I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the
file from * to the FAX under HT488(firmware 1.0.1.2).
My OS is 2.6.11-1.27_FC3smp, CVS-v1-0-04/20/05 with ztdummy.
spandsp is fun!
I made a file:
Channel: SIP/4881
MaxRetries: 0
WaitTime: 20
Application: txfax
Data: /usr/home/z
Hello all.
How to compile chan_unicall.c
i have problem when compile chan_unicall.c, error message
please help
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DAS
Hi:
Please help me understand how a remote server with IP
address define a Asterisk client behind a nat.
Thanks.
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I've configured SIP softphone to work with asterisk n it's working fine. but i m unable to connect my netphone IP phone I've connected my phone to LAN and assigned an IP address to it but how can i make call... plz tell me step wise.FREE pop-up blocking with the new MSN Toolbar MSN
> Only if you have your clothes on and they don't... ;-)
I should _hope_ they don't have your clothes on :)
Andrew
--
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, i
Anyone know the differences between the two?
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Clue or clueless? Your call.
Steve Underwood wrote:
Michael D Schelin wrote:
Steve, you should really test the Codec and have G729 running as a
pure IP to IP call you can not hear the difference on good networks!
Well, it does to anyone without hearing damage. It sounds very obviously
di
Joseph wrote:
I'll be trying AG-468 4 x FXS about 88.00USD from ATComm and let you
know when I get one (though it might be a while)
Where yo purchase it?
--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
e-mail: [EMAIL PROTECTED]
I just installed the latest cvs head and seems a lot of commands haven been
depricated.
Where can I see the changes on all cvs head versions in order to keep up
with the changes needed on my side.
I checked the wiki and it still shows all the old commands and no mentions
about the changes.
I'm having some trouble with Polycom Soundpoint phones. I have had good luck
deploying them on a local network, but now I've tried putting some in place
which access their * server across the network.
The * server is on a public IP and the polycoms are behind a NAT on a cable
modem broadband conn
Nguyen Trung Tin wrote:
Hello All
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling
for Vietnam
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?
libr2 is a piece of useless junk, which I have asked Digium to remove.
The software at soft-switch.
Hello All
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling for Vietnam
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?
how to compile chan_unicall.c on asterisk. asterisk update CVS-head- May 27 2005.
Thanks
I don't use mpg123, I use madplay
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Viernes, 27 de Mayo de 2005 07:41 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Asterisk stoppin
http://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=288160
Thx__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
The first problem I would consider is the reliability of the system.
Currently, I assume you are using standard analog lines stretched
between your various buildings, which means you 'only' rely on power at
the central location (which you say is generator provided) and the
physical cable survives.
Not sure about E1's but we are paying $1200 per month for a PRI. Get
this, for a PRI without Caller ID services its $950...
Free incoming calls and free local calling.
I think that is really high, but I don't think I have a whole lot of
choices here in central Illinois with Consolidated Communic
On Fri, 2005-05-27 at 20:44 +0200, Michiel van Baak wrote:
> This can be used on client pc's that are configured to allow
> external apps to be installed/executed.
> What if you are on a terminal in a hotel or fair or
> whatever? The solution I made is intended to be used in a
> webbased CRM/Groupw
How does one process / capture a timeout that has happened in using an AGI
script.. Preferably PHP.
I know you can set the wait timeout for a certain time, but how does the
script continue?
Thanks
Ben
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Michael D Schelin wrote:
Steve, you should really test the Codec and have G729 running as a
pure IP to IP call you can not hear the difference on good networks!
Well, it does to anyone without hearing damage. It sounds very obviously
different.
Please do not get me wrong that G711u sounds
In .SG, the basic monthly recurring cost is S$360 (roughly US$200)
excluding outgoing per minute charges (at S$0.014 peak - that's one
point four cents). The one time installation charge is S$2000.
FYI.
X - Filter wrote:
Hello List,
I'd like to ask how much you guys pay for an E1 (30 voice
Hi Moises, I made some changes, but the idea is the same. This is my
extensions.conf file, my extension is the 403 and I'm trying to
configure it:
[context]
...
[ask_voice_mail_no_answer]
exten => s,1,Background(presione_1_para_voicemail)
exten => 1,1,Voicemail(u${voicemail})
exten => 1,2,Hangup(
Has anyone tried or had success statically linking Asterisk? I'd like
to do this to deploy to smaller boxes that don't have the toolchain
and libraries to build the thing.
I've tried using statifier (at sf.net) which claims to take a dynamic
executable and transform it into a static executable. I
> Guys.
>
> Anybody having problems with asterisk taking all the cou and then not
> responding anymore without a reboot?
>
> How can I diagnose what asterisk is doing and why is it taking all the cpu?
>
> Hope you can help...
9 times out of 10 its not asterisk, but mpg123.
Try stopping asteri
Do you have any link? Isn't it PA-1688 Chip?
Isamar
On Fri, 27 May 2005, Waldo Rubinstein wrote:
> Has anyone had any experience with the Soyo G688 phone? I'd like to
> use it as a agent's phone. Is it reliable? How well does it work with
> *? How's the quality? Features?
>
> Thanks,
> Waldo
>
Guys.
Anybody having problems with asterisk taking all the cou and then not
responding anymore without a reboot?
How can I diagnose what asterisk is doing and why is it taking all the cpu?
Hope you can help...
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Asterisk-
They find the users but not the extension.
Have a look at
http://www.asteriskguru.com/tutorials/xlite_softphone.html for a
complete configuration guide.
http://www.asteriskguru.com/tutorials/softphones.html is the first page.
zoa
Romain Barrallon wrote:
Hi all,
I'm working on an implementat
On 5/28/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> Could you configure your "normal ring" to be recognized as a distinctive
> ring and go into a different context? That would essentially allow you
> to distinguish between the calls.
Excellent suggestion Jay! Thanks!! I changed the default context
+ o - va asi !!!
if (metho == INVITE ){
rewriteuriport(192.168.45.12: 5061)
forward(192.168.45.12,5061)
}
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Hi jeromy,
They are codec's problem?
How are you configured the files:
extensions.conf
oh323.conf
sip.conf
??
Rafael
Mensaje citado por Jeromy Grimmett <[EMAIL PROTECTED]>:
>
> May 28 01:58:32 WARNING[6821]: chan_sip.c:1830 sip_write: Asked to transmit
> frame type 256, while native formats i
> I just got through trying to set up a Sipura 3000 and am still looking
> for answers. There is a low volume problem (caller is underwater) on
> the FXO port that I wish someone would have told me about and I would
> have gone the other route. (even after upgrading firmware and
> adjusting gai
SOLVED!
I removed the overlapdial=yes from the Zapata.conf
Issue is resolved.
When a call comes into the box, the PRI Channel "Actual Hook
State is set to OFF HOOK" and thus the echo cancellation routines work
as directed with the echocancel=XXX
Ron
PS> I will write up
Hi all,
I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address
Could you configure your "normal ring" to be recognized as a distinctive
ring and go into a different context? That would essentially allow you
to distinguish between the calls.
> -Original Message-
> From: Gonzalo Servat [mailto:[EMAIL PROTECTED]
> Sent: Friday, May 27, 2005 3:34 PM
> T
We've been talking about
moving from our NBX100 to an asterisk solution for some time now.
Well, to make a long
story short, the call processor on our NBX has went south. We are now
forced to purchase a new call processor >$500 or move our solution to asterisk.
(Which is up and runnin
We too are a carrier / clec and our lucent iMerge for the PTSN is all
g711-ulaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Friday, May 27, 2005 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [A
Hi All,
I've recently got a "second" number installed on my PSTN line,
trusting the Asterisk distinctive ring detection would work as
expected. It appeared to work fine at the start, as the second number
generated a different ring pattern to 0,0,0 (in the console) only to
realise that almost every
Hi,
I am having problems with callerid name and the time with my
dvg-1120S. Every time I receive a call, it reverts the phone to
January 1st 12:00am. I've looked everywhere in the browser and telnet
configuration to change this.
Also, it never shows the name of the caller. I've even tried forcing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Wolfe
Sent: Friday, May 27, 2005 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue
Maybe I should my pictures in with me an
Maybe I should my pictures in with me and supermodels. :-)
Cheers,
-Scott
- Original Message -
From: "Wiley Siler" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, May 27, 2005 12:26 PM
Subject: RE: [Asterisk-Users] VoiPSupply Dot Com: E
> >It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
>
> Perhaps a more pressing question might be how you are going to backhaul
> Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
> metres reliably, and using Ethernet repeaters every
Hello all,
I have an asterisk box on the open internet (one port). I can get
sip clients from behind a nated network to register with this box and
make phone calls, but when the other person picks up they can hear me
just fine but I cannot hear them at all. Is there anyone that can
help me get t
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> brandt Milczewski
> Sent: Friday, May 27, 2005 10:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Newbie here. Tips on setting up 100
> phones wanted.
Still having trouble, I even tried to remove it from the
allowed methods, but the other end still sent a REFER anyway which asterisk
accepts but kills the call anyway.
Asterisk is being used in a back-to-back sip arrangement to
take advantage of its IVR and ACD capabilities. So, inbou
Cool. Sounds like you've got your poop in a group already. As to your
original question, I am partial to the Snom 190 phones, they are easy to set
up, look and perform great, users really like them, and they seem quite
tough. I have two of them running in a shop environment where they get
covered i
LOL - You mean he actually 'met' Newt Gingrich? How dare you not extend
him credit!!!
I mean seriously... For such a distinguished individual...
Hey, not only have I met the heads of several multi-billion dollar
corps, I have gotten absolutely blasted drunk with them.
So I should get credit, a
Has anyone had any experience with the Soyo G688 phone? I'd like to
use it as a agent's phone. Is it reliable? How well does it work with
*? How's the quality? Features?
Thanks,
Waldo
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h
Steve, you should really test the Codec and have G729 running as a pure
IP to IP call you can not hear the difference on good networks! Please
do not get me wrong that G711u sounds better through the PSTN. Thats a
given! You can't convert G729 up and down to G711 and expect the sound
quality t
Stable Version 1.0.7
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Friday, May 27, 2005 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] PRI "Actual-HookState" not showing offhook
oninbound
On May 27, 2005 10:57 am, Ronald Hartmann wr
Yes I have group=1. Also I've try with
exten => 203,1,Dial,Zap/1/number
and always the same message:
May 27 20:56:20 NOTICE[1041]: app_dial.c:759 dial_exec: Unable to
create
channel of type 'Zap'
== Everyone is busy/congested at this time
I think that t
I promise this will be the last chapter in the story, at least from my
end. If we could only get George Lucas to say the same thing about the
godawful new Star Wars movies he keeps cranking out.
I did some research this morning, interviewed the involved parties on
our end, and have come t
Sorry wrong model CS1000 not BS1000. typo
Thanks,
Jerry
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Any WORKING uses of asterisk connected to a nortel BS1000 using SIP?
I can setup asterisk fine. I have never seen a BS1000 the Nortel guy is not
being able to set it up? Anyone have setup steps?
Thanks,
Jerry
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Wow, I am pretty sure you should update both the bootrom
and sip.ld at the same time.
I would do this with at least one phone and see what
happens.
W
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
CoulthurstSent: Friday, May 27, 2005 11:37 AMTo:
asterisk-users@li
On 09:01, Fri 27 May 05, Max W Blackmer Jr wrote:
> hello everyone,
>
> I just had a thought on this subject why not create a daemon process on
> the Client PC That registers its self and What phone the user is
> connected. An AGI script could monitor the progress and when answered
> could send a
So in order to answer the background and backbone questions here is
the system as it is. I hope this isn't too much for the list but I'll
post it in response to a few inquiries.
The current system is quite interesting.
We have an office in a town that is about 50 miles from
the ski area. The ski
Ever since I upgraded my Polycom 500 to the newest sip.ld
(kept the old bootrom), when I dial things like “*98” for
voicemail, the screen shows “9*8” and doesn’t dial my
voicemail system! Is this user error, or errors in the new firmware?
Chris Coulthurst
[EMAIL PROTECTED]
_
I've configured SIP softphone to work with asterisk n it's working fine.
but i m unable to connect my netphone IP phone I've connected my phone
to LAN and assigned an IP address to it but how can i make call... plz
tell me step wise.
On 5/27/05, Karl J. Vesterling <[EMAIL PROTECTED]> wrote:
> At 08:59 AM 5/27/2005, you wrote:
>
> [ snip for brevity ]
> I just wanted to clarify ... this isn't a voipsupply.com problem at all,
> but
> rather a courier screwup... which happens anywhere and at anytime... right?
>
> TWO scre
Is pstn call waiting working on a Digium TDM-400 with FXO ?
Configuration in zapata.conf:
callwaiting=yes
callwaitingcallerid=yes
callprogress=yes
If an incoming call happens while the FXO channel has a call in progress,
and the call is routed to a FXS channel (which has callwaiting=yes in
zapa
Title: Message
anyone got any ideas
on this?
TDM > H323
Gateway > SIP
Inbound H.323 call
'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and
attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting
channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) nativ
Title: Message
May 28 01:58:32 WARNING[6821]: chan_sip.c:1830
sip_write: Asked to transmit frame type 256, while native formats is 4
(read/write = 4/4)May 28 01:58:33 WARNING[6821]: channel.c:2126
ast_channel_make_compatible: No path to translate from SIP/3901506-efd7(4) to
OH323/L4592(256)Ma
Peter Bowyer wrote:
On 27/05/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
The person on 617 is unavailable --- Why
Maybe he's in the bathroom?
No, I am testing phone (not in the bathroom ;-) )
The condition being reported is coming from the UA on the end of the
SIP ca
On May 27, 2005 01:27 pm, Michael D Schelin wrote:
> I have used G729 and it sounds almost as good as G711U. The problem is
> the way Asterisk uses it. It does not sound robotic and it's not suppose
> to sound that way. Most Carriers want the calls to be in g711u so
> thats why I use G711u otherw
Title: Message
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
AndrewsSent: Thursday, May 26, 2005 6:33 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] VoiPSupply Dot Com
Karl, fir
Michael D Schelin wrote:
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not
suppose to sound that way. Most Carriers want the calls to be in
g711u so thats why I use G711u otherwise I want to save money on
b
I am using version 1.4.1.0040 and 2.6.1.0003. I have read about newer
versions out there, how can I get them? I'm having an intermittent issue
regarding DHCP with one of my phones, and I recall when I loaded 1.4.1 on
this one phone, it failed once, then it succeeded the second time.
I am wondering
I've got the same issue with a Swissvoice IP10S SIP phone. I couldn't
find much information with this issue, but it seems to appear because
Asterisk does not support variable length for g.729 (don't ask me what
it really means). Anyway, it is recommanded to disable the silence
suppression, whic
On Fri, 27 May 2005, Mike Clark wrote:
> brandt Milczewski wrote:
>
> >I work for a ski area. I currently use a 3Com Superstack for in our
> >office. And an old small town phone system for up at the mountain. The
> >phone system is dying and I'm hoping to bring IP to replace the old
> >phones. It
Have you placed a group=1 in zapata.conf ?
For a trial, you can use: exten => 203,1,Dial,Zap/1/onetelephonnumber
Aitor a écrit :
I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] =>
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not suppose
to sound that way. Most Carriers want the calls to be in g711u so
thats why I use G711u otherwise I want to save money on bandwidth. G729
on Asterisk
I thought he meant that as well but I hope that what will occur is that
there is DSL somewhere already that can be utilized.
That conflicts with the 'old town PBX' scenario as well though.
So, assuming there is DSL already, that even makes you wonder why bother
if a phone line already exists and
On Fri, 27 May 2005, Colin Anderson wrote:
> >It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
>
> Perhaps a more pressing question might be how you are going to backhaul
> Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
> metres
Well, that will be pretty preferential
As stated before, I love the Polycom IP500. I think it is just a great
phone for less than $200.
It configs easily once you get used to the config file and Polycoms have
great speaker phones.
Many love the Ciscos... Admittedly a beautiful phone but hard
Wireless bridges??
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Colin Anderson
> Sent: Friday, 27 May 2005 1:18 PM
> To: 'brandt Milczewski'; 'Asterisk Users Mailing List - Non-Commercial
> Discussion'
> Subject: RE: [Asterisk-Use
On Fri, 27 May 2005, Mark Elkins wrote:
> I tried to do an HTTP update from the Grand Stream web site...
You upgraded the firmware over the Internet? You are braver than I am. I
would have used a local http server.
> Is there a magic re-incarnation routine ?
> (Power on whilst holding down some
brandt Milczewski wrote:
I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to
If you have the money, I would recommend the Cisco 7900 series, but if
you need cheap phones, go with sipura. I can get you all you need if you
want. The Sipura phones run about $100.
Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC
>It will be about 100 phones at about 20 locations all within
>about 4 miles of each other.
Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres reliably, and using Ethernet repeaters every hundred met
On 27/05/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> The person on 617 is unavailable --- Why
Maybe he's in the bathroom?
The condition being reported is coming from the UA on the end of the
SIP call - is there a DND setting or something there?
Peter
--
Peter Bowyer
Email: [EMAIL PR
I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
At 08:59 AM 5/27/2005, you wrote:
[ snip for brevity ]
I just wanted to clarify ... this isn't a voipsupply.com problem at all,
but
rather a courier screwup... which happens anywhere and at anytime...
right?
TWO screw ups in the shipment.
1.) It was shipped to the Bill-To address. Since there
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> MF Hulber
> Sent: Thursday, May 26, 2005 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Size of extensions.conf
>
> Although you may not see
On May 27, 2005, at 6:50 AM, Andrew Kohlsmith wrote:
On May 27, 2005 03:20 am, Nathaniel Angelo A. Torres (247talk) wrote:
Any idea how I can generate a B* message on the asterisk box (out of
order)
message?
Easy. Do *not* have an exten => line in your PRI incoming context that
matches the
Not sure I understand your meaning. You have a phone with 105 as the
registered extension but you want to dial 105 and get voicemail?
Lets assume that *98 is your voicemail extension.
If you dial *98 from any phone it should ask for the extension and
password.
So extension *98 looks something l
On 27/05/05, Max W Blackmer Jr <[EMAIL PROTECTED]> wrote:
> hello everyone,
>
> I just had a thought on this subject why not create a daemon process on
> the Client PC That registers its self and What phone the user is
> connected. An AGI script could monitor the progress and when answered
> could
The person on 617 is unavailable --- Why
*CLI>
-- SIP Seeding peers from Astdb: '617' at [EMAIL PROTECTED]:6990
for 3600
-- Executing Dial("SIP/601-f18a", "SIP/617|60|tr") in new stack
-- Called 617
-- Got SIP response 480 "Temporarily Unavailable" back from
192.168.250.107
hello everyone,
I just had a thought on this subject why not create a daemon process on
the Client PC That registers its self and What phone the user is
connected. An AGI script could monitor the progress and when answered
could send a push to the registered daemon which would push a link to
the r
Define happily?
Jitter is obviously important, but latency is too. For day-to-day
business calls, 250ms is a little high. Both parties will definitely
notice it. In my experience you will find yourself talking over one
another quite often. Even with 100ms this continues to happen from time
to tim
On May 27, 2005 10:57 am, Ronald Hartmann wrote:
> Actual Hookstate: Onhook
What version of Asterisk is this? CVS HEAD from about an hour ago does not
show hookstate for anything but FXS interfaces, and certainly not for PRI.
-A.
___
Asterisk-Users ma
I've got three GS 100 Phones with same problem.
Some lights.
Some no lights.
Some garbled display.
I would welcome suggestions for a resurrection.
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EM
Check the documentation for the Directory application on
http://www.voip-info.org .
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
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> Sent: Friday, May 27, 2005 10:54 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk
Waldo Rubinstein wrote:
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of calls
when there is such high latency? Can anyone comment?
L
may be something like this?
[macro-sipextens]
exten => s,1,SetVar(voicemail=${ARG1})
exten => s,2,Dial(SIP/${ARG1},40,r)
exten => s,3,GotoIf($[${DIALSTATUS} = NOANSWER] ? 66 : 3)
exten => s,4,GotoIf($[${DIALSTATUS} = BUSY] ? 68 : 4)
exten => s,5,Playback(iss_invalid_sipexten)
exten => s,6,Ha
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of
calls when there is such high latency? Can anyone comment?
Thanks,
Waldo
I have a Grandstream GSX-2000 with ..
Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3
I tried to do an HTTP update from the Grand Stream web site...
After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
Gran
I tried it a while, its impossible.
(Well you can get it to work, but not in a stable way)
Zoa.
Laurent Tostain wrote:
Hi,
Someone released a succefull interconnection in H323 with WTL equipement
?
I'm trying to do that with an IPNx. But get dead air.
With chan_oh323 it's fine, al
Good day list,
I have been fighting echo problems on my PRI card.
Everything is working great outbound, however inbound calls have
echo.
I have found the issue but need help in fixing it.
During outbound calling "zap show channel 18" shows that the
call is off-ho
I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, PRI Signalling signalling
--
Is it possible to have a Dial my name menu in Asterisk?
AZ
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Hi,
Someone released a succefull interconnection in H323 with WTL equipement
?
I'm trying to do that with an IPNx. But get dead air.
With chan_oh323 it's fine, all works. With chan_h323 => dead air.
The configuration is GW to GW.
This is my configuration from h323.conf:
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