could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
I would recommend using a normal doorphone, and connecting it to a SIP gateway
like the PAP2T.
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten = s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0...@sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: VC
Hi,
I've been used a Alphatech doorphone (SIP) with asterisk and works fine.
Cumps
Com os melhores cumprimentos,
Best regards,
CÉSAR SEQUEIRA
IT Expert
M: +351 961 355 772
@: cesar-seque...@justbit.pt
skype: cesar.sequeira.justbit
msn: cesar-seque...@justbit.pt
-Mensagem
On Tue, Mar 08, 2011 at 02:58:02PM -0600, Tilghman Lesher wrote:
On Tuesday 08 March 2011 06:49:55 Faisal Hanif wrote:
You can also set it in dialplan using Set(LANGUAGE=FR)
Actually, the right way to do this is:
Set(CHANNEL(language)=fr)
The LANGUAGE pseudo-variable is read-only.
Also
1.8 supports static peers along with realtime peers. I have tested.
On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Thanks Faisal, in fact I made a test that confirmed that in realtime
asterisk doesn’t supported static peers, like you told me.
Do you know
Hi,
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks,
-- Raj
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hi,
With Asterisk 1.8 is it now possible to register the same SIP account at
multiple endpoints and for both to ring when the associated extension is dialed
?
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by
I highly recommend Yealink phones.
On Wed, Mar 9, 2011 at 7:01 PM, Raj Mathur r...@linux-delhi.org wrote:
Hi,
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks,
-- Raj
--
Grandstream GXV3140 Multimedia IP Phones. Fully SIP capable. work well
with Asterisk.
On Wed, 2011-03-09 at 16:31 +0530, Raj Mathur wrote:
Hi,
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks,
-- Raj
--
My personal experience :
Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom
IP330, IP650.
DECT : Siemens C470, Polycom Kirk KWS300 and 600v3
Work well
AB2L
+33 (0)367100783
sebast...@ab2l.eu
Le 09/03/2011 13:09, John Kosmas a écrit :
Grandstream GXV3140 Multimedia IP
Arstan,
Very Nice phone indeed.
Yealink and the GXV3140's Grandstream's are quite nice!
On Wed, 2011-03-09 at 20:04 +0800, Arstan Jusupov wrote:
Re: [asterisk-users] [Opinion Request] SIP phones that work well with
Asterisk
--
Siemens IP A580 works fairly well.
2011/3/9 Sébastien BERGER sebast...@ab2l.eu
My personal experience :
Corded : Snom 3xx and 8xx, Aastra 6731i, 6755i and 6757i and Polycom IP330,
IP650.
DECT : Siemens C470, Polycom Kirk KWS300 and 600v3
Work well
AB2L
+33 (0)367100783
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice
Hey guys,
Currently we have non HWEC sangoma pri card but now we are planing to
replace card with HWEC support card for echo cancellation. So in this
case do I need to re-install everything? Like zaptel, asterisk etc..
Or just replace the card?
--
Sent from my iPhone
--
Phones what I'm using:
D-Link: 150s, 300S
Nokia E60, E65, E71, E52
Cisco: 7940, 7960
All above works perfect :)
On Wed, Mar 9, 2011 at 2:33 PM, asterisk asterisk aster...@ck-lee.com wrote:
Siemens IP A580 works fairly well.
2011/3/9 Sébastien BERGER sebast...@ab2l.eu
My personal
From: --[ UxBoD ]-- ux...@splatnix.net
Sent: Wednesday, March 09, 2011 6:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Multiple SIP endpoint registrations
Hi,
With Asterisk
You can register multiple end users with only one sip account but asterisk
does not support ringing all the registered phones on single account.
Whenever a new registration comes, asterisk updates its contact info in
memory. So if the registration is coming from multiple end users (multiple
ip
On Tue, 08 Mar 2011 07:47:39 EST, ken...@gnat.com (Richard Kenner)
wrote:
Maybe something like:
exten = s,n,SayDigits(${NBR2CALL:0:1})
exten = s,n,SayNumber(${NBR2CALL:2:2})
exten = s,n,SayNumber(${NBR2CALL:4:2})
exten = s,n,SayNumber(${NBR2CALL:6:2})
exten = s,n,SayNumber(${NBR2CALL:8:2})
Or
Only by replacing it.should not be a problem.
Juan.
Linux User #441131
On Wed, Mar 9, 2011 at 8:13 AM, Satish Patel satish...@hotmail.com wrote:
Hey guys,
Currently we have non HWEC sangoma pri card but now we are planing to
replace card with HWEC support card for echo cancellation. So
On Wed, 9 Mar 2011 12:43:37 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
On Tue, Mar 08, 2011 at 02:58:02PM -0600, Tilghman Lesher wrote:
On Tuesday 08 March 2011 06:49:55 Faisal Hanif wrote:
You can also set it in dialplan using Set(LANGUAGE=FR)
Actually, the right way to do this
On Fri, 4 Mar 2011, Steve Edwards wrote:
I'm starting a new project similar to a previous project where I used
OpenSER to front a bunch of Asterisk servers.
Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely
candidates.
I'm leaning towards OpenSIPS because it's in EPEL so I
...or for DAHDI channnels - the same thing in chan_dahdi.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bakko
Sent: 07 March 2011 19:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
*From*: Jonas Kellens jonas.kell...@telenet.be
*Sent*: Wednesday, March 09, 2011 4:18 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
We use Polycom phones. Mainly the IP321. We chose them because they can be
easily provisioned using an FTP server which allows us to configure settings
without visiting the phone,
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, March 09, 2011 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: OpenSIPS vs
I use Yealink, Linksys and Grandstream phones.
Com os melhores cumprimentos,
Best regards,
CÉSAR SEQUEIRA
IT Expert
M: +351 961 355 772
@: cesar-seque...@justbit.pt
skype: cesar.sequeira.justbit
msn: cesar-seque...@justbit.pt
-Mensagem original-
De:
On 03/09/2011 02:57 AM, Dan Journo wrote:
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
I would recommend using a normal doorphone, and connecting
Hi,
Latest SIP firmware for Polycom 601 is 3.1.6.
With this, is Directed Pickup supported ?
At the moment, when an extension is ringing, I can see BLF turning to solid
Red but I can't see it turning to Blinking Red.
Regards
--
You can Try:
Helios from 2N
http://www.2n.cz/en/products/communicators/doors/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
2011/2/17 Mike l...@net-wall.com
Hi,
Is there ANY way for me to see the status of the Polycom DND buttons in the
Asterisk hints? I`m using the BLF buttons to see the status of other
people`s lines, and DND should logically be somehow reflected (I don`t care
as much about Polycom showing
Sorry, just realised I posted this to the wrong mailing list.
Dan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
2011/3/9 Russell Bryant russ...@digium.com
- Original Message -
I tried to work around this by centralizing DND requests in Asterisk
and sending back a short (You're in DND mode) text to Polycom's
screen (using sipsak for this).
This was rather disappointing as Poycoms redirect
- Original Message -
Hi,
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
thank you,
Csaba
--
On Wed, Mar 9, 2011 at 12:53 PM, Darrick Hartman (lists)
dhart...@djhsolutions.com wrote:
On 03/09/2011 02:57 AM, Dan Journo wrote:
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in
On 3/9/11 6:35 AM, Tóth Csaba wrote:
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
I don't have direct Asterisk exerience, but when I tested
Hello!
Client is using ulaw, however server sometimes fills the log with following:
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to transmit
frame type slin, while native formats is 0x8 (alaw) read/write = 0x4
(ulaw)/0x8 (alaw)
[2011-03-09 21:23:07] WARNING[27204] chan_sip.c: Asked to
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, March 09, 2011 12:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Do Not Disturb button and asterisk hints
Is there a way to configure asterisk 1.8 and ALSA so I dont read
anything in for the input port.
I tried this in asound.conf
pcm.nullpcm {
type null
{
then in the alsa.conf file
input_device=plug:nullpcm
This did not seem to work as I still get feedback.
Is there a way to do this?
Thanks,
Hi,
We have a site where we'd like to move from mISDN/chan_lcr to DAHDI with
a b410p card.
We've tried everything we can think of to get it working but we never
seem to receive any calls etc - even though the card has no alarms.
We've tried replacing the card, changing the jumpers etc but
On 11-03-09 12:16 PM, Jerry Geis wrote:
Does asterisk 1.8 still need dahdi installed if you only doing SIP and
ALSA/console.
Only if you plan to use MeetMe().
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at:
I am having trouble with no return audio on inbound calls. I have been
working on this for 18 hours and even built a fresh server and moved
everything over and I am getting the same results. I need someone that can
help get this resolved tonight. I can provide access to all machines
involved.
So that suggests audio is flowing as follows in a unidirectional manner
199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732
209.216.2.203.60362
Somewhere near the end this pops up which is slightly different, I am guessing
74.204.4.5 is your asterisk box
19:18:36.389548 IP
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers.
BTW Did you try config_1 option. Please send us your configuration and we
will help you configure it properly. Either you can post them here or you
can send them directly to contact-supp...@didforsale.com
Jai
You can use this link too.
http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale
Keep the context as
context=from-trunk.
-Jai
On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote:
209.216.2.203 is sip signaling server and 199.173.66.22 is media
Hello All,
Some new security stuff is going on I suppose in 1.8 that I am not familiar
with and would appreciate your help
In a scenario such as the following:
Internet -- SBC -- Asterisk
upon trying to register an endpoint, the following is being observed on the
Asterisk Console. Have Googled
On Wednesday 09 Mar 2011, Raj Mathur wrote:
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks to all who replied.
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org/
GPG: 78D4 FC67
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits). ChanSpy is working fine for listening in to conversations
initiated by these channels, and I can use '*' to randomly switch
channels. However, is there any way in this scenario to be able to
switch ChanSpy
I think in the chanspy application you can give it a template to prepend to
what is entered. If you do chanspy(ab_) you might be able to enter the
remaining digits.
Short of that you can set up a loop that reads the digits, calls
chanspy(ab_${digits}), if the version you are using has my S
It just have ACL concept. You can add permitted IPs List to any peer then
only from that IPs user can register. If you want to permit all you can add
0.0.0.0 to ACL
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR
Sent: Thursday,
On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote:
It just have ACL concept. You can add permitted IPs List to any peer then
only from that IPs user can register. If you want to permit all you can add
0.0.0.0 to ACL
Thanks. but could you be a little more specific? I have
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use that peer's account from any IP
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR
Sent: Thursday, March 10, 2011 11:17 AM
To:
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote:
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use that peer’s account from any IP
Thanks. But Like I said, that's all done. Here's the Endpoint config:
Hi,
I'm discovering Polycom phones and specifically SPIP 3.1.6 powered ones.
Default display is showing :
- a blank line at the top of the screen
- then the date (2nd line)
- then the time (3rd line)
Is there a way to display something on the first line (the one above the
date line) (?
I saw
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