Hi,
I'm using the polycom branch and have been trying to get the
outboundproxy=xxx to work. Is this something that should work in the version
of software?
Thanks,
Dean.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Hi,
When I make a call using asterisk (SIP - provider - PSTN), it is cutting out
all the background noise, making it a bit like a walkie talkie when they
stop talking. Is this something asterisk is doing?!?
I have a 2MB leased line just for calls, so bandwidth is fine, I have tried
with G711a/u a
Hi,
I'm trying to call an exten from inside extensions.ael, as below, ddi calls
ael and then ael needs to call the extensions.conf (8000 exten) for the call
queue.
Is this possible? Or is there an easier way to combine the exten 8000 to the
ael?
Thanks,
Dean.
ddi.conf
exten => _441234567890,
I have
the same problem here, why does asterisk not use ulaw with Sip1 -> Sip3
? As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto
ulaw when the g729 fails?
Thanks,
Dean.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf O
Hi,
I'm trying to get asterisk to use two different codecs, is this possible?
In the sip.conf I have:
[3002]
type=friend
host..
..
..
disallow=all
allow=alaw
allow=g729
When I make internal calls, ie voicemail or pstn calls, I would like it to
use alaw (which it does), but when I make an IAX ca
Hi,
I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc options on and off, turned down the SPA - PSTN gain
Hi,
I trying to get the softhangup option to work. I'm using the
Polycom_acd_functions branch of asterisk, so not sure if it works with this,
or I'm doing something wrong.
Below is what I have in the dial plan, using 444 and a mobile for testing,
as I would like to use this for emergency services.
] Asterisk Polycom_acd_functions and G729
On 7/31/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm trying to get G729 to work with the branch polycom_acd_functions, but
> having no luck. I have read through some of the G729 install pages, but
I'm
> new to l
Hi,
I'm trying to get G729 to work with the branch polycom_acd_functions, but
having no luck. I have read through some of the G729 install pages, but I'm
new to linux and asterisk, and not much made sense. Are there any idiot
proof instructions? I'm using Debian 3.1
I have tried the digium ones,
I'm trying to use the wav49 attachment, but it will not play on my machine.
I'm running windows xp with media player 10, it comes up with codec
'Microsoft GSM 6.10' not available. Microsoft stated that the GSM 6.10 is
included in media player 10.
Has anybody else had this problem? Could it be the
fic folder
HTH
Mat
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Dean @ INKnBITs
> Sent: 28 July 2006 14:40
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Change the from@ using th
Hi,
I'm trying to setup the voicemail.conf to email messages, but my mail server
fails because the from user is [EMAIL PROTECTED] Does anybody know away
to change the user part from root? I'm using exim4 to send the emails.
Thanks,
Dean.
___
--Bandwid
Hi,
I'm having trouble getting asterisk to send a voicemail message via email. I
can do a mail [EMAIL PROTECTED] from the linux command line and I receive the
email fine, and if I look in the exim4 logs it looks ok, has from user, to
user and completed but no email is received.
Any thoughts?
Tha
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom_acd_functions SIP trouble
Did you by chance have to make changes to get Zaptel to compile?
Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean @
INKnBITs
I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk
and for asterisk to use the SPA for outbound calls. This works fine, but is
there anyway to make the asterisk call the FXS port? So that I can call the
phone when needed and use the PSTN for calls if needed.
Thanks,
Dean.
I've got the same problem, the only version I can find that works is 30432,
but the meetme conference does not compile in this version. A fix for the
newest version for username/auth name would be great!
- Original Message -
From: "James Fromm" <[EMAIL PROTECTED]>
To:
Sent: Monday, July
Is it a Sipura 3000? If so you can use the link below, and if the ata is on
the network, you can enter the IP address and it will setup the ata for you,
and gives you the details to enter into asterisks. (If you use the bottom
option)
Only thing is you have to signup to use the wizard.
Worked gre
t;ManxPower" Wieling
Sent: 24 July 2006 13:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail not sent via email
Dean @ INKnBITs wrote:
> I have setup the voicemail.conf as below, but I not receiving any emails.
> Any thoughts?
>
>
I have setup the voicemail.conf as below, but I not receiving any emails.
Any thoughts?
voicemail.conf
[default]
3002 => 1234,Bob Wright,[EMAIL PROTECTED],,|attach=yes
I have also uncommitted the mailcmd=usr/sbin/sendmail -t
but that does not work.
Thanks,
Dean.
_
I trying to find a way of using two different callerid numbers. I have the
callerid="Agent 1" <33x> (in the sip.conf) being the direct dial
phone number, but for internal calls I would like it to show the extension
number. My internal dialplan is.
exten => 3002,1,Set(CALLERID(NUM)=xxx)
ex
I have setup 3 Linksys SPA-3000 devices to pass/send our analog voice calls
into/out of asterisk. The inbound calls work fine as I have set the
spa-3000's to forward all calls to an extension. I have added them to the
sip.conf as spa-3k1, spa-3k2, and spa-3k3. Is there a way for when some
picks up
= 1000
host= dynamic
nat = no
qualify = 1000
canreinvite = no
disallow= all
allow = ulaw
dtmfmode= rfc2833
agentlogin = yes
agentcbcontext = default
I also have an agent defined in the agnt.conf as:
agent => 2000,1234
In the phones.cfg find the below, you can
change the 8500 to your voicemail exten in extensions.conf of
asterisk
phones.cfg (for polycom)
The below will use the calling extension
number as the voicemail mailbox when called.
extensions.conf (asterisk)
exten =>
8500,1,VoicemailMai
lthough this will be a big leap forward if
it works and I would be willing to put up a bounty to move this forward.
Thanks,
Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean @
INKnBITs
Sent: Monday, July 17, 2006 3:38 AM
To: Asterisk Users Mailing L
Do you have a soft button on the IP301? I use the 501 and it works fine, you
do have to use the special asterisk code for it to work correctly. It lets
me login, logout, make the agent available/unavailable.
You can read about it at http://bugs.digium.com/view.php?id=6119
I found you must also us
Thanks, it was the parameter on the queue command, it was set for mins, not
seconds.
Regards,
Dean.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jon Farmer
Sent: 14 July 2006 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [
I'm trying to setup a call queue, but it keeps dropping calls that are
waiting for 1 min. Is there any way to make the queue unlimited amount of
time waiting? or is there a maximum?
Thanks,
Dean.
___
--Bandwidth and Colocation provided by Easynews.com -
I have an ACD asterisk system running, and if a call gets put through to an
agent and they hit the reject key (if they are busy), it puts the call to
their voicemail. I would like the call to stay in the queue and try another
agent. Is this possible?
Thanks for your help.
Dean.
__
Has anybody had this issue before?
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED]
Sent: 12 July 2006 11:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Possible polycom_acd_functions BUG
I have noticed a couple of issues, unless I'm
I have noticed a couple of issues, unless I'm doing something wrong?
I pulled with svn the
svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ which got
release 37416
This complies fine, in particular the meetme app.
If I setup a sip device in the sip.conf with a username and passwor
ycom 501 (I have tried it
with the eyebeam software and it does the same thing)
Thanks,
Dean.
-Original Message-
From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: 11 July 2006 23:10
To: Dean @ INKnBITs
Subject: Re: [asterisk-users] Polycom ACD, Asterisk, Kernel 2.6
On Tue, Jul 11, 2006
rcial Discussion ; Dean @ INKnBITs
Sent: Tuesday, July 11, 2006 10:09
PM
Subject: Re: [asterisk-users] Polycom
ACD, Asterisk, Kernel 2.6
What OS are you using? There is a known issue
with the kernel sources on CentOS 4.3 and I assume RHEL 4 that will keep
Zaptel from compiling?
I'm trying to build another asterisk server as I'm
having a problem with the current one. Unless anybody can tell me how to compile
the meetme app? Everything else works fine, asterisk just will not compile
meetme?!? (Under kernel 2.4)
I used svn to pull the trunk versions of libpri,
zapte
I have recompiled with mpg123 and music on hold is working fine. But the
asterisk will not compile the meetme app, using the release 30432. Is there
any way to compile the app manually?
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED]
Sent: 10 July 2006 13:25
To
t the proper zaptel exists now.
On 7/10/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:
> After using the trunk versions as below, it all compiled ok, and the
polycom
> acd is working great, but the music on hold and meetme will now work. I do
> not have any digium cards, is the ztdumm
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke
Sent: 04 July 2006 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk
polycom_acd_functionserror message
On 7/4/06, Dean @ INKnBITs <[EM
Has anybody managed to get the ACD function to work with Polycom IP501
phones and Asterisk?
I have used the trunk versions of libpri and zaptel and the
polycom_acd_functions -r30432 branch of asterisk, but it still will not
work.
Are there any special settings that need to be in the polycom confi
ssage-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke
Sent: 04 July 2006 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Libpri + Zaptel + Asterisk
polycom_acd_functionserror message
On 7/4/06, Dean @ INKnBITs <[EMAIL P
I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make
install), there was no errors.
I used svn to get the polycom_acd_functions asterisk branch release 30432, I
have to run make 3 times as it as it comes up with making opts re-run make.
It then completes and I run make insta
Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501
thats working with the agent login, I need to get this sorted to go live
next week. If anybody can share their experience or pointers.
Thanks,
Dean.
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED
Could anybody post a working sip.cfg and phone1.cfg for the polycom IP501
thats working with the agent login.
Thanks,
Dean.
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED]
Sent: 28 June 2006 17:25
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users
We have the Polycom IP501's here and for £140+VAT
there great value and have excellent call quality.
Regards,
Dean.
- Original Message -
From:
Daniel Salama
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, June 28, 2006 8:18
PM
Sub
-polycom
did make clean, make, make install, make samples
Edited the samples to get it to work.
Does that sound right?
Thanks again for you help,
Dean.
-Original Message-
From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED]
Sent: 28 June 2006 11:22
To: BJ Weschke
Subject: RE: [Asterisk-Users
Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501
Hi Dean -
It should be working. If not, please email me a sip debug trace along
with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf.
Thanks.
BJ
On 6/26/06, Dean @ INKnBITs <[EMAIL PROT
Hi,
Has anybody got the polycom acd function to work? I have the following
setup:
Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
zapt
Has anybody got the polycom acd function to work? I have the following
setup:
Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
zaptel, I
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