2010/1/12 Kevin P. Fleming kpflem...@digium.com
...
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
In general, dial-string feature codes like this are not used on
'intelligent
This weekend is CampKDE in San Diego, California, USA. For those who
would like to participate, but can't physically attend, the BerkeleyTIP
group will (unofficially, as a public service) provide assistance to
everyone who would like to communicate via voice about /or with the
conference, other
Hi,all
What about the performance visit MYSQL in DialPlan code? if use MySQL
RealTime connection
--
Best regards,
Sucan
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asterisk-users mailing list
Hi,
How can I strip + from the front of the caller ID?
I have tried this:
exten = s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1})
But it is not working.
Szasz Szabolcs
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Hi,
running Asterisk 1.6.2.0 and have started to see in messages:
[Jan 14 05:43:43] NOTICE[29231] chan_sip.c: Peer '100' is now Lagged. (4007ms /
3000ms)
[Jan 14 05:43:53] NOTICE[29231] chan_sip.c: Peer '100' is now Reachable. (9ms /
3000ms)
[Jan 14 05:44:02] NOTICE[29231] chan_sip.c: Peer
On 14 Jan 2010, at 10:08, --[ UxBoD ]-- wrote:
running Asterisk 1.6.2.0 and have started to see in messages:
[Jan 14 05:43:43] NOTICE[29231] chan_sip.c: Peer '100' is now
Lagged. (4007ms / 3000ms)
[Jan 14 07:20:37] NOTICE[29231] chan_sip.c: Peer '100' is now
Lagged. (4007ms / 3000ms)
Hi
We run a hosted VoIP service for multiple customers off the same server
and I'm having an odd issue with just one customer in particular. We're
using realtime in a MySQL DB and this is their dialplan
*** 1. row ***
context: pcsu-Identifier
Ishfaq Malik wrote:
Hi
We run a hosted VoIP service for multiple customers off the same server
and I'm having an odd issue with just one customer in particular. We're
using realtime in a MySQL DB and this is their dialplan
*** 1. row ***
Szasz Szabolcs wrote:
How can I strip + from the front of the caller ID?
I have tried this:
exten = s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1})
But it is not working.
I thought that worked, but I haven't tested it in quite some time.
How about setting the value to a temporary
hi,all
while you can set ring groups in the queue.conf file. what's diff between them.
is that all members in a queue are a group? if that true.
it should not need to define group at all.
--
Best regards,
Sucan
--
_
--
I saw something like this in another answer, but here's an example that
should work (would on 1.4)
exten = s/_+X.,1,Set(TMPNAME=${CALLERID(name)})
exten = s/_+X.,n,Set(CLEANNAME=CUT(TMPNAME|\+|2))
exten = s/_+X.,n,Set(CALLERID(name)=${CLEANNAME})
in my installations ${X:1} is a hit or miss
In followme , is it be possible to have a third option
Whereas,
takecall=1
declinecall=2
proposed option
transfercall=3 or, transferring the call directly from followme
isn't really neccessary, if the callee could answer the call, and transfer
it someplace, that would work as
You could patch app_followme.c to add a transfercall option; if the caller
has to answer and transfer themselves, sort of defeats the need for the
option (IMO).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
over PSTN. I've tried turning hardware echo cancellation off but it
makes no difference. This is what I get in /var/spool/hylafax/log:
[r...@faxhost
Here is a scenario...
Sales guy is using follow-me.. received a call that should have been
directed to support... would it be possible to transfer the call to the
support number after the caller has explained that he doesn't really want to
buy anything..just talk to a support engineer...
Kingsley Tart wrote:
try several fax machines and see if you get the same results
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
over PSTN. I've tried turning hardware echo cancellation off but
Assuming all other options are correct, followme shouldn't stop him from
transferring the call (if the followme call is still being handled via
Asterisk). Let's say that you have 4 extensions for simplicity's sake;
extension 100 is the operator, extension 101 is support, extension 102 is
sales
Kingsley Tart wrote:
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
Without seeing your config files for iaxmodem and hylafax and also
seeing a dialplan snippet on how you're launching calls
On Thu, 14 Jan 2010, Doug Lytle wrote:
Kingsley Tart wrote:
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
Without seeing your config files for iaxmodem and hylafax and also
seeing a
Hello,
My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port
modular card and a single FXS module.
Got the Rhino card installed and the machine sees it:
r...@pbx:/etc/dahdi# dmesg | grep rcbfx
[ 71.985309] rcbfx :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ
21
[
I'm on 1.4.26.2 and have to have DAHDI entries in user.conf for Asterisk to
see the DAHDI line (dahdi_genconf users).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, January 14,
Hello,
What are the current methods for playing digits on different languages? I
presume the big ones like German have been dealt with, saying 2 and 20 to
announce 22. How is this currently decided? What about languages that say 20
and 2?
Is there a way of configuring via config files or
On Thu, 14 Jan 2010, Danny Nicholas wrote:
I'm on 1.4.26.2 and have to have DAHDI entries in user.conf for Asterisk to
see the DAHDI line (dahdi_genconf users).
Hmm, that would seem to coincide with the entries I already have in
chan_dahdi.conf. I did it anyway, but unfornuately there is no
AIR, French is set up to handle this, but I could very well be wrong;
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Örn Arnarson
Sent: Thursday, January 14, 2010 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello,
I found the respective functions in main/say.c -- now to find out if any
language uses the same rules as Icelandic :-)
Regards,
Örn
On Thu, Jan 14, 2010 at 3:40 PM, Danny Nicholas da...@debsinc.com wrote:
AIR, French is set up to handle this, but I could very well be wrong;
On Thu, 2010-01-14 at 15:25 +, Jeff LaCoursiere wrote:
Actually it is fairly clear that his dialplan is correctly routing the
calls to iaxmodem, and that iaxmodem is simply not completing the
training. I would say that the fax machine you are testing with is either
on a horribly noisy
Hello list,
I have the following in my voicemail.conf :
[general]
format=wav|wav49
When I receive a WAV-file from my Asterisk-server, I am unable to play
the file... There is no player on my Fedora that wants to play the file.
When I make my Asterisk-server unreachable, the incoming call goes
You don't have SOX on your Fedora box?. There is a difference between the
wav and WAV (wav49) format.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Thursday, January 14, 2010 10:04 AM
To: Asterisk
exten = did,1,Answer
exten = did,n,Playtones(ring)
exten = did,n,Wait(10)
exten = did,n,StopPlaytones()
exten = did,n,BackGround(sound file)
did = the DID number as presented and note the '1' before Answer.
This works for me.
exten = 820055,1,Answer()
exten = 820055,n,PlayTones(ring)
exten =
Perhaps this more belongs on the FreePBX list, but for the archives, this
is what I did to make it work:
chan_dahdi wants to read /etc/asterisk/chan_dahdi.conf
FreePBX, at least how I installed from source, seems to think I am still
running Zaptel. It created zapata_additional.conf when I
On Thu, Jan 14, 2010 at 03:22:41PM +, Jeff LaCoursiere wrote:
I am running FreePBX, so it created /etc/asterisk/zapata_additonal.conf,
so I linked /etc/asterisk/chan_dahdi.conf to it:
r...@pbx:/etc/asterisk# ls -ltr {chan_dahdi,zapata_additional}.conf
-rw-rw-r-- 1 asterisk asterisk 678
ESET NOD32 Antivirus, versión de la base de firmas de
virus 4771 (20100114) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
__ Información de ESET NOD32 Antivirus, versión de la base de firmas de
virus 4771 (20100114) __
ESET NOD32 Antivirus
- Jeff LaCoursiere j...@jeff.net wrote:
Perhaps this more belongs on the FreePBX list, but for the archives,
this
is what I did to make it work:
chan_dahdi wants to read /etc/asterisk/chan_dahdi.conf
FreePBX, at least how I installed from source, seems to think I am
still
running
Could you use NVFaxDetect?
On Thu, 2010-01-14 at 17:35 +, --[ UxBoD ]-- wrote:
Hi,
We have a issue where one of our clients is receiving a high volume of calls
from automated fax machines and passing through their context which means all
phones get rung.
I am looking for a way to
Hi,
We have a issue where one of our clients is receiving a high volume of calls
from automated fax machines and passing through their context which means all
phones get rung.
I am looking for a way to detect the fax tone, on answer, and route it to a
extension/macro where a announcement
Yeah sounds like you wanna use NVFaxDetect
it would allow you to add something like exten = fax,1,Swift(number
has changed); to your inbound call part of your dialplan
On Jan 14, 2010, at 11:45 AM, Juan C. Villa wrote:
Could you use NVFaxDetect?
On Thu, 2010-01-14 at 17:35 +, --[
Has anyone managed to get these two phones to make a video call to each other ?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
--
Hi,
Our guest this Friday is Himanshu Dwivendi, author of the book Hacking
VoIP. You're welcome to come discuss it with us on the conference.
Find your local time by going to http://vuc.me/next - the conference
begins a little before 12 Noon Eastern Time.
VUC has an IRC channel #vuc on
On Thu, 14 Jan 2010, Julian Lyndon-Smith wrote:
Has anyone managed to get these two phones to make a video call to each other
?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
I've done
Julian Lyndon-Smith wrote:
Has anyone managed to get these two phones to make a video call to each other
?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
Yes. Have done it often.
Are you actually trying to strip off the + or are you doing it as part of
trying to check the callerid number to see if it is valid. If the later,
then consider REGEX()... here is a snippet from my privacy manager script...
; First lookup number in asterisk DB for a Caller ID name. exten =
Urgh. That means my problem is probably beyond my control. The xlite
shows the video from the gxv, but as soon as I hit the send video
button xlite segfaults. I was hoping that there was a magical don't
crap out on me setting in xlite that someone as found.
Nuts.
Thanks anyway.
Julian
hi all,
just subscribed to the list and first mail, nice to be here.
Hopefully i'm in the right place for this question since i'm planning a
little VOIP implementation at the moment and ran in to something while
going through the shopping list.
i noticed that a lot of VOIP phones have a
Dear Team,
I have two questions for Asterisk feature.
1. Can Asterisk
support presence feature ?
I found following information. but it is too
old...?
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