Hi,
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
thx
rich
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Danny, Doug
thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().
rich
On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle supp...@drdos.info wrote:
Coco Richard wrote:
Hi,
i need to save into a local
Hi,
there are several possibilities do to it
REGISTER Username/Extensions Enumeration
INVITE Username/Extensions Enumeration
OPTION Username/Extensions Enumeration
for more information:
http://www.hackingvoip.com/presentations/sample_chapter3_hacking_voip.pdf
rich...
On Thu, Nov 19, 2009 at
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote:
On Monday 09 November 2009 15:38:54 Coco Richard wrote:
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK
I took a look in chan_sip.c an for 1.4.13 ALLOWED_METHODS doesn't add
INFO. So I will upgrade to 1.6...
thank you for the replies...
rich...
On Tue, Nov 10, 2009 at 9:21 AM, Coco Richard
richard.kingc...@gmail.com wrote:
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO Method. But in the RFC3261 (20.5)
you can read:
All methods, including ACK and CANCEL, understood by the UA MUST be
of that
method in order for the other UA to be willing to send messages with
that request method to it.
Coco Richard wrote:
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO
Hi all,
our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)
[from_sip_proxy]
exten = 36122512,1,Answer()
exten = 36122512,2,VoiceMailMain()
exten = 3612252,1,Answer()
exten = 3612252,2,MeetMe(313,MI)
exten = 3612252,3,HangUp()
Hello
I asked the same thing some time ago, but nobody answered.
I founded some workaround.
Use this in your dialplan:
exten = _7.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:1})
exten = _7.,n,Pickup(${EXTEN:[EMAIL PROTECTED])
This worked for me.
Cosmin
--- On Thu, 11/27/08, Bruno Castelo Branco
/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse
I want to put
believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse I want to
put the users in call queues and I want anybody from the company to be able to
pick a
Hi all,
How can i use different VLANs for signaling and audio, e.g vlan 100 for sip
and vlan 200 for rtp? Where can i find documentations for this?
Comments and suggestions are welcomed (a sample config too :-)))
thx in advance
rich
___
-- Bandwidth
Hello
I have a problem with my asterisk server, I want to disable the call on hold
function when flash hook is pressed.(actually to fully disable it for the users
connected to the box)
It does call on hold when I use the asterisk as a rtp proxy, when it does
nattive bridging, the box has
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the
--- J. Oquendo [EMAIL PROTECTED] wrote:
richard Coco wrote:
Hi all,
i have asterisk 1.2.17 with sip tcp support and i
am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the
HG3540.
But when i try to place a call from Asterisk
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:* 9722/asterisk
972 is the tie access code from Hiapth to Asterisk.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard
Coco wrote:
Hi all
sorry, it works with upd... I am now able to make and
to receive calls.
thx...
--- richard Coco [EMAIL PROTECTED] wrote:
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:*
9722/asterisk
972 is the tie access code from Hiapth
Hi all,
according to
http://sip.fontventa.com/content/view/15/44/ i have
compiled the mpeg4ip libries without problem. After
copying the app_mp4.c file into de Asterisk apps
directory and changing the Makefile like.
[...]
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@
can use the
variable ${CALLERID(number)} .
- Original Message -
From: richard Coco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 13, 2007 10:53 PM
Subject: Re: [asterisk-users] voicemail scenario
Hi all,
i need help to implement a voicemail scenario. What i
am trying to do is the following.
user X dials a direct access for user Y voicemail and
is asked to enter a number (e.g 12345678) and then
leaves a message. Then asterisk sends a notification
with attachement. The problem is that i
Hi,
i finally managed to get it work using GlobalVar.
I still have a question. I have several context in my
voicemail.conf like
[default]
[customer_1]
[customer_2]
[customer_3]
How can i set a different emailsubject for each
context?
thx
--- richard Coco [EMAIL PROTECTED] wrote:
Hi all
Hi all,
i have ser and asterisk on the same box with a public
ip address. When an UA behind NAT registred on SER try
to call the Voicemail or another UA registred on
Asterisk i have one way audio (caller cannot hear the
callee).
[UA/SER]--[router/nat]--[SER/Asterisk]
UA has private
How (and where) could you provision those phones ?
Do you have any support from Siemens or anyone ?
We have a HiPath4000 V1.0 interconnected with Asterisk
using oh323. I have flashed several OptiPoints (from
the HiPath) to SIP firmware. But again OptiPoints seem
to work well with Asterisk but
Hi,
http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm
rich.
--- Olivier [EMAIL PROTECTED] wrote:
Hi,
Is anyone aware of a wired sip hardphone supporting
802.1x authentication ?
I've been told some Avaya and Alcatel ip phones
supported 802.1x.
As 802.1x
FreeRadius)
a howto about 802.1X Port-Based Authentication are
avalaible at
http://tldp.org/HOWTO/html_single/8021X-HOWTO/
2007/1/2, richard Coco [EMAIL PROTECTED]:
***
This message was sent to your KasMail disposable
email address:
Asterisk Users Mailing List - Non-Commercial
Discussion
))
{
hmap.get(((HangupEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_green.png));
}
}
}
}
thx in advance!
--- Tim Panton [EMAIL PROTECTED] wrote:
On 4 Oct 2006, at 16:33, richard Coco wrote:
Hi all
Hi all,
first of all sorry for the question. I know there is
an asterisk-java mailinglist but i am not subscribed
to this list and i am sure there are asterisk-java
guru on this list who can help me.
I am trying to get the status of a peer using
SipShowPeerAction. Unfortunately the getStatus
|wav
attach=no
maxmessage=180
maxgreet=60
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emailbody=Dear ${VM_NAME}:\n\n\t you were just left a
${VM_DUR} long message (number ${VM_MSGNUM})\nin
mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on
${VM_DATE} \n
[default]
2001 = 2001,2001,Coco
Hi all,
we plan to install several IAX softphones.
http://www.voip-info.org/wiki-Asterisk+IAX+clients
lists a lot of IAX phones for Windows and Linux. Which
one would you recommand? We will install IAX client on
Linux and Windows.
thx richard
__
Hi,
i have xlite too and it works without any problems.
ps: what about ekiga? (www.ekiga.org)
rich
--- Joao Pereira [EMAIL PROTECTED] wrote:
Hello to all
can someone recommend me a nice SIP client with
video for windows??
I tried X-Lite 3.0 but it's a lousy piece of
software.
Hi all,
i have following setup
[]--[asterisk]--[oh323]--[HiPath]--[8000]
is my voicemail access
exten = ,1,Answer()
exten = ,2,VoiceMailMain()
8000 is an Optiset phone registered on the HiPath.
When 8000 calls i have no voice (depends on the
setting of FastStart). When
)
would be this parameter?
Best Regards
Josué
2006/6/26, richard Coco [EMAIL PROTECTED]:
Hi Josué
if the Siemens phone calls Asterisk, it didn't get
a
dial tone from Asterisk? Is it correct?
if yes, this is depending of Asterisk which didn't
in
my HiPath 4000, what I have seemed in the COT is
TR6T (1tr6 isdn tie link)
would be this parameter?
Best Regards
Josué
2006/6/26, richard Coco [EMAIL PROTECTED]:
Hi Josué
if the Siemens phone calls Asterisk, it didn't
get a
dial tone from
Hi,
which Hicom and which version is installed?
Hicom 300 or Hicom100?
rich
--- Lito Lampitoc [EMAIL PROTECTED] wrote:
Hello all,
I'm new to asterisk. Our company wants to setup an
asterisk server and will
eventually move to IP centric phones, but they don't
want to just throw away
hi all,
The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.
I'am not sure but i thing that the feature CallerID
Name was introduced in version 3 of the H.323
standard. More informations about the owerviews at
http://www.packetizer.com/voip/h323/.
-Concerning HiPathv3.0.
In version 3.0 the
Hi Josue...
i have taken a short look at the configuration you
sent to me off list.
First of all, try to change the protocol from ECMAV2
to ETSI or EDSS1 (set the segmentation to 1) and like
suggested by Silviu change the switchtype=EuroISDN
too. EcmaV2 is normaly used to interconnect Siemens
Hi Josué
if the Siemens phone calls Asterisk, it didn't get a
dial tone from Asterisk? Is it correct?
if yes, this is depending of Asterisk which didn't
generates a ringback messages as it expexts dial ton
generation localy. So try this workaround for HiPath
local dial ton generation:
- Add
Hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
Hi all,
i am playing around with several optipoint4x0 and run
into trouble trying to get hint functionality to work.
I notice that there is no status notifications. But
afaik this should be implemented via the
SUBSCRIBE/NOTIFY mechanism.
I can see INVITE, TRYING, RINGING, ACK, BYE but no
Hi,
first of all, sorry for this long thread... I have
changed my extensions.conf like you suggested and
delete the line with subscribecontext=notify. But
unfortunately i still don't see subscribe request in
the sip debug trace.
SIP Debugging enabled
kingcoco*CLI
-- SIP read from
configuration on
the IP-phone?
thx in advance
--- Avi Miller [EMAIL PROTECTED] wrote:
On 17/05/2006, at 8:27 PM, richard Coco wrote:
unfortunately i still don't see subscribe request
in
the sip debug trace.
Have you configured your phone to subscribe to the
extension? :)
cYa,
Avi
Hi,
i have change my sip.conf and my extensions.conf but
unfortunately nothing change. Should i not see the
hint priority in the CLI?
richard
--- Steve Davies [EMAIL PROTECTED] wrote:
On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:
I believe the hint priority must be in the same
context
Hi all,
i am desperating, trying to configure an OptiPoint410
with the hint priority.
Here what i have...
OptiPoint410std- exten 2001
X-Lite - exten 2002
But unfortunately no LED ON on my OptiPoint410
sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
Hi all,
i have an Asterisk box with an Eicon 4BRI with
chan_capi-cm and every thing works fine. We now plan
to install a new Asterisk using a Dialogic BRI/2VFD.
Is the Dialogic card supported and can i use
chan_capi-cm? Has anyone managed to install this card?
Unfortunately i was unable to find
Hi list,
i am playing around with asterisk manager interface
(and astriskjava) and i notice that the login is not
case sensitive.
so i can use
username: admin
secret: admin
---
# telnet localhost 5038
Trying 127.0.0.1...
Connected to
Hi,
if yo are looking a way to interconnect Asterisk with
a HiPath 4000 via IP, so you have 2 ways to do it.
- via oh323 (for HiPath 4000 version 1 and 2)
- since HiPath4000 version 3 you are able to
interconnect using sipQ (SIP Trunking)
--- Viktor Tatianin [EMAIL PROTECTED] wrote:
Hello
Hi again,
i don't think that the HiPath2000 is an Asterisk based
system. AFAIK the HiPath2K is only configurable using
a Web-based tool (no console access). For the moment
the HiPath2K will only be release with CornetIP (HFA).
No SIP (panned in a second step) and unfortunazely no
IAX are
Jan 2006, richard Coco wrote:
Hi Armin,
thx for your feedback, but what do you mean with
Did
you load the card with config for DID on that
port?
I have loaded the modules with:
modprobe capi
modprobe kernelcapi
modprobe divacapi
modprobe divas
and then loaded divactrl
that this is ok (it works without did)? Or
have i forgotten something?
thx in advance..
--- Armin Schindler [EMAIL PROTECTED] wrote:
On Mon, 16 Jan 2006, richard Coco wrote:
Hi all,
i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and
one
Hi all,
i have asterisk 1.0.9 with an Eicon Diva 4bri and
chan_capi-cm-0.6. I have 2 NTBAs (one with did and one
without).
When using the one without did, i am able to place
outgoing and incoming calls. When i use the NTBAs with
did i have a layer 2 error.
Anyone an idea?
-- Executing
Hi,
we have interconnected Asterisk with a HiPath4000 V1.0
using a H.323 Trunk. You have to install the oh323
channel from [1]. On your HiPath4000 V1.0 or V2.0 you
need a HG3550 board for IP-Trunking.
If you have the version 3.0 then the HiPath supports
SIP-Trunking but i have not tested it yet.
Hi all,
i run into problems using park calling with chan_capi.
My setup looks like this
[200X]--[Asterisk]--[PSTN]
For internal calls [1] and for incoming call from
PSTN[2] every thing works fine. Unfortunately when a
sip extension (say 2007) makes an outgoing call to
PSTN and 2007 tranfers to
Hi all,
i'm wondering if anyone has ever managed to get moh
working on Siemens OptiPoint400?
if yes, can you please explain how you did it...
thx.
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
Hi all,
I'm trying to configure a remote user with a DrayTek
2600Vgi. The setup looks like this.
[SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]
I can place calls to the DrayTek and recieve calls
from the analog phone. However, the calling party does
not hear the
called
Hi Alessio
[SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]
I tried a similar setup some times ago and it was
working, have you
put the private ip address of the asterisk box in
the vigor setup ?
Can you ping the private address of the vigor from
the asterisk box
Alessio, Sergio
So an upgrade is of course necessary.
i have upgraded the vigor. Bad news... i am not able
to register the draytek anymore. But using a XLite on
my pc behind the Vigor works now fine (no one way
audio).
however i have an other question. I saw you put for
the bindaddr same
Hi all,
i'm trying to install a EICON DIVA 4BRI (on CentOS 4.1
2.6.9-22.0.1.EL) using latest package from
sourceforge (chan_capi-cm-0.6.tar.gz).
I have installed divactrl_2.1.tar.gz and untared
protocols_all.tar.bz2 in /usr/share/eicon.
---
Hi Jacky,
thx for the feedback
rich.
--- Jacky [EMAIL PROTECTED] wrote:
Hi, Richard,
I still try, but fail with rtp transfer.
2005/9/27, richard Coco [EMAIL PROTECTED]:
I still find out how to let LCS 2005 accept SIP
invite from Asterisk,
Need more help.
Hi jacky
I still find out how to let LCS 2005 accept SIP
invite from Asterisk,
Need more help.
Hi jacky,
can you please share your experience and explain how
to let LCS accept SIP invite from Asterisk.
I deseperate trying to place a call from asterisk to
LCS. (calling from Asterisk to LCS using
to place a call from lcs to *.
thx in advance...
--- richard Coco [EMAIL PROTECTED] wrote:
Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where
Hi,
i have the same setup too.
[exten_3008]-[asterisk/TCP_SUPPORT]-[LCS]-[exten_20]
Unfortunately i don't know how to configure the
dialplan in my LCS. Can you please give me a hint
where to configure this.
thx.
--- Jacky [EMAIL PROTECTED] wrote:
LCS 2005 just support SIP TCP or
Hi,
The only thing i know is that you need a netbootserver
using five special files. So, if possible, ask Siemens
for the optipoint 600 netboot upgrade procedure. AFAIK
it is a known problem...
hope it helps...
--- Anthony Cox [EMAIL PROTECTED] wrote:
Not strictly a problem with Asterisk but
I already have OH323 support in Asterisk, but have
no clue how to
configure the HiPath.
hi...
oh323 is the only thing you need for Astersik. For the
HiPath it depends on which version you have.
FOR HiPath4000 V1.0
---
for version 1.0 you need a HG3550 V1.1 Board.
-Configure
Hi all,
i am looking for informations about large installation
with Asterisk (~3000 users). Has anybody experience
with such a setup. Any comments, suggestions or
problems would be appreciated.
thx in advance...
__
Do You Yahoo!?
Tired of spam?
--- [EMAIL PROTECTED] wrote:
I'm trying to setup Asterisk trunk to Siemens HiPath
4000 V2.01
i suppose you mean version 2.0 ;-)
What would be the best way to do so? I am a bit
confused because as far
as I've understand this PBX doesn't support H323,
but I saw somewhere
someone who
Hi Franz,
ok, can you please inform me (the list) if the Optipoint 420 with the firmware 4.0.22A work with Asterisk. If so i will try to contact our contact at Siemens and organize some Optipoint 420.
chan_cornet, which would support the proprietary Siemens protocol, is notpart of the CVS tree
Franz Knipp [EMAIL PROTECTED] wrote:
Hi,today I've got two Siemens optiPoint 420 phones and I want to connectthem to an existing Asterisk server.I didn't find any SIP firmware for that phone, according to announcementsit will be released later this year (hopefully soon).
The latest firmware for
Hi,
my setup
[pbx]---[oh323]--[asterisk]
calling from the pbx into the voicemail gives following outputin the console
-- Executing VoiceMailMain("OH323/R1909", "") in new stackApr 5 19:05:46 DEBUG[11862037]: res_adsi.c:212 __adsi_transmit_messages: No ADSI CPE detected (0) -- Playing 'vm-login'
Hi List,
Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens)an register itwith asterisk?
any suggestions?
thx in advance.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hi,
we use the oh323 driver (see the post from Joao for installationhttp://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg00931.html).
in the oh323.conf
[general]
listenPort= 192.x.x.x /the ip @ of the HG3550
fastStart=yes /*enable fast start
context=voip-h323
codec=G711A
in the
Hi,
i agree... no H.323 support for endpoint (HG3530) but you still have h.323 (for the moment only version 2.0)support for ip-trunking (HG3550).
So what if you have the following setup.
[OPTIPOINT400_HFA]--[HIPAT4K][oh.323][ASTERISK]--[OPTIPOINT400_SIP].
Am i right when i suppose that
GIBERT Frédéric [EMAIL PROTECTED] wrote:
STLS4 is a 4 BRI ports card to connect to carrier.
STMD8 is a card to connect 8 ISDN Siemens phones (optiset)
STMD8 is not a board for Optisets. You have to use a SLMO/SLU board to register an Optiset/Optipoint500.
Do you Yahoo!?
The all-new My Yahoo!
Hi,
The HG1500 is a HiPath3000 board and i don't have experience with Asterisk and HiPath3K.
What we have is an Asterisk connected to a Siemens HiPath4000 over a H.323 trunk using oh323 and the HG3550 board. It works fine. But the Siemens HG3550 only supports H.323 V2.0 (so not a lot of features
... but not yet a direct Asterisk-HiPath connection.
But doesnt Digium have Asterisk-HiPath solutions?
Joao
- Original Message -
From: richard Coco
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 05, 2005 12:13 PM
Subject: Re: [Asterisk-Users
Hello Steffen,
hey it sounds very good...!!! but what do you mean with *new hipath version doesn't support H.323 anymore*? What version are you talking about? As far as i know the new version of HiPath4000 V2.0 still supports H.323 (STMI2).Steffen Koepf [EMAIL PROTECTED] wrote:
Hello, I dont know
Hi all,
i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint?
Any help would be much appreciated!!
thx.
Do you Yahoo!?
Meet the all-new My
Hi Erik,
thx for the feedback. I turned off the Silence Suppression but unfortunately it doesn't work.
(i don't think that throwing it out of the window will resolve my problem... but it will think about it... to be continued;-))"E. Versaevel" [EMAIL PROTECTED] wrote:
If youre using G.711
hi
check the voicemail.conf
Attach=yes
Attach causes Asterisk to copy a voicemail message to an audio file and send it to the user as an attachment in an e-mail voicemail notification message. The default is not to do this. Attach takes two values yes or no.
extension_number =
richard Coco [EMAIL PROTECTED] wrote:
Peter Svensson [EMAIL PROTECTED] wrote:
On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one
Peter Svensson [EMAIL PROTECTED] wrote:
On Sun, 19 Dec 2004, Jens Kübler wrote: I've bought the Wildcard TE110 some days ago but I'm unable to get it to work with Siemens HiCom 300. I've tried this so far: 1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4 and 2 to
Hi all,
I have some OptiPoint400 standard SIP connected on Asterisk. They work pretty good. I only notice that calling from OptiPoint to OptiPoint doesn't show me the Caller ID name (only Caller ID number). But calling from an OptiPoint to a SoftClient (e.g X-Lite) shows me both on the
Hi all,
this is my actuel setup
[SIP 2005]--[asterisk]--H.323 Trunk--[PBX]--[ext. 8900]
Linux CentOS 3.3 (2.4.21-20.EL.c0)
asterisk-1.0.1
asterisk-oh323-0.6.3b
openh323_1.12.2
pwlib_1.5.2
Calling from SIPphone to the extension 8900 works always.
Calling from 8900 to SIPphone works only
Hi,
here what i have:
[2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004]
Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 onRedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL.
Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following
85 matches
Mail list logo
