What I'm not trying to understand is how Ringback works in this context.
err, I mean what I'm now trying to understand.
Paul
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To U
Guys.. Im trying to setup a fotphone using iaxcomm and when I dial that
softphones extension, * complains of this:
Mar 5 01:54:54 NOTICE[9962]: app_dial.c:936 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 3)
Any hints?
___
Asterisk-U
Hi Robert,
I tried yours and Steven's scenarios and you're absolutely right. I get
ringback when the initial call takes place, but if I then try to do a
transfer to another extension after the fact I do not hear ringback on the
line. So I absolutely agree that there is a problem.
What I'm not
- Original Message -
From: "Ryan Laginski" <[EMAIL PROTECTED]>
Anyways, I haven't found anyone that offers a toll free number that
works in Canada for 1.29 cents a minute. If there is others, please
let me know.
You're LiveVoip toll free number costs 1.29 c/min from Canada? My toll free
I'll keep pointing toward the asterisk-bluetooth bounty here:
http://www.voip-info.org/wiki-Asterisk+bounty+bluetooth+cell-phone+suppo
rt
Once done, this should allow * to use a bluetooth enabled phone via a
bluetooth dongle. It'll be grand *when* it works.
Until then, you'll need to consider a
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures & the specs they seem to be very similar beasts
but the firmware is supposedly not interchangeable.
Does anyone know the difference between the 2, do they work with Asterisk?
To paraphrase:
Ignore them and they will go away.
- Original Message -
From: "David Brodbeck" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: March 4, 2005 11:04 AM
Subject: RE: [OT] - [Asterisk-Users] Why should I answer a Newbie
question,ther
I use a startup script with nothing in modprobe.conf:
#!/bin/bash
#
# System startup script for the isdn-capi subsystem
case "$1" in
start)
echo -n "Starting mISDN and CAPI"
modprobe capi
modprobe mISDN_core
modprobe mISDN_l1
modprobe mISDN_l2
modprobe l3udss1
modprobe mISDN_ca
Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,[EMAIL PROTECTED]
[EMAIL PROTECTED]
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Just what I was looking for... Thx Richard.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard J.
Sears
Sent: Viernes, 04 de Marzo de 2005 11:01 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk B
Howard.
My handytones 286 worked great with it as soon as I defined on sip.conf the
propoer context for the mailbox setting for each sip phone, like this:
mailbox:[EMAIL PROTECTED]
Then asterisk sends the MWI to the ATA and it gives you its stutter tone.
-Original Message-
From: [EMAI
Hi guys,
Sorry to bug you on this. Any ideas ? Really
stuck with this.
Hi guys/girls,
We are running a TDM04B card with Asterisk in a
Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as
an operator console. The FXO ports in the TDM04B are plugged directly in
The grandstream 101's have stutter tone.
Works great with [EMAIL PROTECTED]
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard
Lowndes
Sent: Friday, March 04, 2005 11:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Su
Digium Has a pretty nice one on their site:
http://www.digium.com/downloads/marketing/asterisk.pdf
On Fri, 4 Mar 2005 22:57:05 -0600
"Anton Krall" <[EMAIL PROTECTED]> wrote:
> Guys.
>
> Anybody has developed and asterisk brochure for commercial purposes
> (consultant, etc) that I might be abl
On Sat, 2005-03-05 at 14:10, Anton Krall wrote:
> I think I have something misconfigured regarding voicemails. They work
> great, I have this setup:
>
> Sip.conf
>
> [ext1]
> Context=phones
> Mailbox=201
>
> Voicemail.conf
>
> [home]
>
> 201,password,name,[EMAIL PROTECTED]
>
> Voicemail deliv
Guys.
Anybody has developed and asterisk brochure for commercial purposes
(consultant, etc) that I might be able to take a look at?
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The VPN approach might resolv a lot of nat issues I guess... Depending on
the scenario I guess.. You could put another * box inside the second nat and
interconnect using IAX, or if using a single phone, just use your setup, and
finally, if using 2 or more phones and cant put a second * box, well, t
Yes, only port 5060. If you do not forward 5060, you can not call this
phone
from outside. Seem to work OK without other ports being forwarded.
You mean on the remote sip phone firewall? What if there arem ore than 1
sip
phone on that network behidn that firewall?
Then you are in trouble. Asterisk
Nah conferences start when they start and finish when they finish.
Theres no need for timers etc.
The other thing that you need to implement is recording of the conference calls.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Friday,
www.mutualphone.com
On Fri, 04 Mar 2005 21:18:41 -0600, Tim <[EMAIL PROTECTED]> wrote:
> Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
> calls. Calls that are being routed to wrong numbers. DID's that ring
> busy. For the pass 2 days I am unable to pass CID. Is anyone e
Worth a try Karl... Thx! Ill let you know how it went in a few minutes.
You were right! Sip.conf needs to have the voicemail context on the Mailbox
line on each sip phone config.
Thx Guys!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl H. Putz
Sen
True. I remember it was working on time but cant remember what config it
had.
Anybody using Granstreams handytone 286 atas?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Viernes, 04 de Marzo de 2005 09:26 p.m.
To: Asterisk User
First off, let me thank Belaïd Arezqui (aka Areski) for his
PHP gui. I knew nothing about PHP last week, and the code
makes for easy editing and additions.
>Lots of interest here for conferencing.
>I've probably convinced more people to start using [EMAIL PROTECTED] for
>this feature than anythi
-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Steven
>Critchfield
>Sent: Friday, March 04, 2005 10:26 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Stutter Tone
>
>
>On Fri, 2005-03-04 at 21:10 -0600, Anton
On Fri, 2005-03-04 at 21:10 -0600, Anton Krall wrote:
> I think I have something misconfigured regarding voicemails. They work
> great, I have this setup:
>
> Sip.conf
>
> [ext1]
> Context=phones
> Mailbox=201
>
> Voicemail.conf
>
> [home]
>
> 201,password,name,[EMAIL PROTECTED]
>
> Voicemail
>I am using Polycom SP300 phones. You have to separate 'user' and 'peer'
part of it to >>>get it working. Search the wiki for description of the
problem.
Nice to know ... I don't own any of those but its good general knowledge.
>You have to forward port 5060 so that phone from outside can registe
Anyone having problems with LiveVoIP lately? I am seeing failed outgoing
calls. Calls that are being routed to wrong numbers. DID's that ring
busy. For the pass 2 days I am unable to pass CID. Is anyone else have
these problems? Can anyone recommend a Quality VoIP provider?
__
Why are the sip.conf extensions mentioned twice each?
I am using Polycom SP300 phones. You have to separate 'user' and 'peer' part
of it to get it working. Search the wiki for description of the problem.
Also, if you * box is behind another firewall, by forward ports 5060 and
1-2 and may
I think I have something misconfigured regarding voicemails. They work
great, I have this setup:
Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,[EMAIL PROTECTED]
Voicemail delivery and all works great but when I check sip extension ext1
(analog phone using
Hi ALL
I'm looking for feedback on how well this unit integrates into asterisk via an
ata. Is the audio quality any good as thats the first thing to upset the wife
if its no good.
I'm looking for a "reasonably priced" GSM gateway 1800mhz for use in Australia
that works with an ata. Quite happy
Why are the sip.conf extensions mentioned twice each?
Also, if you * box is behind another firewall, by forward ports 5060 and
1-2 and maybe 5004 from the firewall to the * box will that help on
the NAT issue?
If phone 2 is behind another firewall, do you need to forward port 5060 only
t
Guys, this error has been driving me nuts and I see no indication anywhere
as to what it may mean.
Anybody has any clues on this?
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
-- Playing 'vm-review' (language 'en')
-- Saving message as is
-- Pl
Lots of interest here for conferencing.
I've probably convinced more people to start using [EMAIL PROTECTED] for
this feature than anything else.
Can I input some suggestions;
Need to change the rinky dink call icons (let me know if you need some
better samples)
Need to change the ability for a
Hi, all
This is the souktion that worked for me.
Here is my config again
PHONE 1 -- * BOX
|
NAT/Firewall
|
|
NAT/Firewall
|
|
PHONE 2
Firewall on Asteri
On Fri, 2005-03-04 at 18:17 -0800, Dan Austin wrote:
> There does not seem to be too much interest in this, but it has
> helped me sell the idea of dumping a very expensive, but poorly
> functioning, existing VoIP conferencing system. In the future
> I can send announcements directly to the few p
I have a queue and some agents. The agents are not logged in but are
members of the queue.
Now, joinempty=no I would assume that you cannot join this queue but will
fall into the goto(queuefail|1|1). This is not so. I join the queue just
fine.
If I remove the three members from the queue (comm
Last update before I head to Korea for an IP phone deployment
New diffs and PHP web interfaces at www.fitawi.com/Asterisk
-New in the interface:
New fields for conference conference owner and title
Delete conferences
Changed date/time to listbox to reduce user input error
Hi,
I have been using/working on asterisk for some time now and presently
was trying to configure asterisk to work with digium cards. It worked fine
with the fxo/fxs cards, but now i'm trying to get it working by interfacing it
with mediant t1 port. no avail ...
anyone out there got it w
Thanks for the tip! I will get a phone that supports 3 way calling to
test the agi script again, as those that I have now currently do not
have support for it
On Fri, 04 Mar 2005 08:57:56 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> On Fri, 2005-03-04 at 22:31 +0800, mechaman wrote:
> >
I’ve had the same
problem..
Same error and everything..
Del your mpg123 version u
got now..killall -9 and rm –rf the folder and do this:
Wget http://www.mpg123.de/mpg123/precompiled/mpg123-0.59q-1.i386.rpm
rpm –Uhv mpg123-0.59q-1.i386.rpm
Should fix your issue
Grüsse
Most T1 circuits are delivered via HDSL2 these days. Hence the single
pair.
On Mar 4, 2005, at 5:07 PM, Tom wrote:
Thanks for the quick reply,
we didn't reboot we'll try that, and I've been planning on building a
new
kernel, I know the fc kernels have issues... I'll report back after I
try thes
In a word - No. Generally, BT-capable phones can only control a headset
or handsfree-set, but not be turned into a headset themselves. It's
akin to expecting to watch TV on your remote, as it controls the TV so
nicely :)
There is, however, an effort to have asterisk become the headset to a BT
ca
John,
Basic intelligence would tell you that if
you have a single pstn phone line you will only be able to get a single phone
call occurring over that phone line correct?
If you want to have a conference call with
multiple people all dialing in then you would need multiple lines.
I
I am brand new to Asterisk. My question is if I
want to have multiple participants all listening, or listening and talking, do I
need to have a separate telephone line for each, or can they all dial in using a
single telephone number and a single line?
Thanks,
John Fistere
What mobile phone do you have Chris?
Is it able to transport voice over bluetooth to another switch? Or only
able to receive Bluetooth from a paired headset?
Do you use a Bluetooth headset for this handset already?
Is it capable of pairing to more than one Bluetooth device?
Can you sync this heads
Did you install the version of mpg123 that is 59r and not
59q or 59g?
This
is a problem with version of mpg1223 almost assuredly.
Did you install just as the installation says to do?
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf
More stuff..
http://www.voip-info.org/tiki-
I don't want to use a bluetooth headset but instead want to use my
bluetooth enabled mobile phone as a SIP phone for asterisk. Is that
possible?
Chris
On 4 Mar 2005, at 19:11, Linn Boyd wrote:
Chris,
I will take your $100.00 bounty :-D I am using a bluetooth headset
with firefly and my lapt
Nigel,
I have
bugetone phones working with 2, 3, 4 + extension numbers.
Check
you config's, or post them here and lets see if we can find the
problem.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Nigel
BurgessSent: 04 March 200
I wonder if you could share your configuration (sip.conf and
extensions.conf) on handling incoming calls from VoipLive, since I'm
trying to set it up also.
Thanks a lot,
Roman Zhovtulya
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Fielding
Sent
Tom wrote:
we didn't reboot we'll try that, and I've been planning on building a new
kernel, I know the fc kernels have issues... I'll report back after I try these
two things.
FYI We are running asterisk with a PRI on FC3 with no problems
Trev
___
Ast
Yes, I've had this before with sipgate.
Try using either "31557110304" or "557110304" in both places in:
register => 31557110304:[EMAIL PROTECTED]/557110304
And use use this number as a context for incoming calls
What also might work: the incoming number (557110304) in
register => 31557110304
Hello,
I am quite new to asterisk (I've been playing
with it for just about 2 weeks).
I am trying to do music on hold, but I get this
error:
res_musiconhold.c:309 monmp3thread: Request to
schedule in the past?!?!
I have read on some forums that usually
this message comes when all resou
Thanks for the quick reply,
we didn't reboot we'll try that, and I've been planning on building a new
kernel, I know the fc kernels have issues... I'll report back after I try these
two things.
On another note we just did some wire tracing and it might be an issue of
wiring... We have an adtran ca
Hi all,
I am having problems with hangup detection on fxo line.
In Turkey, we don't have polarity reversal, so hanguponpolarityswitch option
in zapata.conf does not help.
The callprogress in dsp.c is just for the US, and the way it does call
progress is quite unsuitable for the tones in Turkey,
Could you also post your extensions.conf where to route the call
further?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Woojin Lee
Sent: Freitag, 4. März 2005 16:38
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice + incoming cal
Areski wrote:
Dear ALL,
As everybody seems to like very much "Asterisk-Stat",
I decided to make couples of improvements...
so here we go with a new version :D
FEATURES :
- CDR report (monthly or daily)
- monthly traffic reports (pie graph)
- DAILY LOAD !!!
- compare call load with previous days
I installed capi support,
For HFC PCI card i must use Capi4Linux o Isdn4Linunx
(my kernel is 2.4, i dont support mISDN)
How install HFC Card ? How see HFC Card PCI ?
(my card is chip billion s0 isdn card)
Help me
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Aste
Go to www.voip-info.org and search on the word CDR
http://www.voip-info.org/wiki-Asterisk+CDR+Areski+GUI
You can also search on the word "queue"
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Scully
Sent: Friday, March 04, 2005 3:37 PM
To: aster
On Fri, 2005-03-04 at 14:21 -0700, Wiley Siler wrote:
> Also lookup AGI
>
> The WiKi and via google by using this: site:lists.digium.com words>
>
> W
>
Thanks Wiley, for the pointer API should do it.
Found this webpage via wiki it should help me start
http://home.cogeco.ca/~camstuff/agi.html
I have not found anything in the docs on real capacity in terms of number of
people, PRIs or calls per hour.
What is the largest installation of * in use today? I don't mean
theoretical capacity, but actual installed systems.
I am an old "big iron" call center guy, and used to run several 1,000
hi * users,
due to the fundamental code changes in cvs tonight, it was necessary to
update chan_capi to the new channel_tech design.
it completely replaces my former patch from november 2004.
the patch can be downloaded at
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
regards
Hello - I have just joined the lists and am considering installing quite a
few * systems.
I am looking for an IP-PBX with both solid standard features and
call-center/ACD features.
I have read the documentation and the list archives and did not see any
references to real call-center type reportin
On Fri, 2005-03-04 at 15:27 -0700, Tom wrote:
> Hello,
> I have searched and searched, and come up with nothing. I am running Asterisk
> with a wcte110p configured for t1. Our PRI is staying up, and we can make
> calls however our service provider's logs are flooding with errors and we are
> gett
Hello,
I have searched and searched, and come up with nothing. I am running Asterisk
with a wcte110p configured for t1. Our PRI is staying up, and we can make
calls however our service provider's logs are flooding with errors and we are
getting lots of HDLC Abort (6) on Primary D-Channel Errors.
That was the fix.
For other people who might be having this problem, I am now using:
exten => _8NXXNXX,1,Dial(Zap/g1/ww${EXTEN:1})
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Friday, March 04, 2005 1:49 PM
To: Asterisk Us
On Fri, 4 Mar 2005 13:40:25 -0800
"Jeff Busch" <[EMAIL PROTECTED]> wrote:
Thought I had this fixed, but it turns out it is not.
I've been
wracking my brain. Here is what I have done:
- Tried 3 different Qwest PSTN lines (just in case it
was a line issue)
- Tried calling same number from an anal
And here...
www.digium.com (see documentation link)
And when you have some early questions look here...
www.google.com enter this: site:lists.digium.com
And here
www.asterisk.org
Read as much as you can...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTEC
I've called using G729 SIP phones over my LAN, and I think it sounds
quite good. YMMV.
On Fri, 04 Mar 2005 15:58:23 -0500, Martin Roy <[EMAIL PROTECTED]> wrote:
> I have 2 Asterisk servers connected with IAX. It's working fine I can
> call an extension from one phone in an office to another phone
Hi Steve,
> -Original Message-
> > I am having a problem with periodic breaks in audio over an
> IAX trunk.
> > The interruption only happens in one direction, and (I think) only
> > with clients built on the open source libiax.
> >
> > Codec is irrelevant, and jitterbuffer on/off seem
You have to bring this up with broadvoice.
On Fri, 04 Mar 2005 15:39:07 -0500, Randy Johnson
<[EMAIL PROTECTED]> wrote:
> I set up an asterisk box with a broadvoice sip connection for incoming
> connections
>
> it works great when I use a cell phone, vonage line, calling card to
> call the aster
Thought I had this fixed, but it turns out it is not. I've been
wracking my brain. Here is what I have done:
- Tried 3 different Qwest PSTN lines (just in case it was a line issue)
- Tried calling same number from an analog phone plugged directly into
the Qwest line - NO PROBLEMS.
- Because of t
Pull up a chair and start reading:
http://www.voip-info.org/tiki-index.php?page=Asterisk
-Matthew
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Friday, March 04, 2005 3:34 PM
Subject: [Asterisk-Users] Im a noob
> Im completly new to the whole PBX thing. I have a toshiba un
Yes it does support a basic analog line (or many many lines...). It also
supports T1's, ISDN, etc. FXO would provide an analog connection to the phone
company (your wall jack)
FXS would allow you to plug analog phones into Asterisk.
Phone <--->(FXS)<--->Asterisk<>(FXO)<>Phone Company
Y
I am pretty sure the answer is yes.
Umar
On Fri, 4 Mar 2005 00:33:46 +0100, Robert Rozman <[EMAIL PROTECTED]> wrote:
>
> - Original Message -
> From: "Umar Sear" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, March 03, 2005 11:0
Yes, they do blame everyone else. There is another thread where I
posted that I couldn't get my toll free number working. I waited days
for support to get back to me, and I ended up emailing this list and
then livevoip again. A LiveVoip representive blasted me on this list,
stating that I was handl
Im completly new to the whole PBX thing.
I have a toshiba unit now and we're moving our office in the next few months.
I want to use asterisk but would like to test it out first. Does it support
a basic analog phone line like the one in my house? Is that FXS? Are there
any FAQs I should read to le
I'm looking to purchase and deploy a bunch of hardphones for agent
use. The phones will have to register with Asterisk and/or SER,
depending on where the phones go. They need only one line, G729 codec,
and no super fancy features. Preferrably something that is easy to
provision.
I would think the
What is your price range is the question. BudgetTones are OK but have
some limitations. Polycoms are my choice for around $160-$200. Ciscos
work well for some people too. Just a matter of how many dollars. The
budgettone lack of 3-way can be gotten around with a proper dial plan.
W
-Origi
Also lookup AGI
The WiKi and via google by using this: site:lists.digium.com
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, March 04, 2005 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Fri, 4 Mar 2005, Randy Johnson wrote:
> I set up an asterisk box with a broadvoice sip connection for incoming
> connections
>
> it works great when I use a cell phone, vonage line, calling card to
> call the asterisk box, but when I try to call it from our verizon land
> line it is busy and as
Hi,
Can anyone confirm that if I want to do h323 proxying that I do not need
codecs installed? For example if the codec being used is g723.1, I don't
need the codec installed locally because there is no compression or
decompression being done on my server; the incoming traffic is simply
being s
On Fri, 2005-03-04 at 13:58 -0700, Joseph wrote:
> On Fri, 2005-03-04 at 14:45 -0600, Steven Critchfield wrote:
> > On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote:
> > > Is it possible to dial number from the command line and passing the
> > > connection to one of my extension (or speakerphone) if
Asterisk Flash Operator Control Pannel
-Matthew
- Original Message -
From: "Will McCown" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, March 04, 2005 2:45 PM
Subject: [Asterisk-Users] Options for Attendant Console.
> We've been playing
Due to management concerns our asterisk system has been setup to record
all phone calls for some time now (before the 1.0 release). Everything
was working fine until we upgraded 1.0.5 where all calls are recorded
except those that pass through a queue (we are not using the queue
record functio
If using MySQL go get phpMyAdmin. Its not "for" asterisk but I use all the
time.
-Matthew
- Original Message -
From: "Kanishka Somaratne" <[EMAIL PROTECTED]>
To:
Sent: Friday, March 04, 2005 1:43 PM
Subject: [Asterisk-Users] Web based tool asterisk real time
Is there a webbased tool t
I have 2 Asterisk servers connected with IAX. It's working fine I can
call an extension from one phone in an office to another phone in the
other office. The only problem I have is lagging. What codec should I
use? I have Cisco phones 7960 on both end. Currently in the IAX trunk I
configured it
On Fri, 2005-03-04 at 14:45 -0600, Steven Critchfield wrote:
> On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote:
> > Is it possible to dial number from the command line and passing the
> > connection to one of my extension (or speakerphone) if the other party
> > answers the call?
> >
> > I was thi
On Fri, 2005-03-04 at 12:44 -0800, asterisk phones wrote:
> Has anyone done Voice Over Frame Relay with Asterisk.
> With Frame Relay work reliably with Asterisk? Any
> experiences?
Doesn't look like you visited google first. Nor did you bother to look
at the code.
channels/adtranvofr.h
[EMAIL
Has anybody got the
Early Dial feature working on Asterisk with a grandstream phone
?
I can do two digit
dials eg 12 and it works fine. When I press a 3rd digit I get a busy
response. I did add the auth=plain text in my sip.conf file but to no
avail.
I have also done the
redirect line
On Fri, 2005-03-04 at 13:41 -0700, Joseph wrote:
> Is it possible to dial number from the command line and passing the
> connection to one of my extension (or speakerphone) if the other party
> answers the call?
>
> I was thinking of implementing this sort of feature with and accounting
> applicat
We've been playing with Asterisk with an eye towards possibly
replacing or augmenting our existing PBX serving about over 600
phones (and needing to expand). The one missing bit that I can't
find any mention of is an Attendant Console. Are there any
good solutions out there?
I've considered that
Has anyone done Voice Over Frame Relay with Asterisk.
With Frame Relay work reliably with Asterisk? Any
experiences?
__
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Is it possible to dial number from the command line and passing the
connection to one of my extension (or speakerphone) if the other party
answers the call?
I was thinking of implementing this sort of feature with and accounting
application. The customer phone number is in the database, so clicki
I set up an asterisk box with a broadvoice sip connection for incoming
connections
it works great when I use a cell phone, vonage line, calling card to
call the asterisk box, but when I try to call it from our verizon land
line it is busy and asterisk logs do not show incoming call.
Any ideas
> I would think the BudgeTone would be good, but then I've read
> so many people complaining about them, and some people seem
> to recommend the Sipura adapters.
For agent use, the BudgeTone's lack of three-way calling would be an
issue.
Nabeel
___
Aste
I'm looking to purchase and deploy a bunch of hardphones for agent
use. The phones will have to register with Asterisk and/or SER,
depending on where the phones go. They need only one line, G729 codec,
and no super fancy features. Preferrably something that is easy to
provision.
I would think the
I have a new HP IpaQ 6315. I run SJPhone on it with a bluetooth headset. Works
great!
Paul
paul mahler
www.signate.com
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matthew Boehm
> Sent: Friday, March 04, 2005 11:23 AM
> To: A
I've patched the chan_capi to let it compile under the new CVS head
Give it a try please
You have to start from the original chan_capi
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
and then apply the patch
http://www.c-net.it/chan_capi.diff.bz2
it also includes the fax patch fr
I know that there are some patches being worked on to cache realtime
users that might ultimately fix this problem, but until then, here is a
little script that brings back the MWI when using the excellent mysql
realtime architecture with sip:
http://www.cheapnet.net/~mike/asterisk/send_mwi.txt
Stuart Ford wrote:
Seriously, this has to be the simplest NAT problem there is with
Asterisk. What's the secret? How do I learn the dark art? What am I
missing?
I'm guessing here, but the NAT'd grandstream does not have the correct
external IP configured.
The phones are trying to establish a d
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