Hi,
I am interested about that, too. If someone have some more informations...
Greg
Hi,
Using call files, is there a way to identify no answered calls from
disconnected numbers (no longer in service). Both return the same value
and so far I can not find a way to know one from the other.
http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels
On 12/6/06, Apesys [EMAIL PROTECTED] wrote:
Hi everybody.
It is possible to announce the parking position through a paging to a
group of extensions?
I would like that when someone parks a call, some phones will announce
(Posting 2nd time - I've had this problem on both my 1st and 2nd
installation of asterisk)
Hi,
In short - Asterisk is not able to recognize that the 'other' person to whom
call was made (using ZAP channel) has hung up - hence the channel stays busy
and unusable. This is when zone is set to
On Wed, Dec 06, 2006 at 10:12:25PM -0600, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
into some gotcha down the road where there is some missing file that
needs to be put who knows
Hi everyone!
I'm having an issue calling the Numbers Information Service (similiar to 411
in the US) in my country.
I use: TE110P, connected to a PRI line running on an E1.
Besides that specific number, all calls pass through fine, in and out, no
problems whatsoever.
I called my Telco, and
Doug Crompton wrote:
I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it
to a spare serial port on my linux server (asterisk resides there) and
implemented with some mods the code mentioned earlier
http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world
I decided to write my own simple voicemail application via AGI and store all
voicemails in MySQL. The nice thing was the user can retrieve via phone (local
and remote), via email attachment and also via web download.
You can listen to old and new messages and change your outgoing message too.
Hi all,
I have a question: how to configure Asterisk to support SIP INFO method?
I encountered this problem when I find my UA don't send INFO message to
another UA, actually it should. Asterisk was used as a SIP proxy in this
scenario (I know that SIP is not a SIP proxy:-)).
Then I captured
Hi all,
I would like to know if it exists the possibility to send to different
context according to the caller IP Addres
I receive H323 calls, and I have to route this to different devices
according to the caller ip.
I tried to use the
context=first-context
alias=99
context=second-context
I have logs full with this messages...
I must have qualify turned on, because phone is behind firewall,
main problem si, that phone is each hour about one hour unavailable! :'(
I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing
help me.
I'm using latest firmware 8.4 in phone,
Hi (Armin?) !
Today I had a problem with Diva Server 4BRI-8M 2.0.
Asterisk 1.2.12.1
chan_capi-cm-0.6.5
divas4linux-melware-3.0.f-106.622-1
Asterisk could not receive and make calls on the BRI ports, although the
ports looked fine within Asterisk.
I usually use /usr/lib/divas/divactrl
On Dec 7, 2006, at 4:14 AM, Jon Farmer wrote:
I decided to write my own simple voicemail application via AGI and
store all voicemails in MySQL. The nice thing was the user can
retrieve via phone (local and remote), via email attachment and
also via web download.
You can listen to old and
OOH323 Debugging Enabled
-- Executing Answer(SIP/3513-090f7d40, ) in new stack
-- Executing Wait(SIP/3513-090f7d40, 1) in new stack
-- Executing DeadAGI(SIP/3513-090f7d40, a2billing.php|1) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
a2billing.php|1:
Help me ooh323 core dumped.
2006/12/7, Ümit AYDINLI [EMAIL PROTECTED]:
OOH323 Debugging Enabled
-- Executing Answer(SIP/3513-090f7d40, ) in new stack
-- Executing Wait(SIP/3513-090f7d40, 1) in new stack
-- Executing DeadAGI(SIP/3513-090f7d40, a2billing.php|1) in new
stack
--
Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360),
I noticed my client is having some form of feed back on the phone.
Because of Snom's inner oddities this is how I got it to work.
Plantronic -- RJ11 -- SnomHandset Port (on Snom Base)
Handset -- Plantronic jack (bottom
Hi list,
Can anyone who has successfully ran asterisk on a home router please give me
the modell number as well as how they did it ?
Thanks.
Dovid___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE
On Wed, 2006-12-06 at 23:51 -0700,
[EMAIL PROTECTED] wrote:
Date: Wed, 06 Dec 2006 22:37:01 -0500
From: Steve Prior [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Is there any Asterisk controllable
thermostat?
To: Asterisk Users Mailing List - Non-Commercial Discussion
JO == J Oquendo [EMAIL PROTECTED] writes:
JO Plantronic -- RJ11 -- SnomHandset Port (on Snom Base) Handset
JO -- Plantronic jack (bottom base in the front) If I placed
JO Plantronic(RJ11) -- Snom's Headset port, the auto lift on the
JO Plantronic wouldn't work until the person pressed the
Hi,
the german telco Colt Telekom has assigned the phone number block 56830-xxx to
one of our customers. In the diaplan we have setup extensions like the
following ones:
exten = 56830910,1,Answer()
exten = 56830910,2,Dial(SIP/bduerring,10,tr)
exten = 56830910,3,VoiceMail,u20
exten =
On Thu, 7 Dec 2006, Klaus Darilion wrote:
Hi (Armin?) !
Today I had a problem with Diva Server 4BRI-8M 2.0.
Asterisk 1.2.12.1
chan_capi-cm-0.6.5
divas4linux-melware-3.0.f-106.622-1
Asterisk could not receive and make calls on the BRI ports, although the ports
looked fine within
Hello,
I try to use the background cmd for send incomings call on dial plan.
I try in an internal number for resting:
exten = 405,1,DigitTimeout,5
exten = 405,2,ResponseTimeout,10
exten = 405,3,Background(vm-accueilcreat)
exten = 1,1,Goto(creat-in,s,1)
exten = 2,1,Dial(IAX2/301,15,tr)
exten =
If you want a standardized ivr ui pattern, wouldn't something like VoiceXML be
interesting?
That's a standard for use with IVR applications.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: 7. december 2006 15:53
To:
It may not be what you're thinking, but I use Astlinux on an older PIII.
With a couple of options it has become my home router and works very well.
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote:
Hi list,
Can anyone who has successfully ran asterisk on a home router please give
me the modell
John,
Two questions on your comments
I have no seen an Insteon computer controller similiar to the old bottle
rocket. Is there such a device? I am thinking of getting an Insteon
starter kit bit I have so many X10 devices it will be awhie before, if
ever, that I get it all changed over.
Benny Amorsen wrote:
If auto lift means the mechanical lifter, then you should not use
the head set jack on the Snom at all.
/Benny
I don't follow... Remove the mechanical lifter? Then do what, go from
the Plantronic to the headset jack on the Snom, leave the receiver in
its normal
Has anyone done any fax machine detection on outbound calls? I've heard
of NV's fax detect app but I haven't seen any indications that it
supports outbound fax machine detection.
-MC
___
--Bandwidth and Colocation provided by Easynews.com --
I am experiencing this:
1 - A,B,C are SIP users logged on QUEUEA with ringall strategy
2 - I call QUEUEA
3 - A,B,C start ringing
4 - nobody answer
5 - D logs on the QUEUEA
6 - D doen's receive any call, but A,B,C are still ringing
How can i avoid that?
I'd like that when D joins the QUEUEA
Thanks guys for all the help. For this setup I just did a GoToIf(), I will
look into multiple context though, looks like thats whats needed for having
alot of different outbound caller ids!
Thanks again!
On 12/6/06, C F [EMAIL PROTECTED] wrote:
Asterisk supports whats called context, using a
JOB DESCRIPTION:
We are looking for an Engineer/Sales Engineer combination. Primary
focus will be working with Asterisk/Linux and VoIP. Asterisk systems
administrator and experience with Carrier/Service Provider
Telecommunication experience. Talented individual with a thorough
knowledge of
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 05, 2006 10:56 AM
Subject: [asterisk-users] SER/OpenSER + Asterisk + Queue
We are in the process of redesigning our single Asterisk server that
handles several queues for our
On Thu, 2006-12-07 at 07:20 -0700,
[EMAIL PROTECTED] wrote:
Date: Thu, 07 Dec 2006 02:11:59 -0700
From: John Marvin [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Is there any Asterisk controllable
thermostat?
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi all.
Done some research, Googled a lot, but can't find out if there is a USB
FXO adapter that works well with Asterisk. If someone knows of one or
has used one, I'd be very interested to hear about it.
Many thanks,
Nathan
--
www.nathanpralle.com
Armin Schindler wrote:
I don't have a changelog.
If this problem appears again, please create a memory dump of the cards
memory (divactrl can do that). This will help to find the problem, but the
latest driver/firmware should be used.
Hi Armin!
Can you please tell me exactly the proper
Hello,
Does anyone know what is this traffic from Polycom IP300 to asterisk
server on RTP port range ?
17:49:35.355673 IP 192.168.2.215.2229 192.168.2.210.19615: UDP, length 72
17:49:42.372713 IP 192.168.2.214.2223 192.168.2.210.16487: UDP, length 72
17:49:44.353414 IP 192.168.2.216.2237
Hi to all,
I got a Cisco 2651XM wired to an E1 PRI.
What I want to do is to pass all incoming calls to my asterisk.
This is my actual conf:
http://pastebin.ca/270677
with this I'm able to call my number from outside, but the call stop on the
2600, infact I can hear the tone, but I'm not able to
On Wed, 6 Dec 2006 12:58:32 -0800, Brad Templeton wrote:
On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote:
Some companies offer PSTN failover on DIDs, which I think is a good
idea. Works at least if your equipment, or their middle equipment is
down but doesn't work if the PSTN failover
Is this for a pots line ?
- Original Message -
From: Ron McCarthy
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, December 06, 2006 9:10 PM
Subject: [asterisk-users] Setting outgoing caller id on a zap channel forone
sip extension only
Hi
I need a router for a reason. My client is in the middle east where they have
lots of fun with tacking on money ;). A crappy router wont do much.
- Original Message -
From: Tom Lynn
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, December 07, 2006
On Thu, 7 Dec 2006, Klaus Darilion wrote:
Armin Schindler wrote:
I don't have a changelog.
If this problem appears again, please create a memory dump of the cards
memory (divactrl can do that). This will help to find the problem, but
the latest driver/firmware should be used.
Hi
On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote:
Hi all.
Done some research, Googled a lot, but can't find out if there is a USB
FXO adapter that works well with Asterisk. If someone knows of one or
has used one, I'd be very interested to hear about it.
Take a look:
The Message Waiting Lamp (neon) on these phones requires a 90v signal
which is generated and switched to the phone via a special station card
on an analog PBX. This feature was developed mainly for Hotel and Motels
but I doubt there are any manufacturers who would develop this
functionality for
On Thu, 2006-12-07 at 17:51 +0100, FaberK wrote:
Hi to all,
I got a Cisco 2651XM wired to an E1 PRI.
What I want to do is to pass all incoming calls to my asterisk.
This is my actual conf:
http://pastebin.ca/270677
with this I'm able to call my number from outside, but the call stop
on the
voice service voip
sip
session transport tcp
Last I checked, asterisk doesn't support TCP SIP signaling (or RTP
over TCP). See what happens if you change it back to the UDP default.
On 12/7/06, FaberK [EMAIL PROTECTED] wrote:
Hi to all,
I got a Cisco 2651XM wired to an E1 PRI.
What I want to
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,.
WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,.
WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,.
WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native
It's the tos that probably saves the.
- Original Message -
From: Paul [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 07, 2006 7:47 AM
Subject: Re: [asterisk-users] any possibility of Vonage
Hi
In dial-peer voice 697617664 voip
your must specify into voip dial peer
session protocol sipv2
and check if session target sip-server is corect doing a ping to sip-server
.
I think you must configure it with ipv4:ip_addres or map a host entry with
ip host sip-server x.x.x.x in global
Have a look at TIMEOUT(digit)
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
On 12/7/06, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote:
Hi,
the german telco Colt Telekom has assigned the phone number block 56830-xxx to
one of our customers. In the
Does anyone know if Xorcom's Astribank can work within a Xen VM ?
Guillermo Salas M. wrote:
On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote:
Hi all.
Done some research, Googled a lot, but can't find out if there is a USB
FXO adapter that works well with Asterisk. If someone
I ran across this article today:
http://economictimes.indiatimes.com/articleshow/726843.cms
Anybody know what the implications are for asterisk servers in and out
of the country used by people in the country?
Tom
___
--Bandwidth and Colocation
On Thu, 2006-12-07 at 18:17 +, jose luis peche baldera wrote:
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar
algun
parche a esta version, tengo el siguiente error en la consola de
asterisk
cuando establesco llamada a traves del VICIDIAL,.
WARNING[21235]:
On 7 Dec 2006, at 17:19, Dovid B wrote:
I need a router for a reason. My client is in the middle east where
they have lots of fun with tacking on money ;). A crappy router
wont do much.
It isn't a router, but the linksys NSLU2 runs asterisk quite nicely
if you cut the
config back.
If
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and
they all are able to register and make calls with no problem . My voip
carrier supports gsm as well as ilbc .. Server takes calls from sip phones ,
does call recording in between and forwards to voip carrier . My problem is
You can run dnsmasq on the machine for local caching of the dns names.
(http://thekelleys.org.uk/dnsmasq/doc.html) and then apply this patch
that will allow dnsmasq to set a minimum time to live
(http://lists.thekelleys.org.uk/pipermail/dnsmasq-discuss/2005q2/000253.html).
dnsmasq can be then
Thursday, December 7, 2006, 7:31:42 PM, Tomer Horn wrote:
Does anyone know if Xorcom's Astribank can work within a Xen VM ?
Well, I think running asterisk in xen domain is very hardcore :)
--
Best regards,
Csibra Gergomailto:[EMAIL PROTECTED]
Jon:
I will second that motion ... This is something I would be very interested
in seeing as I have a similar requirement ...
Have a number of folks on my system who work from home ... A number of them
have Asterisk servers that register with the main office Asterisk server ...
Right now I am
HI All,
Something weird has happened to my (*) setup.
Setup:
I'm using a Realtime-Driven (*) server for voicemail which has the
knowledge of all mailbox users on the softswitch which is remote to
this (*) box. Since that's all this box is used for, all I have in the
sip.conf is the definition
Thursday, December 7, 2006, 7:40:09 PM, Guillermo Salas M. wrote:
Please, make a new message for a new question, do not reply a thread
with a different topic, and finally, use english.
Quoting a whole mail with headers and footers is bad too...
--
Best regards,
Csibra Gergo
I know this is not asterisk specific but we all know this group is used
for Polycom issues as well...
I have hints working ok on Asterisk. However the Polycom phone will
only show the buddies key if there is not a call. This defeats the
purpose of using the buddies to see if you can transfer
Figures I email this and realized I can hit
Menu
1 (Features)
4 (Presence)
2 (Buddy Status)
Wow that's a lot of key strokes. Anyway to reduce that to a one button
touch? I don't mind doing that but I guess I should think of the users
:-)
Bill
I ran across this article today:
http://economictimes.indiatimes.com/articleshow/726843.cms
Anybody know what the implications are for asterisk servers in and out
of the country used by people in the country?
Ummm
Anybody offering VPN IAX services yet?
--
Henry J. Cobb
http://pastebin.ca/270840
This is the newone with some changements.
Unfortunately, always the same problem.
Fran, if I add the dial-peer voice 10 pots, I receive the advise that the
number does not exist.
Also, I do not find the way to add authentication username
asterisk-uername password
going to depend on how the driver is provided. If its a binary driver,
very unlikely that it'll work. If you get the source and can compile it,
you can usually hack it into submission. I got my PCI FXO cards working
in xen this way.
Tomer Horn wrote:
Does anyone know if Xorcom's Astribank
In sip.conf set dtmfmode=INFO
On 12/7/06, CheungJenny [EMAIL PROTECTED] wrote:
Hi all,
I have a question: how to configure Asterisk to support SIP INFO method?
I encountered this problem when I find my UA don't send INFO message to
another UA, actually it should. Asterisk was used as a SIP
you would be surprised. I run it at home in a xen vm with one of those
cheapy FXO cards and have had great luck. CPU usage isn't too
outrageous and the zttesst gives me the same results in a VM as I get on
hardware. The only trouble I've had is that I need to wakeup the FXO
card if I do a
(2)
exten= 0072,4,set(MONITORFILENAME=${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Hello,
In short – Asterisk is not able to recognize that the 'other' person to whom
call was made has hung up – hence the channel stays busy.
http://kb.digium.com/entry/1/6/
I would try with busydetect and busycount.. Best regards,
--
Nicolás Gudiño
Buenos Aires - Argentina
Asterisk doesn't seem to be relaying 183, Session Progress SIP messages
received from an upstream host back to the phone.
Anyone know why? Here's the SIP message that Asterisk receives, and it does
nothing with it. It doesn't pass it back to the phone.
-- SIP read from xxx.yyy.142.234:5060:
you can try this patch,
0004825: [patch][post 1.4] New codec negotiation algorithm
http://bugs.digium.com/view.php?id=4825
I'm think, this is one of the most wanted feature,
but unfortunately will not be in asterisk 1.4 and we must wait for 1.6
to be officially supported feature :'(
PJ
what about to try mgcp to control gateway?
I haven't try this yet, but mgcp is standard signaling protocol
supported by asterisk for controling voip gateways,
advantage of mgcp is centralized configuration/dialplan/call processing
in asterisk.
PJ
FaberK wrote:
http://pastebin.ca/270840
This
}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
I am still on asterisk 1.2 branch svn ( afraid of word beta on server :( ) .
I will try out that patch.
On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
you can try this patch,
0004825: [patch][post 1.4] New codec negotiation algorithm
http://bugs.digium.com/view.php?id=4825
I'm think, this
Hi All,
I recently received my Cisco 7970 and have it up and running with
8.0.4 firmware, with asterisk 1.4. Seems to function pretty great so
far, aside from a few issues.
Here is what I have noticed so far, anybody have any fixes for these issues?
1. Contrary to the forums and lists I've
Hi,
Am Donnerstag, 7. Dezember 2006 19:31 schrieb Forrest Beck:
Have a look at TIMEOUT(digit)
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
I don't see how this function could help me.
If I change
exten = 5683091,1,Answer()
exten =
When I use AgentCallbackLogin() to logout an agent, it always ask for new
extension. I can press # to logout. But I'd like the remove this new extension
prompt so when agents are trying to logout, they do not have to press #.
Does anybody know how to do this?
I am using Asterisk 1.2.12.1
On 12/7/06, Henry J. Cobb [EMAIL PROTECTED] wrote:
Anybody offering VPN IAX services yet?
I'm not sure, but does this only apply to VoIP service providers?
What about self run asterisk servers?
Tom
___
--Bandwidth and Colocation provided by
JO == J Oquendo [EMAIL PROTECTED] writes:
JO I don't follow... Remove the mechanical lifter? Then do what, go
JO from the Plantronic to the headset jack on the Snom, leave the
JO receiver in its normal port? If I do this, the person has to hit
JO the headset button on the Snom...
JO
I have been using the Linksys BEFRS81 Version 2 8 port router for some
time now, using IAX to and from other Asterisk boxes, and before that
the 4 port version, but discovered that after 18 minutes or so, SIP
traffic ( Vonage or Stanaphone through Asterisk ) would hose the
router, and all
= 0072,4,set(MONITORFILENAME=
${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller
For what its worth,
We use plantronics headsets on snom360. Plantronics headset gets
connected to the headset jack, then you use the headset button to
activate it. The feedback is due to grounding issues. The phone MUST
be connected to a hub/switch using a STP cable. UTP cables will NOT
I know this has been added to SVN but I'm looking for the source for the
original module. It used to be at http://redice.krisk.org/ but this page
no longer seems to display anything. I'd like to add it to my 1.2.13
stable install. Does anyone have a copy of the original? I used to have
this
Just out of interest are they openwrt compatible?
Bails
John Novack wrote:
I have been using the Linksys BEFRS81 Version 2 8 port router for some
time now, using IAX to and from other Asterisk boxes, and before that
the 4 port version, but discovered that after 18 minutes or so, SIP
traffic
Doug, Everyone:
I'll make you an offer you (hopefully) can't refuse:
I've been fixing manager bugs here and there, and am willing to take on
any manager issues out there, for 1.4, and trunk, especially, so as to
have things nice and solid for 1.4 before it gets out of beta.
So, give me some
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
tacking pn = adding on - sorry for not being more specific.
I have seen that people in the past have used a linksys router to run
asterisk. It would be to expensive to bring in a PC for every location. So
we want to import cheap home routers put asterisk on them as use them as
the go in
Hi,
I have installed the latest version of asterisk(1.4.0-beta3), and built
app_rxfax/txfax. I'm using spandsp from here,
http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz
Everything builds ok. I had to manually apply the patch from the site so
configure would spot
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote:
Hi list,
Can anyone who has successfully ran asterisk on a home router please
give
me the modell number as well as how they did it ?
Thanks.
Dovid
Sure. I have 5 units out there on Linksys WRT54GS v1.1 through v4
units. The software is
Same thing occuring here, on gentoo as well :(
On 07/12/06, Chris Glover [EMAIL PROTECTED] wrote:
Hi,
I have installed the latest version of asterisk(1.4.0-beta3), and built
app_rxfax/txfax. I'm using spandsp from here,
http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
into some gotcha down the road where there is some missing file that
needs to be put who knows where.
Not listed as one that is.
JN
bails wrote:
Just out of interest are they openwrt compatible?
Bails
John Novack wrote:
I have been using the Linksys BEFRS81 Version 2 8 port router for
some time now, using IAX to and from other Asterisk boxes, and before
that the 4 port version, but
Carla Schroder wrote:
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
into some gotcha down the road where there is some missing file that
Hi Steve.
Thanks, but unfortunately, I can't be involved in that. We are running Asterisk
in a production environment and we're using 1.2, not 1.4. I don't have the
resources to work with 1.4. Last time I filed a bug against 1.2 I got told off.
Here's an example of that cruddy output.
On Thursday 07 December 2006 17:42, John Novack wrote:
Carla Schroder wrote:
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote:
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't
run into some gotcha down
I have a site running asterisk 1.2.8 with a hand full of polycoms and
grandstream 2Kxp's. When a call is missed and you look at the missed call logs
on either, its has the persons exten, not the incoming caller id. Any ideas?
\\\|///\\ ~ ~ // ( @ @
Gang,
I'm wondering if anyone has run into this problem and found a solution.
When I use the manager interface to generate a call, I don't get very
much information in my CDR records when the dial status is BUSY, FAILED,
NOANSWER, etc. I am putting the dialed number into the CDR Userfield in
All,
This basic question might have been asked thousands of timesbut anyways:
when can Asterisk send out an re-INVITE to the line/trunk side?
It seems that the canreinvite does NOT matter for calls toward the trunk.
E.g. When I put a phone on hold, the re-INVITE is sent from phone to the
Michael Collins wrote:
Has anyone found a workaround or a best practice that allows CDR records
to contain the dialed phone number for every Dial() or Originate that
Asterisk processes?
I got around this by generating a call to a Local channel which is
always (well...nearly always)
Buenas noches
Tengo montado el sistema de marcador predictivo ASTGUICLIENT en 2
Servidores, en uno de ellos que es un Server HP Proliant G3 350 3GB RAM con
Slackware 11 y Asterisk 1.2.12.1 , dicho server solo se encarga de la voz.
Luego otro server un poco mas modesto (HP ML110 3.2GB), que tiene
its a bit off topic for asterisk but not for a bunch of telcom guys :
does anyone have a word or other wordprocessor or spreadsheet template they use
(and are willing to share) to create labels for nordx IBDN punchdown designator
strip labels ?
(something with a box line you can cut on to the
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