On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back.
It goes on hold just fine. But when I press the resume button, nothing
Use #exec directive to execute external script which retrieves registration
data from DB, and outputs correct registration string as text.
Do not forget to enable #exec in asterisk.conf
You will need to do sip reload each time you change registration settings.
With reload you will lose all
Anyone know how many users i can record in sip.conf. (NO..NO i am not
discussing the simultaneous sip calls).
I tried with 50k users in sip.conf, but the sip module didn't reload. tried
with few hundred of users and it works. any idea what is the limit in
sip.conf
regards
Mike
Hello,
I had an 1.4.21-2 Asterisk running on Debian/Etch with app_nv_faxdetect
running on it without any problem.
I upgraded the server to Debian/Lenny and Asterisk 1.4.27 and
app_nv_fax_detect is not working anymore: on an incoming call,
application is launched and never exit :-(
I
Hi out there,
I think i've everything set up properly, outbound calls are working fine, but
at incoming calls I can't hear anything, but the other one is able to hear me
perfectly.
I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to
my sip-provider using a trunk.
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back.
It goes on hold just fine. But when I press the resume
Kurian Thayil wrote:
I understand that new TE412P interface card comes with VPMOCT128
echo-cancellation module. I already have a server installed with
asterisk-1.2.17, zaptel-1.2.17.1 in production with TE412P card
installed already and having VPM450 echo cancellation module and it
works. I
John Novack wrote:
This fix has been around for several years, originally posted somewhere
by Max Park, and many users have applied it to systems used with antique
switching equipment.
I would hope it will find its way into the mainstream, and not just for
debian systems.
PLEASE
For the
Kevin P. Fleming wrote:
John Novack wrote:
This fix has been around for several years, originally posted somewhere
by Max Park, and many users have applied it to systems used with antique
switching equipment.
I would hope it will find its way into the mainstream, and not just for
It's now formerly a Mantis bug https://issues.asterisk.org/view.php?id=16339
John, can you please add to the bug.
I have just deployed this patch, so it needs a bit of time.
Alec Davis.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, 27 Nov 2009 06:50:15 -0800 (PST), bilal ghayyad wrote:
Hello All;
Anyone can advise for the good phone (Polycom, Linksys, ... etc) that is a
stable and support the codecs: g723, g729, and speex?
Actually I would like to have the speex codec because it have the ability to
compress to
I disagree with the Grandstreams. I have had pretty good luck with the
2000's and the GXP2010 especially. The Aastras i've had more of an issue
with, but i'm not completely against them. The fact that you have to use
I.E. on the Aastra web UI is kinda crappy, but you do what you have to do.
On Fri, 27 Nov 2009 12:47:52 -0500, Noah Miller wrote:
Hi Blaz -
Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40
USD?
I don't think there are any IAX hardphone in production anymore. You
might be able to find a used Atcom 320, but probably not for anywhere
close
I've got a single TDM 400P board with two internal ports and 1 external.
chan_dahdi.conf:
context=internal ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS
group=0
channel = 1 ; Telephone attached to port 1
channel
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on
Darrick Hartman wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant
to upgrade a system that doesn't currently work right.
There's no particular reason that you need to move to 1.6.x,
On Fri, Nov 27, 2009 at 11:17 PM, Michael Munger
mich...@highpoweredhelp.com wrote:
In 2007, I released a Polycom Provisioning Tool. I retired the package
earlier this year, and have had so many requests for it, I have revived the
concept, new, improved, and still FREE.
Any chance of you
During a call, I get the animated arrows. When I put a
call on hold, I get the flashing phone with the handset upside down. When
I
try to retreive the call, I get the animated arrow for a second,
This is normal and expected behavior so far, at least on my Polycom 500,
asterisk 1.4.27.
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant to upgrade a system that doesn't currently work right.
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with
Hello everybody,
I'm using Asterisk ( 1.6.1.9 ) Voicemail.
For example, if i call extension *11 which is not registered, from
extension *12, i have no greetings at all, i only have the please
leave a message after the beep.
I tried to record the busy, unavailable and temporary greetings for
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been
Do you have *11 registered in your voicemail.conf file? What does the
cli output look like when you try to leave a voicemail?
Thanks,
--Warren Selby
On Nov 28, 2009, at 7:22 PM, matthieu Nicaise techni...@thinkrosystem.com
wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9 )
Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial(SIP/*15-0849a370, SIP/*11,60)
in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Nov 29 03:38:13]
On Sat, Nov 28, 2009 at 8:39 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial(SIP/*15-0849a370, SIP/*11,60) in
new stack
The content of the voicemail directory is :
ls -lh /var/spool/asterisk/voicemail/default/*11/
total 324K
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:49 INBOX/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Old/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Urgent/
-rw-r--r-- 1 root root 3.5K
On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
Hello everybody,
I'm using Asterisk ( 1.6.1.9 ) Voicemail.
For example, if i call extension *11 which is not registered, from extension
*12, i have no greetings at all, i only have the please leave a message
On Sat, Nov 28, 2009 at 7:34 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
I made an error in my first mail, i'm calling voicemail in extensions.conf
this way :
exten = _*.,1,Dial(SIP/${EXTEN:0},60)
exten = _*.,n,VoiceMail(${EXTEN:0},u)
exten = _*.,n,Playback(ss-noservice)
You
Thank you Jonathan and Warren,
I now have the answer i needed !
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
Le 29 nov. 09 à
On Sat, Nov 28, 2009 at 9:34 PM, matthieu Nicaise
techni...@thinkrosystem.com wrote:
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.WAV
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.gsm
-rw-r--r-- 1 root root 40K 2009-11-28 23:47 unavail.wav
I made an error in my first mail,
sean darcy wrote:
I've got a single TDM 400P board with two internal ports and 1 external.
chan_dahdi.conf:
context=internal ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS
group=0
channel = 1 ; Telephone
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