I'm trying to get something like this:
1. incoming call (Zap) comes to *, person 1 answers
2. person 1 cant help so s/he put caller to hold
3. person 1 calls to person 2 and tells him the situation
4. call from hold is redirected to person 2 and person 1 get out of th
Kevin wrote:
When I dial with my Grandstream 101 telephone to another sip phone or
Zap FXS, the call rings, but no audio is passed. Eventually the call
gets disconnected. The same thing happens if I dial the Grandstream.
Any Suggestions?
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Asteri
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