After setting the pridialplan=unknown I seeing the Called Number TON change
to Unknown Number Type but not the Calling Number TON. Should both be
following this parameter or not. If not is their another option to change
the Calling Number TON?
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT
When I first started looking at a similar problem started out on the same
path with app_queue but even having access to a friend of mine who actually
help write some of the Queue code we decided it wasn't the right tool for
the job. We approached this a little differently using valetparking and
Just woundering if the intentend functionality of leastrecent and
fewestcalls it to continually dial only the first chosen ext. in the queue.
In other words if a memeber is logged into the queue but doesn't answer the
call the call never moves on in my configuration from that ext. This could
be
Title: Channel Banks
Well, you have several options. A T100P and a
device such as a Adtran Altlas or simpler Channel bank. But since at this time
as you point out Digium only has 1 FXOport per PCI slot(FYI I hear they
are working on a 4 port per PCI slot). The other options are MediaTrix,
Works okay but user interface is a little like using RegEdit to program your
router.
In the version of software the one I have it lack some security features and
I am unable to find any DMTF controls
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list
I have had several installations where I was unable in any configuration to
make the FVS318 work with VOIP traffic. I don't belive it is related to any
paticular Phones or VOIP GW have see same problems with even Cisco 7960's
Has anyone opened a ticket with Netgear on this issue?
- Original
This is my working config for x100p tdm400 so if you change the channel
entries from 2-5 to 2-3 you should be good to go.
/etc/rc.d/rc.local
modprobe wcfxo
modprobe wcfxs
/usr/sbin/asterisk
/etc/zaptel.conf
fxsks=1
fxols=2-5
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf
[channels]
Title: Message
I don't really have a answer for you on you issue
but have a question about what "find-me" is. I see it on the feature list but am
unable to find any real information about it. Is this simply call forward or is
their more to it.
thanks
- Original Message -
Anyone have any
experienceconfiguringVegaStream's with Asterisk.
Ihave run
into afew of questions. 1. It appear that after turning on
registrations I am seeing two request for registration per
linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is
purpose and how do I handle this?2. DTMF
Please include your sip.conf and extension.conf files. Hard to say what is
wrong without seeing the configuration
- Original Message -
From: Bill Michaelson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 09, 2004 3:15 PM
Subject: [Asterisk-Users] asterisk-grandstream call
I have had similar issues with mine TDM400 w/4
modules. I get both no dial tone and sometime a large level of static on the
port and although sometimes manually unloading and reloading the drivers will
correct the problem most of the time I have to reboot the system. Also, do you
get
Are their any options for ADSI capitableports
indenisties of 20 to 50 ports that will work with asterisk and ADSI
phones.
Are there any special feature that the channel bank has to support? or as
long and it is connected a Wildcard T1 port it should fine with any channel
bank?
Thanks
- Original Message -
From: Jon Pounder [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 18, 2004 5:23 PM
I could be wrong but I think I remember seeing mention of recommendation
about the number per server although I don't remember the number.
- Original Message -
From: Christian Hecimovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 19, 2004 12:26 PM
Subject: Re:
I seem to be having trouble with cvs login. anyone
having similar problems
It just hangs after entering the
password
I have been doing the following and it seems to work fine# cd /usr/src#
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot# cvs login -
the password is anoncvs.# cvs checkout zaptel libpri# cvs checkout -r
v1-0_stable asterisk This will create just the asterisk directory.
Can anyone verify this?
I have 2 voicemail context and when using the Directory app I seeing odd
results.
If I spesify the context as (default) I can only access default context
users as expected and it uses default extension.conf context to dial
If I specify the context as (group1) I can access
I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted the
modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with
kernel 2.4.20-28.9 and all is working fine. I didn't have time to work in
the issue but have seen it on two systems.
Anyone have any idea what the
kernel that work is 2.4.20-20.9 not kernel 2.4.20-28.9(haven't tested)
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 9:45 PM
Subject: [Asterisk-Users] Redhat lastest kernel problems with Zaptel drivers
kernel 2.4.20-30.9
I updgraded to kernel 2.4.20-28.9 to kernel 2.4.20-30.9 and my digium card
drivers refused to load I then rebooted with the previous kernel and all
work fine. Not sure if it is related or not thought it might help
- Original Message -
From: Michael Zheng [EMAIL PROTECTED]
To: [EMAIL
Okay, I need to sleep. I just need to recompile the drivers!!!
Sorry for the false alarm
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:05 PM
Subject: Re: [Asterisk-Users] Redhat lastest kernel problems with Zaptel
Below are some links that should point you in the right direction. Assuming
you don't have any IP Phone on hand I would recomment starting with to
computers with softphone(one example
http://www.eutecticsinc.com/download6/DISK1/IPP200_SJSoftPhone.htm) and
have them talk to each other.
I have an IconnectHere account
with a Inbound number and have setup the sip.conf to register and am recieving
the call but When I answer the call it disconnect. I have tried sending the call
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon
as I accept the
; SIP Configuration for
Asterisk;[general]port =
5060
; Port to bind tobindaddr =
0.0.0.0
; Address to bind tocontext =
sipinbound ;
Default for incoming callsregister =
1410344:[EMAIL PROTECTED]/1410344
--=-=-=-= extentions.conf-=-=-=-=-=- have also
tried sip phone same
www.diguim.com
Wildcard T100P - Single t1
http://www.digium.com/downloads/product_sheets/T100P.pdf
Wildcard TE410P - 4 port - I believe support independant config of each T1
http://www.digium.com/downloads/product_sheets/TE410P.pdf
Wildcard T400P - 4 port -
I have an IconnectHere account
with a Inbound number and have setup the sip.conf to register and am recieving
the call but When I answer the call it disconnect. I have tried sending the call
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon
as I accept the
Title: Leterhead
What do you have configured in your
/etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci
this should show adevice Tiger Jet Network Inc. if the pci bus recognized
the card.
- Original Message -
From:
David J Carter
To: [EMAIL
I am try to come up with a way to dial multiple ext
and require one or more of the extension to require input before actually
bridging the calls.
Example:
Acall from PSTN into * a match is made in
extensions.conf and *then dials a
local(fxs) ext 1 and a Cell Phone
If ext 1 picks up *
No this most likely willn't work unless you have open the correct ports on
each NAT device. The problem is that NAT in general only allows packet in if
a packet has gone out first. I am assuming you have left have the fact that
* is used to setup the SIP call setup and then should drop out. If so
You could also look at products like
http://sales.netxusa.com/vegastream/vega50.php
- Original Message -
From:
Andy
Hester
To: [EMAIL PROTECTED]
Sent: Thursday, October 30, 2003 3:46
PM
Subject: RE: [Asterisk-Users] Newbie
hardware question
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
have asterisk on public side and phones on the private side. I am able to
get the phones to register and make outbound calls but the inbound calls are
intermittent. I have NAT enable in asterisk and on the Cisco 7960.
Any
Does anyone know if it is possible to force a extension to not allow
transcoding? If you spec canreinvite=yes the cal may still transcoded if the
parties do not choose a the same code on each end. In my situation it is
better that the call fail than have it transcoded.
Also, I see some limited
I think this is what you are looking
for
Exten = 1000,1,Answer,1Exten =
1000,2,Wait,1Exten = 1000,3,Voicemailmain([EMAIL PROTECTED])
- Original Message -
From:
Mitchell S. Sharp
To: [EMAIL PROTECTED]
Sent: Monday, April 05, 2004 5:27
PM
Subject: Re:
When a call is place to xxx9931211 from the pstn the call proceeds normally
until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of
call being sent with INVITE sip:[EMAIL PROTECTED] SIP/2.0. It gets
sent with INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 and this seems
to cause
. Johansson [EMAIL PROTECTED]
To: Glenn Dalgliesh [EMAIL PROTECTED]
Sent: Tuesday, April 20, 2004 1:29 PM
Subject: SIP re-invite
Could you please test this with my chan_sip2. I have a hunch it will work
with
that channel.
/Olle
___
Asterisk-Users mailing
build_user: Ignoring
username in user definition of 555
Apr 20 16:16:48 WARNING[1184099120]: chan_sip2.c:8293 build_peer: Ignoring
unknown option type
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: Olle E. Johansson [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday
SDP question if * recieves a=rtpmap:103 telephone-event/8000 it shouldn't
it send out the same a=rtpmap:103 telephone-event/8000 to the other side
of the connection? and not something like a=rtpmap:101
telephone-event/8000?
Thanks
___
Sent this to grandstream support but has anyone else seen this issue. All of
the previous sdp rtpmap are correct until the grandstream sends this. I have
been using disallow=gsm and canreinvite=no to get around the problem.
- Original Message -
From: Glenn Dalgliesh
To: [EMAIL PROTECTED
Please reply with sip.conf extension.conf for both servers. Hard to tell
what the problem is without see config info
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 11:39 AM
Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk
I want to dowload cvs of v1-0 with cvsup and was wondering what the options
file will look like to make this happen.
I am assuming the some thing on the line *default release=cvs tag=.
- options file for cvsup to download cvs head
*default host=cvs.digium.com
*default
First Thanks to brian for work on valetpark it seems to work really well
I was working on some apps using ValetParking and having good success but
was wondering when you think valetparking will make it into the
CVS/releases? So, I can build around it with a little more confidence.
Thanks
Yes,
- Original Message -
From: Neil Cherry [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, October 13, 2004 10:54 AM
Subject: [Asterisk-Users] Telco POTS - FXO ?
Maybe I'm just doing this wrong. Is the FXO card (X100P)
I want to authenticate against the asterisk internal database but don't seem
to be able to figure out the syntax for the Authenticate cmd. I am assuming
I have something wrong in line s,4
-- Executing Authenticate(SIP/5006-a54e, /iaforward/5001pass|d) in new
stack
-- Playing 'agent-pass'
Does anyone have a copy of lastest source I seem to have delete my copy and
http://www.bkw.org/app_valetparking.c seem to not exits at the moment.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Does anyone that the source for app_valetparking.c
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Symptom: Phone after about 15mins
will stop functioning
Problem: DHCP lease renewed but default route
dropped
Fix: Assign a static ip and problem
is resolved. Upgrade to new firmware once it is
released
It turn's out thatthese phones have a few
issue in 1.0.3.81 firmware. Thephone may
, 2003 at 10:42:02AM -0500, Glenn Dalgliesh wrote:
Symptom: Phone after about 15mins will stop functioning
Problem: DHCP lease renewed but default route dropped
Fix: Assign a static ip and problem is resolved. Upgrade to new
firmware once it is released
It turn's out that these phones
I have been doing some testing and have found issue
with certain devices and negotiating codecs in doing this Ihave noticed
something that seems peculiar to me. It seems that including allow=all
yields different results than having no disallow or allows in the sip.conf.
Could someone please
What version of the BudgeTone software are you
running?
- Original Message -
From:
Chandra
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 12:09
PM
Subject: [Asterisk-Users] Grandstream
budgetTone registration time out
hi,
i have been using
It looks like you have you * on public IP and your phones on private, most
likely behind NAT if so in your sip.conf under each [grandstreamX] you most
likely need: nat=yes
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:44
I am looking for a way to Forward to a external or
internal number and require a digit(s) in order to complete
forward.
Example:
PSTN1Calls * dialsPSTN2
ifPSTN2pressesproper digitsbridge the PSTN1 and
PSTN2
if
no response return to a context
Reasons: 2 actually
1. call is forwarded to
Sorry, If this is a dual post, was having trouble
with email.
I am looking for a way to Forward to a external or
internal number and require a digit(s) in order to complete
forward.
Example:
PSTN1Calls * dialsPSTN2
ifPSTN2pressesproper digitsbridge the PSTN1 and
PSTN2
if
no response
I am looking for a way to Forward to a external or
internal number and require a digit(s) in order to complete
forward.
Example:
PSTN1Calls * dialsPSTN2
ifPSTN2pressesproper digitsbridge the PSTN1 and
PSTN2
if
no response return to a context
Reasons: 2 actually
1. call is forwarded to
It appears that when realtime is enabled in queues.conf
persistantmembers no longer has effect on dynamically added members. I am
wondering if this is a intended or a bug.
___
--Bandwidth and Colocation provided by Easynews.com --
I would make the following changes to your sip.conf and restart asterisk. I
have seen alot of issues with regard to codec with * and grandstream. both
have a little to do with it but this should keep it working.
[general]
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
-
Well, you might be better off at that scale to use a cisco as5850 or equiv
with SER and Asterisk. I might not work so well with 672 calls going thru 1
asterisk box.
ds3 - Cisco as5850 - Asterisk (Possible multiple depending on actual
config and use)
- Original Message -
From: Marcelo
Well, I know to be compatible with porta-billing you need password to do ip
based auth. It's a bit goody but they basically seem to expect
if trusted ip and no Digest support then radius auth has username=src_ip and
password=x.
To put it another way it would be help full to porta-billing
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