Hello, Martin!
At 02:50 AM 02/26/2006, you wrote:
I got my new Welltech 3701a, 1FXS,1FXO gateway.
If you do give up with it (isn't Engrish documentation fun?), you may
wish to take a look at the Sipura SPA-3000. I have one but haven't
put it to use yet. I've heard *many* good things about
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had
much luck in locating a decent web-based voicemail system for
Asterisk to check your VM while you're away from the office without
using a phone.
Can anyone make any recommendations for such packages/applications?
On Sun, 2006-02-26 at 03:57 -0500, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had
much luck in locating a decent web-based voicemail system for
Asterisk to check your VM while you're away from the office without
using a phone.
Can
Yeah! Sorry, the thread has grown a bit...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Joseph
|Sent: Sunday, February 26, 2006 1:55 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] fax
On Sun, 26 Feb 2006, Alexander Burke wrote:
Hello, Martin!
At 02:50 AM 02/26/2006, you wrote:
I got my new Welltech 3701a, 1FXS,1FXO gateway.
If you do give up with it (isn't Engrish documentation fun?), you may wish to
take a look at the Sipura SPA-3000. I have one but haven't put it to use
Please let us know how it turns out, if there are any
issues etc.
--- Richard OSS [EMAIL PROTECTED] wrote:
Thank you very much. Will go ahead and build the
system. Hope everything goes smoothly.
richard
BJ Weschke [EMAIL PROTECTED] wrote:
On 2/22/06, Richard OSS wrote:
Hello,
I have a POTS and a sip incoming into my asterisk server. When I call
the POTS number from outside (cell or landline) and trying to
authenticate myself when enter #, 8 out of 10 times I got an
authentication incorrect. If I call in to the sip incoming line, 10
out of 10 times the authentication
try dtmfrelax=yes in your zapata.conf
W
- Original Message -
From: Wooi Koay [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, February 26, 2006 8:37 AM
Subject: [Asterisk-Users] authenticate problem
I have a POTS and a sip incoming into my asterisk server. When I
Sam Tam wrote:
Hello Dan
I can assure you that our GSM Gateway quality is absolutely excellent and
this fact can be supported by hundred if not thousand of our users.
It is also very simple to use and even a newbie can set it up..
Does it provide Disconnect Supervision? If so, what method
Im starting to get a lot of errores from asterisk when transfering calls
from one phone to another:
Incoming call: Got SIP response 500 Internal Server Error back from
What does this error usually mean?
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Anton Krall wrote:
Im starting to get a lot of errores from asterisk when transfering calls
from one phone to another:
Incoming call: Got SIP response 500 Internal Server Error back from
Not really sure on this one, but I see it on all of our Polycom phones when
doing a transfer. Doesn't
You hit the spot Doug, my phones ARE polycom and this happens exactly when
doing transfers.
So, no problems arise from this?
BTW, have you found a way to make the polycoms keep the volume settings
(handset, ringers and speaker) after they have been restarted or rebooted?
|-Original
On Fri, 2006-02-24 at 10:58 +, Barry Flanagan wrote:
Asterisk Sales wrote:
mailto:asterisk-users@lists.digium.com
Hello list,
Is there any way to use a2billing without the IVR for the sip/iax users.
(authentication is done by the user id and pass as user registers with
I usually see this when doing operations on variables that are blank. In
your case, the input is ' + 1'. Clearly there was something to the left of
the + but it's blank.
If you're adding one to something, make sure there is a number on the left
side of the plus sign. Probably by initializing
Anton Krall wrote:
You hit the spot Doug, my phones ARE polycom and this happens exactly when
doing transfers.
So, no problems arise from this?
Not that I've had reported. I'm not on the same phone system, so don't
have daily interaction with it.
BTW, have you found a way to make the
I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent
at best. Sometimes it's great other
times completely unusable. When it's
bad one usually hears harsh static when the other party speaks or their voice
gets "clipped" to
Hi
I have good results in using, the old very (free) of firefly (IAX2),
with g729!
rgds
Jesper Langpap
hugolivude wrote:
I have a bunch of road warriors who I've set up with Xlite clients.
Unfortunately the sound quality has been intermittent at best.
Sometimes it's great other times
Hello all,
this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe
I would do some reading on "mesh" wifi
networks. Here are a couple links of interest
http://www.wi-fitechnology.com/Wi-Fi_Reports_and_Papers/Introduction_to_Wi-Fi_Mesh_Networks.html
http://www.locustworld.com/
http://www.meshdynamics.com/
http://www.tropos.com/
Cory J
If you ever find a way, let me know Doug.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Doug Lytle
|Sent: Sunday, February 26, 2006 11:19 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Internal
I've used quite a few of the rack-mount servers from
http://www.siliconmechanics.com/. I've had both digium and sangoma
cards in them with zero issues.
-Chris
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Asterisk-Users mailing list
To
hello-
how can i change the default voicemail folders (old, work, friends,
etc...) that asterisk uses?
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Or you could skip the overhead associated with an AGI and use the
dialplan command availabe after installing asterisk-addons MYSQL.
exten = _X.,1,Read(PO-NUMBER,enter-yr-po-num)
exten = _X.,2,MYSQL(Connect connid dbhost dbuser dbpass dbname)
exten = _X.,3,MYSQL(Query resultid ${connid} SELECT
Hello, Dumpexec!
At 12:35 PM 02/26/2006, you wrote:
Is there a sort of high grade cat5 cable that can propagate signals
for up to 1Km?
No. The standard is 100m per leg, maximum, even with STP (shielded
twisted-pair) cable. You could go to multimode fiber to get 2km, but
you'd have to find
What kind of overhead do agi put on? Are they that much worst than running
mysql addon functions?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Chris A. Icide
|Sent: Sunday, February 26, 2006 12:23 PM
|To: Asterisk Users Mailing List -
On 2/26/06, Anton Krall [EMAIL PROTECTED] wrote:
You hit the spot Doug, my phones ARE polycom and this happens exactly when
doing transfers.
So, no problems arise from this?
BTW, have you found a way to make the polycoms keep the volume settings
(handset, ringers and speaker) after they
How is the voice quality?
I've just plugged mine back into the charger after having used it
nearly all day. I didn't have any of the problems you've described.
Sorry you're having such bad luck with it. I'm not certain what the
phones are rated to do, but I probably got better than 3 hours
On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had much
luck in locating a decent web-based voicemail system for Asterisk to
check your VM while you're away from the office without using a phone.
Can anyone
On Feb 26, 2006, at 5:37 AM, Wooi Koay wrote:
I have a POTS and a sip incoming into my asterisk server. When I call
the POTS number from outside (cell or landline) and trying to
authenticate myself when enter #, 8 out of 10 times I got an
authentication incorrect. If I call in to the sip
At 09:35 AM 02/26/2006, you wrote:
Is there a sort of high grade cat5 cable that can propagate signals
for up to 1Km? or does anyone have any idea that i could effectively
cover 20Km radius wing WiFi?
If you're trying for wireless, have a look at dd-wrt.com and sveasoft.com .
Ira
--
No
Hi there,
I am running Asterisk 1.2. I have a Grandstream GXP2000 and Aastra 9133i
with BLF/Speedial configured for other extensions. The hint's are all
configured in extensions.conf and it seems to work as it is supposed to
until I reload the configuration in Asterisk or reboot the server.
On 2/26/06, mustardman29 [EMAIL PROTECTED] wrote:
Since I am using two completely different phones it must be my Asterisk
configuration.
I don't know about the aastra, but on the GXP-2000 this is a bug. (do
you run latest firmware? maybe it's fixed in that one)
cheers
[EMAIL PROTECTED] is believed to have said:
I was not sure of the quality of both these hence i wanted some real
user experiences.
Thanks anyways...
Dan
Dan,
you're welcome...
While I can't speak for the quality of the Dock'n'Talk, as I did never
see one, the GSM gateway's audio quality is
Martin Joseph wrote:
On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote:
Hello, list!
After Googling and checking out the voip-info wiki, I haven't had
much luck in locating a decent web-based voicemail system for
Asterisk to check your VM while you're away from the office without
using
If you do a 'reload' in Asterisk, it deletes all the sip subscriptions. Do a
'sip show subscriptions' before and after a reload command. They will
disappear. I've been bitching about this for a while, and asking why
subscriptions can't be stored in astdb like registrations.
If you reboot the
Is there any way not using group count, to limit calls received by every
endpoint SIP?..
Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch.
Is there another command to do that?
Regards
Alberto
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
what about ARI, it gives web based access to the voicemail and is pretty
good at it... the default vmail.cgi is probably not the best as it has a
gaping security hole that allows anyone to listen to anyone elses
messages :)
Sean
Martin Joseph wrote:
I second this...as a road warrior myself I find too many places where
SIP clients just won't work. So I rely on Firely over IAX2 which has
been 100% reliable.
Also, John Todd has been using the PSGW SkypeSIP gateway software in
new and different ways. Perhaps that's an option.
On Sun, 26 Feb
Hi All..
I've noticed that there are quite a few different billing solutions
availible.
If I want to have both prepaid and postpaid accounts with only ATAs (or
other SIP devices) which one should I use ?
Some users are prepaid, and some are postpaid accounts (invoice)
I do not want the
I am sorry but I am not very technical on the way it provide disconnection.
May be other of our customers can answer this question on our behalf
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Sunday, February 26, 2006
You won't be able to do this with any residential WiFi-products.
Your only options are:
1. Fiberoptic cable
2. Business class WiFi with directional antennas
3. WiMax
In either case it's going to cost thousands... :-(
-- Bjorn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Is there any way not using group count, to limit calls received by every
endpoint SIP?..
Outgointlimit and Incominglimit seems to be deprecated on 1.2.x branch.
Is there another command to do that?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
On Sun, 2006-02-26 at 15:50 -0600, Michael Graves wrote:
I second this...as a road warrior myself I find too many places where
SIP clients just won't work. So I rely on Firely over IAX2 which has
been 100% reliable.
I'm using idefisk softphone with iLBC on my Debian roadwarrior laptop
and
Hi,
Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this
phones but seems to work only with net2phone, in the product page
http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTPmodelid=M_IP100C
the features are showing SIP and H.323 support.
Can be used with my
Hi there,
Well, looks like you are going to start a new telephony company ,-) ...When
it is a quite dense populated area then there
should already be enough cables and operting providers ...
If you can find support by cable owners then you could indeed start to
setup some WiFi cells using
My company One Ring Networks Inc. is a wireless ISP / carrier providing
VoIP services using asterisk to our customers in the metro Atlanta
area. Powering the radios has never been an issue. There is always
power within 30 to 50 meters of where you will want to mount a radio.
If you indeed
Am Saturday 25 February 2006 23:49 schrieb Anton Krall:
Whats mpack tom?
a command line tool for easily sending emails with attachments.
I use sendEmail..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Thomas Artner
|Sent: Saturday,
On Sat, 2006-02-25 at 13:42 -0500, Dean Collins wrote:
I know this is a OT but great article
http://www.theregister.co.uk/2006/02/23/rwanda_terracom/
Will be interesting to see how this project goes.
Hmmm - it is nice to see things like this happening, but I would have
thought that
Matt Roth wrote:
...
What is being discussed here is basically what I was planning on
...
This sounds like a programming project. Something like a stripped
down softphone (or possibly a plugin to an existing phone) with
Hi,
since I need rather a tool not that versatile but within some
There used to be an 'enablebadnames' or similar option for adding user
accountsbut it doesn't seem to be needed for FC4.
PaulH
On Sat, 2006-02-25 at 09:06 -0500, Stagg Shelton wrote:
I've been using vsftpd since fedora core 2. There was a period of
time in FC3 when linux wouldn't let me
|
|
|It's in sip.cfg:
|volume voice.volume.persist.handset=1
|voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/
|Make sure it says 1 and not 0
|
That makes persistent volume between calls for the handset, headset and
speakerphone but not after restating the phone.
Say, thanks to all you for your time in responding.
I hope I don't sound unappreciative (I have no time for flamers) but I
don't understand how changing from SIP to IAX would make any
difference. I don't have any problems with the signalling (i.e.
phones ring when I make and receive calls), the
Hi
I am about the purchase a server and would like to know if anyone has
had any experience with the TE410P Rev 2 in a server that has a
ServerWorks BCM5785 (HT-1000) chipset.
Thanks
Master
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Looking for some feedback on whether nat=yes and qualify=yes
will provide a workable solution in many cases?
The * server is on a public address, no NAT, the UAs (sipura,
linksys, polycom) are behind various types of NAT.
Obviously port mapping in the NAT device works, but what
about
Sipura works, I never tried linksys, Polycom might and might not work.
On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Looking for some feedback on whether nat=yes and qualify=yes will provide a
workable solution in many cases?
The * server is on a public address, no NAT, the UAs
Any idea how to read an external file, grab some stuff and push it back
into an Asterisk variable?
I can do it the other way with:
system(echo ${UNIQUEID} = /home/ast/curr_calls)
but I'm a bit stumped on how to go the other way around
much thanks,
Paul Hales
I thought that's what you were looking for, I don't think there is
anything like what you want.
On 2/26/06, Anton Krall [EMAIL PROTECTED] wrote:
|
|
|It's in sip.cfg:
|volume voice.volume.persist.handset=1
|voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/
|Make sure it says
Thanks, the linksys is a sipura, so what works on one should work on the
other.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, February 26, 2006 7:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Any disadvantage to always setting nat=yes for all UAs just
in case they end up behind a NAT at some point?
Canreinvite=no is always set since a few of our features
require it (transfers, etc.)
What is the impact of qualify=yes for 250-500 UAs?
On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote:
Thanks, the linksys is a sipura, so what works on one should work on the
other.
I wouldn't bet on it, since the SPA-xxx where out before the merge.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On 2/26/06, Sean Cook [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
what about ARI, it gives web based access to the voicemail and is pretty
good at it... the default vmail.cgi is probably not the best as it has a
gaping security hole that allows anyone to listen to
We're setting up asterisk at the office (really doing some testing
right now) and it is going to be hosted on a dual G5 XServe running
OS X. We're an apple certified solutions provider, etc. so we want to
build all our stuff on apple hardware and software. Anyway, the last
sticking point
Like BJ, I'm sorry you had bad luck Phil. I have been playing with
this phone all weekend, and I have had minor problems. The voice
quality is as good as my cisco and polycom sip phones. I asked a
friend to guess what kind of phone I was talking on and he said it
sounded like a regular home or
Hi, I am using asterisk 1.2.1 for building an ippbx for my setup. I am having problem in accessing advanced options option that comes on pressing 5. It says to leave a msg and asks for extension. Now when i dial the extension the IVR silently goes into the top IVR menu. And the asterisk console
On Feb 26, 2006, at 5:43 PM, hugolivude wrote:
Say, thanks to all you for your time in responding.
I hope I don't sound unappreciative (I have no time for flamers) but
I don't understand how changing from SIP to IAX would make any
difference. I don't have any problems with the signalling
I tried 1.0.1.12, 1.0.1.13 and now 1.0.2.13 on the GXP2000. I doubt it's
the firmware if it is happening on two completely different phones. I read
of a bug in Asterisk that exhibited this exact behaviour that was apparently
fixed in Asterisk 1.2.0 as far as I can tell.
On Feb 26, 2006, at 11:59 AM, Paul wrote:
snip
Unless you have good QOS routing be sure that mail server is somewhere
where you don't have voip phones. I had a mail server at an office with
400k sdsl. I would be on a call and let an incoming call go to voice
mail. The incoming email with wav
According to this blurb I found on the Asterisk Wiki, it was supposed to be
fixed so it still works after a reload. Your suggestion is all fine and
dandy but does nothing to rectify a server reboot. If phones have to be
rebooted everytime the Asterisk server is rebooted or the sip.conf is
umm..
Can you please tell me what phone u r talking about??i mean does it support IAX.
Actually i am sick and tired of my DIAX and want a new IAX phone...
I am using an older version of * like 1.0.3
I hope u will not mind replying to me
On 2/26/06, Me [EMAIL PROTECTED] wrote:
How is the voice
On Thu, 2006-02-23 at 02:13 -0600, Anton Krall wrote:
Now thas confusing to me.. How do you actually take 16 calls at a time? I
see 301's have 2 line keys.. And each can handle 16 calls... How do you
actually take all 16 and switch between all of them?
Hmmm, top-posting...
anyway, pretty slow
At 05:03 PM 02/26/2006, you wrote:
I want to match the user from the users callerid. All users have DIDs.
You probably shouldn't do that for security reasons -- rather, match
them according to the SIP username/password pair they provide when
they register.
--
Alexander Burke, A+, CCNA
Hi,
- Original Message -
From: amna saleem [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:12 AM
...
Actually i am sick and tired of my DIAX and want a new IAX phone...
I am using an older version of * like 1.0.3
Can you provide me more details about your sentence...i am sick
Hi,
Can some one advice me that how can I make the FXO channels port answer an
incoming calls, means when I call from Lan line to Asterisk TDM400, my phone
get ring immediately. When POT FXO port is ringing, Asterisk seems like
studying the incoming ringing pattern even it did answer the call.
I
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I am getting repeated messages in my logs with the following:
*
Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default'
Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hello Asterisk community.
We have a small User-group in Melbourne Australia.
Recently I brought up the issue of STANDARDS for dialing Applications on
a PBX.
This generated some interest but also the fact little has been done on
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