Satish,
As if I know, PRI provider give you PRI number at the time of purchase and
even billing documents will be made on the basis of the number only. So how
you can set another Caller-id number for that allotted number.
But you can do only change the PRI number for outside world after
Hi Virendra,
I should have said, you can *set the callerid to one of the numbers
allocated by them* for PRI, * and not to any other number*.
Enjoy.
--Satish Barot
On Tue, Feb 14, 2012 at 1:31 PM, virendra bhati virbh...@gmail.com wrote:
Satish,
As if I know, PRI provider give you PRI number
Hi,
For an RFP, I need to implement screen popup where caller names are
searched in outlook folders.
I would both consider free or paid solutions.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Hi!
I am setting up a little call center, but don't know how the agents system
works, can you guys please give me a little help?
I need to know how asterisk will know when I log agent X, and asterisk know
that agent is in the IP Z with the extension Y.
Thanks a lot.
Hugs,
ARPE
--
Hi,
I am trying to run asterisk on an AVM Fritz!Box Fon 7270 embedded DSL
router. This works quite well after getting rid of the preinstalled
phone server but I am encountering some unexpected behaviour.
Background: I am using two CAPI controllers provided by the hardware
- one in MSN mode
We update from packages.
Will this make its way to packages.asterisk.org or packages.digium.com? I
double checked the sites.
Thanks!
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Thanks David. I will check it out.
-Original message-
From: Klaverstyn, David C david.klavers...@intergraph.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Mon, Feb 13, 2012 04:34:30 GMT+00:00
Subject: Re: [asterisk-users] Polycom
Try TAPIRex
http://www.tapirex.com/en/
It's not free, but I've been using it with Asterisk + Outlook 2010
successfully. Users can also click on the screenpop and it will open up the
contact in Outlook. Pretty handy. You will need to make dialplan
modifications to send out the call info to the
Hi,
For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to make
Asterisk skip authentication even if a secret is defined in sip.conf for the
peer; i.e. similar to insecure=invite for INVITE requests?
If I leave secret blank, Asterisk doesn't require any authentication - this
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to
On 02/14/2012 08:43 AM, Matt Hamilton wrote:
Hi,
For REGISTER and SUBSCRIBE requests coming from UACs, is it possible to
make Asterisk skip authentication even if a secret is defined in
sip.conf for the peer; i.e. similar to insecure=invite for INVITE requests?
If I leave secret blank,
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE
Thanks Kevin.
Seems like remotesecret takes over if secret is not defined - I'll do further
tests..
The authentication for REGISTERs and SUBSCRIBEs are done at a sip proxy
(opensips) - I'll try to take care of the UAC authorization request for NOTIFY
there (if possible).
Regards,
Matt
Hi,
I am using ISDN phones which have a Park call button. The idea is: you
are on a call, push the button and hang up. You can then go to another
phone and pickup the call without having to remember parking slots, etc.
Unfortunately I cannot figure out how to get it to work with asterisk. I
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
2012/2/14, Luke Hamburg l...@solvent-llc.com:
Try TAPIRex
http://www.tapirex.com/en/
It's not free, but I've been using it with Asterisk + Outlook 2010
successfully. Users can also click on the screenpop and it will open up the
contact in Outlook. Pretty handy. You will need to make
In case this helps, when pressing the Park Call button, I get the
following with capi debug:
DISCONNECT_REQ ID=002 #0x037e LEN=0013
Controller/PLCI/NCCI= 0x1303
AdditionalInfo = default
CAPI: ApplId=0x0002 Command=0x04 SubCommand=0x81 MsgNum=0x037e
My apologies, I just realised I copied the wrong section of the debug
log. So once again, when pressing the park call button, I get the
following capi debug output:
CAPI: ApplId=0x0002 Command=0x80 SubCommand=0x82 MsgNum=0xe446
NCCI=0x1403
FACILITY_IND ID=002 #0xe446
On 02/14/2012 11:19 AM, Karsten Wemheuer wrote:
Hi Kevin,
Am Dienstag, den 14.02.2012, 09:46 -0600 schrieb Kevin P. Fleming:
On 02/14/2012 09:30 AM, Karsten Wemheuer wrote:
Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via
Hi,
Am Dienstag, den 14.02.2012, 11:32 -0600 schrieb Kevin P. Fleming:
This does appear to be a bug in Asterisk; please open an issue in JIRA,
and post the issue number here, so we can get someone looking at this
ASAP. Thanks!
Done, issue ASTERISK-19358. If I can do anything to test
On 02/08/2012 04:29 AM, Tony Mountifield wrote:
In article4f324279.70...@message-id.plonk.de,
Jakob Hirschj...@plonk.de wrote:
Raj Mathur (राज माथ�र), 2012-02-08 03:27:
Packets not going out on the same interface as the one they were
received on is a general IP issue, not just
Anyone have an H.323 trunk tied between their Avaya and Asterisk box that
works? I am having some issues trying to get the two systems to connect. I
am using the ooh323 channel to try to make the connection between the two
system. I have all my configs if anyone would like to look over them. If I
On Feb 14, 2012, at 14:56 , Dustin fails wrote:
Anyone have an H.323 trunk tied between their Avaya and Asterisk box that
works? I am having some issues trying to get the two systems to connect. I am
using the ooh323 channel to try to make the connection between the two
system. I have all
I'm wondering how one might implement a transfer where a receptionist
introduces a caller to the recipient in a 3-way conference before hanging up,
leaving the other two parties connected. Something like this, from the
perspective of the customer:
Customer: Hi. I'd like to buy a widget.
As I read this, this is a regular attended transfer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost
Sent: Tuesday, February 14, 2012 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial
On Feb 14, 2012, at 15:34 , Danny Nicholas wrote:
As I read this, this is a regular attended transfer.
No, as I understand an attended transfer, there is no 3-way period where the
receptionist introduces the caller to someone else. In an attended transfer,
from the caller's perspective, he's
I am on a difinity system, Communication Manager version 5.2. Trying to use
asterisk as my voice mail server and get rid of my Intuity Audix.
On Tue, Feb 14, 2012 at 3:02 PM, Phil Frost p...@macprofessionals.comwrote:
On Feb 14, 2012, at 14:56 , Dustin fails wrote:
Anyone have an H.323 trunk
Hi,
Does anyone how I could extract redirected number from a sip packet
I have redirected a cell to a second cell which also rings a sip trunks and
wish to route the call per rdnis
The rdnis variable brings the first redirect (divert) which is the second cell
but the first number also appears
No, as I understand an attended transfer, there is no 3-way period where the
receptionist introduces the caller to someone else. In an attended transfer,
from the caller's perspective, he's talking to the receptionist, then he's on
hold, then he's talking to someone else. No different from
I think you can do the same thing with most Polycom phones.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Tuesday, February 14, 2012 4:11 PM
To: Asterisk Users Mailing List - Non-Commercial
On the snom too
Create a conferance and then press the transfer button. That will join the
parties and release the receptionist
-Original Message-
From: Andres and...@telesip.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 14 Feb 2012 17:10:38
To: Asterisk Users Mailing
On Tue, Feb 14, 2012 at 3:10 PM, Andres and...@telesip.net wrote:
using the Cisco-Linksys SPA Phones you would:
1) Receptionist Answers Call and hits 'Conf' button.
2) Receptionist makes call and when answered hits 'Conf' again.
3) Now everybody is talking
4) Receptions hits 'Join'
In case anybody was following this thread, or someone Googles it in the
future, here is the solution:
This worked fine with Polycom firmware 3.3x:
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
For firmware 4.0+, apparently I needed to add info=, i.e.:
exten = s,n,SIPAddHeader(Alert-Info:
i tried it and it wont work with rtcachefriend=yes
On Fri, Feb 10, 2012 at 11:56 PM, JR Richardson jmr.richard...@gmail.comwrote:
I am facing an issue with Peer registration in my asterisk server .
I am using asterisk version 1.8.5.0 and using SIP real-time
architecture.when i am doing
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