Yes, setting a fullname=xxx in peer definition sets the CALLERID(name)
But if the peer sets it own callerid then will it override this value?
On Wed, May 26, 2010 at 10:52 PM, Danny Nicholas da...@debsinc.com wrote:
I might be wrong, but I think that adding fullname=xxx to the context will
Thanks. Got this working by using setvar=variable=value in the peer
definition
SIPCHANINFO(peername) is giving me the 'name' of the peer i.e.
'TestSIPUser' and not the 'username'.
On Wed, May 26, 2010 at 11:20 PM, Jared Smith jsm...@digium.com wrote:
On Wed, 2010-05-26 at 22:48 +0530,
Hi Motiejus!
I'll look for JACK's configure script and send it off-list, unless someone
else here wants it?
Now about my programs. Scenario: Start Asterisk. Then directly use CLI to
dial. And if possible use asterisk only to pick up calls.
problem: I didn't find an easy way to let the
I updated Asterisk to 1.6.2.7 and now the user introduction in the
meetme application is no longer working:
[May 27 09:26:51] WARNING[2407]: channel.c:4034 ast_request: No channel
type registered for 'DAHDI'
-- Created MeetMe conference 1023 for conference '800'
[May 27 09:26:51]
Should I log this as a bug since it doesn't work?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: 20 May 2010 16:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, May 26, 2010 at 09:52:52PM +0200, Vincent wrote:
On Wed, 26 May 2010 17:30:08 +0200, Vincent codecompl...@free.fr
wrote:
More information, as I investigate:
For those having the same issue, here's what I learned:
1. In /etc/modprobe.d/dahdi.blacklist.conf, blacklist the netjet
DON'T RUN dahdi_genconf, as it overwrites system.conf.
Yes. dahdi_genconf reads /etc/dahdi/genconf_parameters and writes
/etc/dahdi/system.conf and /etc/asterisk/dahdi_channels.conf. You can
set the country as
lc_country fr
in /etc/dahdi/genconf_parameters.
1. When I run dahdi_genconf:
Anyone using an N900 with asterisk yet?
Had mine for a while now and VoIP (voice) has been working really well,
but the new firmware update brings SIP Video calling too - so just given
it a go... The fly in the ointment is that I only have a Grandstream
GXV3000 to test it with ...
And it
On Thu, May 27, 2010 at 11:12:05AM +0800, Michael wrote:
Dear Supports,
I was attempting to install BRI Card(OpenVox B800P) with wcb4xxp in NT
mode .But I can not make it worked!
Could you please give me some hints? Thanks in advance!
What version of Asterisk?
BRI NT PtMP is not
On Wed, May 26, 2010 at 04:41:57PM +0100, salaheddine elharit wrote:
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
What's dialplan.conf ?
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers
On Thu, 27 May 2010 12:29:05 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
This is a bug of the netjet module. It should not try to handle those
devices. While they use the netjet chipset, they are not the ISDN BRI
devices drivven by it.
Thanks for the explanation. On this exact same
On Thu, 27 May 2010 11:41:09 +0200, Leonardo Pistone
l.pist...@sispac.it wrote:
Yes. dahdi_genconf reads /etc/dahdi/genconf_parameters and writes
/etc/dahdi/system.conf and /etc/asterisk/dahdi_channels.conf.
Thanks for the tip.
Do you have asterisk installed? You neet at least to mkdir
On Thu, May 27, 2010 at 03:03:21PM +0200, Vincent wrote:
On Thu, 27 May 2010 12:29:05 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
This is a bug of the netjet module. It should not try to handle those
devices. While they use the netjet chipset, they are not the ISDN BRI
devices
On Thu, 27 May 2010 16:12:57 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Thanks for the explanation. On this exact same hardware, I didn't have
this problem with Dahdi/Zaptel 1.4.
Older kernel did not have the netjet module?
Yup, that could be the reason. Anyway, problem solved :-)
On Thu, 27 May 2010 15:09:45 +0200, Vincent codecompl...@free.fr
wrote:
/usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting
Running dahdi_cfg: [ OK ]
it's harmless. but it's a symtom of building dahdi-tools without
libusb
https://issues.asterisk.org/view.php?id=17189
--
Currently running Asterisk 1.6.2.6 with Polycom 550 phones with the latest SIP
firmware. We use the forward functionality on the phones (primarily forward on
no answer), and it works very well with one caveat: call timeout.
When the phone redirects (usually after 2 rings), asterisk still rolls
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote:
From another thread, I blacklisted netjet and now things are working.
But I wonder what is going on here and where did netjet come from -- it
doesn't look like an dahdi module to me.
It comes from mISDN. It is a very badly misbehaving
On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote:
First I noted that dahdi_dummy is no longer present in
kmod-dahdi-linux-2.3.0.1-1.
Not exactly true.
myhost01 asterisk # lsmod | grep dahdi
dahdi_dummy 5812 0
dahdi_transcode 8968 1 wctc4xxp
Hi all,
i do have the following setup
Incoming call over DAHDI - to another machine using IAX2 - Agent at
this machine starts an attended transfer to an external number
This new initiated call does go over IAX2 - the machine the original
call came in - DAHDI out into the world.
Agent does
GoogleTalk connects ok to Asterisk 1.6.2.7 but how can you choose
voice menu options (press 1 for Bob, press 2 for Betty, ...) from the
GT client?
(There is no dial pad in the Windows GT client, but what you type in
the message box does show up on the console as an incoming Jabber
message.)
Is
On Thursday 27 May 2010 03:55:05 Lee Archer wrote:
On Wednesday 19 May 2010 16:44 Tilghman Lesher wrote:
On Wednesday 19 May 2010 02:28:02 Lee Archer wrote:
Hi, is it possible to add a context from the console using the
dialplan command?
Yes, just add an extension to it. The context
I assume this patch is for 1.6X since I find no code similar to this in
1.4.30?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Thursday, May 27, 2010 1:15 PM
To: Asterisk Users Mailing
Hello all-
My client has purchased these two OpenVox cards and I'm configuring a
system with Asterisk 1.6. In the past I have used bristuff and libpri
with older versions of Asterisk, but now I would like to upgrade to
Asterisk 1.6. Question, should I be using mISDN or libpri for these
Hi,
I have a test server with 2 NICs, each with it own IP address. Let`s say
192.168.1.2 and 192.168.1.3. I would like some phones to register by using
192.168.1.2 and some by using 192.168.1.3 as the address.
Since the default IP is 192.168.1.2, that is the only working address. Every
2 things to try - (1) set bindaddr in sip.conf to 0.0.0.0 instead of
192.168.1.2 - in theory this will let * use both cards (2) start second
instance of asterisk bound to 192.168.1.3 - probably the approach with the
better chance of success.
_
From:
I should have mentionned this is already done. I can see that is a SIP
response when trying 192.168.1.3, but the phones fails to register. I
suspect a NAT/firewall issue because packets are leaving for 192.168.1.3,
but coming back from 192.168.1.2.
Mike
From:
All:
Yesterday I discovered something interesting. I dialed 1800ANCESTRY
from the asterisk system I am testing and got the number doesn't exist
message. I then dialed the same number from our old system and it went
through.
I realized that the Y in ancestry made the number too long, and
Hi listers!
Just ran across a customer who wants to replace an Aastra Nexspan with an
Asterisk 1.6.X, wants also to connect it to a MOCS (Microsoft Office
Comunications Server) though that's not my real concern right now.
I got one of his phones (Aastra conexity i740) and though I have been able
I guess it's the !, sometimes it has a funny behaviour.
try changing (. instead of ! and an X less)
exten =
_91XX!,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
; long distance
to
exten =
_91X.,1,DIAL(SIP/${EXTEN:1...@ia.ntelos.netexten%3a1...@ia.ntelos.net)
;
Hi!
Question, should I be using mISDN or libpri for these cards when they
are in the same system, or does DAHDI now support both cards under
asterisk 1.6 reliably?
I cannot answer that question, but do stay away from mISDN if you can.
Philipp
--
On Thu, 27 May 2010, Eddie Mikell wrote:
exten = _911,1,DIAL(SIP/${ext...@ia.ntelos.net) ; emergency
Unrelated to your question, but 911 doesn't need an underscore.
--
Thanks in advance,
-
Steve Edwards
David Backeberg wrote:
On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote:
First I noted that dahdi_dummy is no longer present in
kmod-dahdi-linux-2.3.0.1-1.
Not exactly true.
myhost01 asterisk # lsmod | grep dahdi
dahdi_dummy 5812 0
On Thu, 27 May 2010, Mike wrote:
Hi,
I have a test server with 2 NICs, each with it own IP address. Let`s say
192.168.1.2 and 192.168.1.3. I would like some phones to register by using
192.168.1.2 and some by using 192.168.1.3 as the address.
Since the default IP is 192.168.1.2,
Hello,
From www.x100p.com, I bought one of those cheap FXO cards. I have a
couple of questions/issues about it:
1. I noticed that...
- after cold booting the host, I see successful Dahdi/wcfxo messages
in /var/log/messages
- then, if I run either /etc/init.d/dahdi restart, or
/etc/init.d/dahdi
On Thursday 27 May 2010 14:05:20 Danny Nicholas wrote:
I assume this patch is for 1.6X since I find no code similar to this in
1.4.30?
You assume incorrectly. You probably missed that this patch is against
pbx_config.c, not main/pbx.c.
--
Tilghman Lesher
Digium, Inc. | Senior Software
That was a simplified example. I actually have two links from different
ISPs, totally different networks. Those on provider A should talk to
provider`s A IP address and have their answers come back from provider's A
IP, and those on provider B should talk to my provider B NIC and get the
response
Hi Guys,
I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri
with the new version of Libpri v1.4.11. The installed one was v1.4.10.
System is running Asterisk 1.4.21.2.
I did the following after:
cd /usr/src/libpri/
make
make clean
make install
Install end with these
On 28/05/2010, Mike l...@virtutel.ca wrote:
That was a simplified example. I actually have two links from different
ISPs, totally different networks. Those on provider A should talk to
provider`s A IP address and have their answers come back from provider's A
IP, and those on provider B
On 5/27/2010 2:33 PM, Philipp von Klitzing wrote:
Hi!
Question, should I be using mISDN or libpri for these cards when they
are in the same system, or does DAHDI now support both cards under
asterisk 1.6 reliably?
I cannot answer that question, but do stay away from mISDN if
- bruce bruce bruceb...@gmail.com wrote:
What am I doing wrong that it's not update to 1.4.11?
Thanks, Bruce --
Did you restart your services to ensure the new library was picked up?
--Tim
--
_
-- Bandwidth and
I suspect the channel is not ceased correctly in Siemens PBX, since you get
dial tone from Siemens PBX the channel from Asterisk is rejected in your
Siemens PBX.
On Thu, May 27, 2010 at 6:15 AM, Daniel Bareiro daniel-lis...@gmx.netwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Fri,
Hi all,
Today at 12 Noon EDT (9AM PDT, 5PM UK, 6PM Western Europe) the VUC
welcomes Ward Mundy from http://NerdVittles.com who will introduce us
to Incredible PBX, an Asterisk-based, easy-to-deploy PBX. Rather than
start a long chain of features here,we invite you to join us live (see
below) or
hi, all
Is ther any way to set up call-waiting feature in asterisk using dialplan or
any other ways. I want to use only
asterisk for that not any other gui.
I am using asterisk 1.4.28.
Regards,
--
Bhrugu Mehta
Sr. S/W Engineer (DD)
VOIP,Telephony Team
India
--
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