Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
Hello,
we are running a Asterisk (1.2) installation with about 80 snom phones
(300,320,360).
Now have the demand for a special manager - assistant setup for a few
extensions.
Since Shared Line Appearance is not available
Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:
Hi group,
I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).
I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo
Am Mittwoch, den 10.01.2007, 13:41 -0400 schrieb H Aranguren:
Thanks for your answer Anselm,
But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the
Am Mittwoch, den 10.01.2007, 18:44 -0500 schrieb cb:
On Jan 10, 2007, at 6:33 PM, M.Hockings wrote:
But as far as I can tell, there is nothing requiring you to use the
service, if you choose not to, they work just like a normal cordless
phone.
There was a similar business model here in
Am Samstag, den 06.01.2007, 23:19 -0800 schrieb Yuan LIU:
After the user navigated some voice menus, how do I give him another (fake)
dial tone?
If you want the user to get the tone meaning please dial a number now,
perhaps DISA is the right you.
Am Samstag, den 06.01.2007, 23:02 -0800 schrieb Yuan LIU:
I'm creating extensions _*XX. But whenever I press *0 or *8, Asterisk
throws out congestion and hangs up. I set verbose to 6 and debug to 6, but
all Asterisk cares to display in console is
-- Starting simple switch on 'Zap/1-1'
On 2007-01-07 01:23:22 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:
Marty,
Where are you paying $1000 for a 1600 series Cisco? I can get you
20% off that price on any quantity (note: Sarcasam). Its not the
1990's anymore. You can get them on eBay ($50-150) for only slightly
more
On 2007-01-04 09:56:58 -0800, Mike [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hi,
I'm looking for opinions on the best value router to use for home offices.
It should work for a scenario in which there are 3 computers and 2 SIP
phones, handling QoS so that the
On 2007-01-05 09:40:18 -0800, Biju [EMAIL PROTECTED] said:
hi,
i am using nokia e61 . we have an asterisk server and i want to use my
nokia phone to register with asterisk server .
anybody can help me to do this.
Try Google, it works.
Marty
On 2007-01-06 00:48:11 -0800, Mark Coccimiglio [EMAIL PROTECTED] said:
Mike
I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router with
Fair-Weight queueing enabled. Works great. The nice thing about
Fair-Weight queueing is that it dynamically adapts to lower the
priority of
Am Donnerstag, den 04.01.2007, 12:27 -0500 schrieb Matt Gibson:
Thanks Tzafrir,
I knew about the ex girlfriend logic - but does that allow to be
looked up from the callerid database instead? I'd like to have it only
be one goto for direct dial, and one for the main menu instead of
having to
Am Donnerstag, den 04.01.2007, 17:47 -0500 schrieb Noah Miller:
Let's say you use the Asterisk DB() stuff for your caller ids, storing
them in a branch called book like
book/16175551234 John Doe
book/12125559876 Jane Miller
Then you could go with a logic like
exten =
Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French:
I have an upcoming install which places the switch close to some
employees in a quiet work environment. Can anyone recommend a quiet
24 port POE switch? The Linksys SRW224P behind me right now would be
objectionable, I'm sure.
I
On 2006-12-21 13:29:47 -0800, cb [EMAIL PROTECTED] said:
Has anyone used either the 8 port or 4 port FXO device from
Grandstream? (GXW-4108 or 4104).
They seem to be the lowest cost multi port FXO devices that I can
find, so I'm getting ready to buy the 8 port version. I just want to
see
function in queues.
but in agents.conf we cannot find a possibility to activate a script after
the recording.
Anyone a idea, how we could run our script after a agent recording? ? ?
Thank you for your help!
MArtin
___
--Bandwidth and Colocation provided
On 2006-12-31 00:52:27 -0800, mitcheloc [EMAIL PROTECTED] said:
Those wifi phones are neat but I'd rather not carry around two
devices, does anyone know of any good dual-mode GSM/SIP phones?
I'm using a T-Mobile MDA right now and it is way too slow.
Apparently the Nokia e61 has a built in SIP
On 2006-12-24 00:35:06 -0800, Martin Joseph [EMAIL PROTECTED] said:
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems
Am Mittwoch, den 27.12.2006, 17:22 -0800 schrieb Tom Lynn:
Can anybody point me to a vendor that can provide a toll free number
that can be used in India to reach the united states? Verizon
Business is telling me they can't get one.
Looks like a -biz related question...
A quick google search
On 2006-12-20 05:18:08 -0800, Chris Blunt [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hi List
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Depends a lot on your geographical
Hi,
I have a spiffy new gateway which seems quite promising.
It's the Audiocodes MP114 FXS_FXO (2 of each).
I have got it configured and working reasonably well, but have a couple
of issues.
1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway
after 10 seconds. This isn't
Am Sonntag, den 24.12.2006, 03:47 -0500 schrieb Dovid B:
Anyone have an answer for this ?
- Original Message -
From: Fernando BERRETTA [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 14, 2006
Am Sonntag, den 24.12.2006, 12:49 +0100 schrieb Michiel van Baak:
On 12:25, Sun 24 Dec 06, Anselm Martin Hoffmeister wrote:
I tried getting something similar, but I got stuck with two points:
- For MeetMe you need a timing source. I do not have a ZAP card, and
cannot install ztdummy
Am Freitag, den 22.12.2006, 09:53 -0500 schrieb Doug Crompton:
Wow what a mess! I can imagine how much easier it would be if the world
adopted a country/area/exchange scheme like in the US with known length.
It must be complicated in Germany just within the country. At least in the
US we know
Am Freitag, den 22.12.2006, 00:53 -0500 schrieb Doug Crompton:
Question... What is the purpose of the + before the number? Does anyone
actually have to enter it? If so how would you do it? It is not used in
the US but do I see it come in on SIP lines CID. I assume the CID ignores
it in the
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango:
How about:
exten = _X.,1,Answer
Does it include all numbers and characters?
As of the docs, no. It should only match 0123456789
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
BR
Anselm
Am Mittwoch, den 20.12.2006, 14:42 -0500 schrieb Doug Crompton:
Anthony,
Ok I understand. The 011 is unique though and I guess the problem is
the length of the remaining digits. This could vary based on country?? and
I suspect there is no unique rule that could be applied??? I have not
Am Dienstag, den 19.12.2006, 01:13 -0800 schrieb Angel Heart:
Hi Paul Eric,
Thank you for you information and quick response. I had enabled
Monitoring in every SIP phone already. Made some Playback see below
truncated config;
exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,
Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and
Am Sonntag, den 17.12.2006, 18:11 -0500 schrieb Time Bandit:
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten =
Am Freitag, den 15.12.2006, 13:08 -0600 schrieb Alvin Austin:
Hello,
In Asterisk 1.4 beta 3, the UPGRADE.txt file says:
Variables:
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP},
${ACCOUNTCODE},
and
Am Samstag, den 16.12.2006, 02:32 +0530 schrieb Vicky:
I have shifted asterisk port to 5091 . Now i am able to register
properly using sjphone but still when dialing number it keep on
showing calling .. and do not go ahead . I change asterisk's rtp ports
too but still i am unable to make call
Am Mittwoch, den 13.12.2006, 11:47 +0100 schrieb Fabian Foerster:
Is there any output on the CLI that proves the BUSY command is run at
all? Because I don't really know if
exten = _X.-BUSY,4,Busy(1)
is gonna work. I would say something like:
exten =
Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin -
Williston, ND:
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:
Hi guys. This is my 1st post here (after much reading). I have a test
asterisk system setup using X-Lite Soft Phones, and the issue I am
running into is that caller id shows up as asterisk on all incoming
calls and on
Am Mittwoch, den 13.12.2006, 20:56 +0100 schrieb Sven Beisiegel:
Hi everybody...
I have a similar problem... I don't get the ID of the person that i
called on my phone... Does anyone know something about this problem?
greets,
Sven
Well, this sounds as specific as my computer is
Am Dienstag, den 12.12.2006, 01:32 -0600 schrieb Henry J. Cobb:
Luki [EMAIL PROTECTED] wrote:
You
don't gain anything QOS-wise by going through a tunnel, except hiding
your traffic in case your ISP purposefully assigns lower priority to
VoIP traffic and doesn't do it to OpenVPN/GRE/insert
Am Montag, den 11.12.2006, 22:19 -0800 schrieb Luki:
thats prety smart... think hard.. wot was the command u gave to exit
the CLI??
OK, come on everyone. This is getting ridiculous. That's the entire
point that stop now was NOT the last command on the CLI, yet it
shows up at the most
Am Montag, den 11.12.2006, 18:48 -0500 schrieb Barry Fawthrop:
Hi Anselm
Thanks for your input
Yes I was thinking of using OpenVPN so it was good to hear your experiences
I'm not so much concerned with the encryption of traffic etc..
But the Level of QoS.
If my IP Phone set QoS and the VoIP
Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry, please read Dial(SIP/exten,,D(9))
Just an
Hi Barry,
I used SIP over OpenVPN when travelling, especially from hotel rooms or
showfloors. Of course I did not expect the performance of a local SIP
connection, but generally it worked OK. The latency would not suffer
much in comparison to direct connection, but a WLAN was involved which
would
On 2006-12-10 04:13:02 -0800, RR [EMAIL PROTECTED] said:
Hello,
does anyone else have a problem with Asterisk crashing right after a
valid password/PIN is entered when trying to access voicemail in the
1.4b3 version? Not sure if this is anything to do with realtime per
se but I keep getting
Am Samstag, den 09.12.2006, 14:37 +1100 schrieb James Harper:
Normally when you think of using Bluetooth with mobile phones you think
of using it to attach a headset wirelessly to a mobile phone... can it
work the other way? Can I have a Bluetooth card on my laptop/desktop
such that my mobile
Am Dienstag, den 05.12.2006, 20:07 -0600 schrieb Eric ManxPower
Wieling:
Attended transfer is supported by every decent SIP device out there. It
is a basic phone feature. There are a few SIP devices out there that do
NOT support attended transfer but I would not call them decent. The
GS
Am Mittwoch, den 06.12.2006, 10:43 +0100 schrieb Giorgio Incantalupo:
Hi,
I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers
registrations: Asterisk freezes when it cannot (re-)register with VoIP
provider (registration timeout). The problem is related to DNS names
On 2006-12-01 09:45:00 -0800, Martin Joseph [EMAIL PROTECTED] said:
snip
So, I would like to purchase another PSTN gateway which WORKS WELL
with asterisk. I need it to hook up via ethernet, since my platform of
choice (mac OSX) has no PCI card support. I only have one PSTN line,
and already
personally (in particular the FXO and
and asterisk)?
Thanks again,
Marty
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Martin Joseph
Envoyé : lundi 4 décembre 2006 20:47
À : asterisk-users@lists.digium.com
Objet
On 2006-12-04 04:10:36 -0800, Giedrius Augys [EMAIL PROTECTED] said:
Hi,
I am testing Nokia E60 with Asterisk. And I noticed that if another side
is busy, nokia is still calling (I hear alerting), it do not show that
another side is busy. Maybe somebody has noticed the same problem too adnd
Am Freitag, den 01.12.2006, 20:41 + schrieb David Bath:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF
Am Mittwoch, den 29.11.2006, 20:48 -0500 schrieb Andrew Joakimsen:
Does anyone know where to source the Siemens Gigaset phones in North
America? I called 1-800-SIEMENS and was told the Gigaset range is no
longer marketed here since a few years ago. How far from being FCC
compliant is the DECT
Am Freitag, den 01.12.2006, 17:57 -0500 schrieb Mark Price:
Hi,
On 12/1/06, Mark Price [EMAIL PROTECTED] wrote:
hi,
I am using asterisk 1.2.10.
I am trying to send sip links in asterisk voicemail, so that
users can easily reply to emails.
Ok,
I am back from my thanksgiving holiday, and I find there was a big
snow storm here in Seattle. Apparently during the storm there where
multiple brown out/black outs.
I have struggled since day one to get a high quality PSTN gateway
configured with my very long loop and Mac based
Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis:
All,
If I have video phones behind an asterisk server (with 2 network cards)
and all the phones have extensions. Internally everything works great.
Now for people that want to call my video phones external to my office
is there a
Am Freitag, den 01.12.2006, 14:26 + schrieb David Bath:
So onto the problem… I’m trying to write a quick on-liner which will
fix up incoming UK format numbers (0 + 10digits) into (44 + 9 digits).
I got as far as this:
exten = incoming extension,1,Set(foo=${IF(REGEX(^0[1-9][0-9]\{9
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not
Am Freitag, den 01.12.2006, 15:27 -0500 schrieb Jerry Geis:
Am Freitag, den 01.12.2006, 13:44 -0500 schrieb Jerry Geis:
THanks, this seems to almost get me there... Once I call into the server
and goes to my locals I no longer get Video.
When I call the extension directly I get video no
Am Dienstag, den 28.11.2006, 17:23 +0200 schrieb Jean-Marc Salsa:
Hi,
I have tried to use the Record Command in Asterisk,
Here is the configuration :
exten = record,1,Answer
...
exten =
record,n,Record(/var/spool/asterisk/record/${CALLFILENAME}:WAV)
exten =
Am Montag, den 27.11.2006, 16:41 +0200 schrieb Altus Snyman:
Good day all
They all connect to the 4 Senao Long range AP’s 11mb
They all have the same ssi but 2 runs on channel 11 and 2 on channel 1
BUT..for some reason each now and the the AP’s will crash, you can
find a signal when you
Am Dienstag, den 28.11.2006, 06:46 -0800 schrieb Khaled:
What is the variable like $peerip to get the registered ip address
for a peer
You can use ${DB(SIP/Registry/sip507)} where sip507 is the section
name as well as username from my sip.conf- no idea which of both to use,
try it out.
This
Am Freitag, den 24.11.2006, 22:22 +0100 schrieb Vincent Delporte:
At 22:07 22/11/2006 -0700, Marco Mouta wrote:
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so
you can override whatever you
Hello list,
I currently try to get ChanSpy working to listen in to what people leave
on my voicemail. The problem seems to be though that ChanSpy only sends
the voicemail part of the conversation, namely the announcements,
which is not really helpful.
Is this a feature, or is spying into
On 2006-11-15 18:30:38 -0800, Lucas Barbuto [EMAIL PROTECTED] said:
Hi all,
Originally tried to post this without being subscribed, apologies if
the list gets this twice.
One of my users has a problem with many of his calls via my Asterisk™
server. He describes the problem as having the
[EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
On 2006-11-13 00:14:40 -0800, zen Perry [EMAIL PROTECTED] said:
Mac OS X, Asterisk 1.4 beta
Yeah, I am
Am Donnerstag, den 16.11.2006, 00:18 -0500 schrieb Matthew Rubenstein:
The US has a Do Not Call list to which people can subscribe to prevent
being called by advertisers. Federal laws (strengthened by some state
and more local laws) assign penalties for calling people/phones on the
DNCL.
Am Dienstag, den 14.11.2006, 12:28 -0500 schrieb joe a.:
Did not know how to make up a subject line for this.
I have a dial plan that allows a caller can try my cell phone. And that's
fine. If the call cannot be made, it sends caller back to voice menu.
However, I'd like a way for the
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson:
Hi!
I am getting inbound caller ID fine bout not out.
I am in Sweden and suing Rixtelcom /POrt80 as provider.
anyone knowing what is wrong?
Assuming that is a SIP provider, it is not your job to set the callerid
but the
On 2006-11-12 14:48:13 -0800, joe a. ([EMAIL PROTECTED])
[EMAIL PROTECTED] said:
Experiencing one way audio using IAX2.
I did see some other posts on this, and see there may be some internal
issues with asterisk and one way audio. Can this be a widespread
problem? So many seem to be
On 2006-11-12 23:08:05 -0800, zen Perry [EMAIL PROTECTED] said:
I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.
-- Started music on hold, class 'default', on
channel 'SIP/XXX'
--
Am Montag, den 13.11.2006, 10:37 -0500 schrieb joe a.:
Making custom voicemail greetings seems fairly straight forward, and I've
done it.
However, I'm looking for a way to make the actual extension answer with
You've reached my Jim Dandy voice mailbox, go take a flying . . .. (OK, so
Am Montag, den 13.11.2006, 23:42 +0100 schrieb Fred:
Hello
I'm trying to set up Asterisk on an older AMD Duron 700MHz with Fedora
5
for use with SIP phones and the Linksys 3102 SIP gateway (ie. no FXO card,
so no need for zaptel and libpri), but I'm stuck: The GrandStream BudgeTone
Am Montag, den 13.11.2006, 16:51 -0500 schrieb mitcheloc:
For about a year and a half now I've had Asterisk set up to unlock my
front door at my house when calling a certain number. I locked it down
by using caller id (not the most secure, but hey nobody knows the
phone number to my door).
Am Sonntag, den 12.11.2006, 08:50 + schrieb Nic Hughes:
Hi all,
I have found that when I use an announcement at the start of the call it
results in a useless answerphone message if the call goes onto
answerphone for any reason - the message being a chopped off version of
the
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the
On 2006-11-10 15:48:23 -0800, Andrew Joakimsen [EMAIL PROTECTED] said:
I am surprised that you have had good success perhaps you haven't done
proper testing?
I see you are skeptical...
I am using the Nokia e60, which also has no problems on the asterisk
side. The phone could use some
On 2006-11-10 09:10:30 -0800, Mario François Jauvin
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
I have had no success in getting the voicemail working on Asterisk
1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried
with or without ztdummy device,
Am Freitag, den 10.11.2006, 11:21 +0100 schrieb Dave Cotton:
I'm running chan_capi on a number of systems in France, France Telecom
offer the possibility of having the caller's name, but say we must
configure for EuroISDN+. Google doesn't show much and the best I could
see was in Dutch.
Am Freitag, den 10.11.2006, 12:56 +0100 schrieb Mike Heininger:
Hi!
We have an installation with WLAN SIP phones only. Sometimes we have
connection drops. What is the best way to debug if we have problems
with the WLAN or the SIP devices or the uplink to the IAX Provider.
I'd go with
Am Freitag, den 10.11.2006, 14:35 +0100 schrieb Ondrej Valousek:
Hello all,
I am just wondering - how can I implement presence awareness in Asterisk?
I know there is the hint feature that might be useful (for someone) but
it is not exactly what I am looking for.
My idea is some fairly
Am Freitag, den 10.11.2006, 00:07 -0500 schrieb Jeronimo Romero:
I’m running asterisk 1.2.8. I would like PSTN inbound calls to do the
following:
1-once PSTN callers enter their desired extension; they have to record
their name
2-recording then announces that it is trying to locate
Am Freitag, den 10.11.2006, 16:33 +0100 schrieb Ondrej Valousek:
Hi Anselm,
Yes it looks promising.
somehow update the status in the Asterisk DB
and that's the problem - how can I access Asterisk DB remotely (in some
nice and elegant way)?
That's why I was more thinking about mysql - it
On 2006-11-08 14:40:09 -0800, Ken Williams
[EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
After about one weeks time I've gone from no VoIP to a completely
configured system for two of our offices to be able to page/communicate
interoffice as well as handle existing
Am Donnerstag, den 09.11.2006, 12:19 -0300 schrieb Frederico Madeira:
I'm registering 5 lines on my asterisk box from one voip provider.
Lines;
4040.
4040.0001
4040.0002
4040.0003
4040.0004
All lines is registered in 5060 port so when someone call to 4040.0001
the call arrive on
On 2006-11-08 06:26:45 -0800, [EMAIL PROTECTED] said:
Hello,
I need (some) help in configuring PAP2.
Try looking in sip.conf
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Tom Vile wrote:
That probably because you are using Webmin. Just change the port
Webmin listens on instead, I use 9000.
On 11/6/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I'll keep that in mind for future. I read about using 10001 as start
port on Nerd Vittles website.
Is there some
On 2006-11-06 13:16:50 -0800, Christian [EMAIL PROTECTED] said:
Hi all,
DO my messages come through to the list? I have had some problems wiht
my email client here.
Looks like your spell checker has issues also...
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Am Dienstag, den 07.11.2006, 02:29 -0500 schrieb Doug Crompton:
I am trying to do something that I see describe in a book and it is not
working
In my sip.conf, I have in my [fxo] context=from-pstn
I then have in extensions.conf
[from-pstn]
exten s,1,answer()
exten
Hello *,
I recently started playing around with the SMS application. Several of
my SIP clients are FritzBoxes, with SMS capable DECT phones connected
via ISDN or analog line.
So far, SMSing works great: I defined extension 0193010[01] to receive
SMSes, which works well with the default settings
Am Montag, den 06.11.2006, 11:04 + schrieb Arik Raffael Funke:
Hi,
want to have calls directed to internal fixed phones, when my employees
are home and automatically to their IAX connection when they are logged
in remotely. How do I do this?
The picture is as follows:
--- HOME
Am Montag, den 06.11.2006, 15:14 + schrieb Scott Pinhorne:
Hi All
I am not sure what I wish to do it possible but I would like to see if
you guys know any better.
I have a site who has the extensions: 1231, 1232. 1233, 1234
Each of these users can dial each other on the extension
On 2006-11-02 07:51:15 -0800, Noc Phibee [EMAIL PROTECTED] said:
Hi
anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ?
Actually my HandyTone 488 are connected to:
wan port to my lan
line FXO port are connected to my local analogic line
i want that when a call in
On 2006-11-02 07:34:15 -0800, mail-lists [EMAIL PROTECTED] said:
snip
My question is this: How do huge voip companies like vonage handle
bandwidth. I'm pretty sure that they have to have sufficient bandwidth
available for X numbers of simultaneous calls, in other words ALL VOIP
traffic runs
On 2006-11-02 20:57:28 -0800, James Harper
[EMAIL PROTECTED] said:
We have a 100 number indial range and every so often get fax calls on
our voice numbers (our fax number isn't in the 100 number range). If you
just hang up the sending fax will often try a few times before finally
giving up.
On 2006-11-02 05:11:28 -0800, Florian Hars [EMAIL PROTECTED] said:
I've long since given up registering to bug trackers, there are far too
many of them, and I don't want to remember a username/password pair for
every program I use.
Don't complain about bugs then!
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally, but then is non responsive to commands (exit works though?).
I am currently
On 2006-11-01 08:28:28 -0800, Jason Walker [EMAIL PROTECTED] said:
Ok sorry for not being specific. I am having a problem when people
outside call in to my number which terminates at VoicePluse then The
send IAX to me and I do not get any tones. People press buttons but it
just goes to the
On 2006-11-01 10:42:12 -0800, Joshua Colp [EMAIL PROTECTED] said:
Martin Joseph wrote:
Good news!
I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems
to have resolved the asterisk hogging the whole CPU issue.
I still can't use the regular console though (asterisk -r
On 2006-11-01 09:09:26 -0800, Martin Joseph [EMAIL PROTECTED] said:
I am testing 1.4 branch on OSX (10.4.8) and although it's running and
passing calls ok, I am still not able to connect using asterisk -r.
When I do open a CLI using asterisk -r, it appears to start up
normally
On 2006-10-31 17:29:47 -0800, Brad Templeton [EMAIL PROTECTED] said:
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service. So I'm
On 2006-10-29 01:35:46 -0800, Alberto Pastore [EMAIL PROTECTED] said:
Martin Joseph wrote:
I think it's cleary true that wiring WIFI infrastructure is easier and
more reliable then WDS.
On the other hand, I have been running my little network with WDS for
over three weeks now, and it has
On 2006-10-28 03:51:57 -0700, Alberto Pastore [EMAIL PROTECTED] said:
Andrew Joakimsen ha scritto:
Are you using WDS? While it won't totally fix every issue, I've found
in my trials that turning off WDS and making sure all the AP were
connected to the same wired network was way more reliable,
On 2006-10-27 11:55:14 -0700, Andrew Joakimsen [EMAIL PROTECTED] said:
Are you using WDS? While it won't totally fix every issue, I've found in my
trials that turning off WDS and making sure all the AP were connected to the
same wired network was way more reliable, no more random
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