Displaying 20 results from an estimated 10000 matches similar to: "Caller ID from POTS lines"
2007 Jun 26
1
Modification of Caller ID based on context
Hi,
I have been looking for an example of accomplishing this, but I've been
unable to locate something similar to what I'm trying to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250). This is set in
sip.conf for each user. Each user has a local DID as well (hosted through
Vitelity, for example (555)111-2222). The problem is that this extension was
2015 Feb 25
0
eFax message from "POTS modem 2 " - 1 page(s), Caller-ID: 1-630-226-2563
[1]
JOIN THE eFax COMMUNITY
[2]
[3]
[4]
[5]
You have a new eFax message. To view your message, see your fax attached
or LOGIN HERE [6].
FAX DETAILS
Caller Id:
Received:
Type:
Number of pages:
Reference #:
1-630-226-2563 [7]
2015-02-24 16:40:51 CST
Attached in pdf
1
pdx_did4-056267093-45815523543-99
WITH EFAX, DID YOU KNOW YOU CAN:
*
2004 Dec 13
4
Caller ID on Snom 190?
Has anyone had success with the Snom 190 displaying caller ID name and
number on the Snom 190 on for an inbound call from *?
Right now our Snom's only show the caller id name, not number. I know
the number is transmitted from the Telco and received by * since the
number shows on the incoming call event at the * console.
We are not setting the caller id in the extensions.conf, simply passing
2005 Jan 24
1
Asterisk Dial Out Issues - POTS Line
I am having dial out issues and was hoping someone could shed some light.
The problem is Intermittent:
extensions.conf
[globals]
; Trunk Info for outbound calls via PSTN - See the zapata.conf file in
/etc/asterisk
TRUNK=ZAP/G1 ;Trunk Interface
;MSD digits to strip (usually 1 or 0), 1 = remove a leading 9
TRUNKMSD=1
; --------------------------------------------------
; [trunklocal] - Defines
2003 Dec 08
1
'asterisk' as caller id
Hi all,
I have asterisk installed with chan_capi, an isdn fritz card
and a snom 200 phone connected to the asterisk box. The asterisk version is
CVS-11/26/03.
When someone calls me I see his msn number (caller id) on the display of my
snom phone.
But, when he hides his phone number and he calls me, the word
'asterisk' is displayed on my snom phone instead of his phone number.
I want to
2008 Jul 14
4
Zaptel problem with pots lines
Hi,
I'm trying to get up and running a TDM400 with a standard italian pots
line but i'm having
problems at getting asterisk to detect when the line get answered on
outgoing calls.
I'm using asterisk 1.6 beta 9 with zaptel 1.4.11.
I tried with and without answeronpolarityswitch=yes but it didn't change
anything at all.
With callprogress=yes answer get never detected.
With
2008 Dec 21
2
Outbound fax issues
Hello all.
I have the following setup:
Fax machine
|
Sipura SPA-3120
|
SIP 100BaseT
|
Asterisk 1.4
|
IAX2 100BaseT
|
Asterisk 1.6
|
ISDN PRI TE210P
|
Traditional Telco
The fax lands on the Internal Asterisk 1.4 box, the sip config for this
extension looks like:
[35081]
type=friend
secret=************
qualify=yes
port=5060
nat=no
host=dynamic
dtmfmode=rfc2833
2007 May 20
1
Caller ID matching
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number to
check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
> requested format = gsm,
> requested prefs = (),
> actual format
2006 May 23
0
A call from a call file always does a redial?
I have an issue with the Snom 360's (any firmware) and asterisk call
files. When you setup a call using a call file from Asterisk and the call
is connected, Asterisk will start to redial the call after about 5 minutes
when the conversation is already ongoing. (Annoying and it can only be
avoided by disabling call waiting)
I tried to reproduce the problem with a GrandStream phone and a
2006 Mar 21
1
Caller ID forwarding with Pickup() application?
Hi,
I'm using the Pickup() application for direct call pickup having the following
line in the dialplan:
exten => _*88XX,1,Pickup(${EXTEN:2})
It works OK, though I would like to have to get the original caller ID number
forwarded to the phone where I do the pickup and have it displayed during the
call. Currently the string *88xx remains on the screen of the phone I do the
pickup. It
2010 Feb 22
1
Caller ID question
Hiya - quick question..
When an external call is answered by an extension and the person answering the call wants to forward it to a different extension, is there any way to change the caller ID when the call is transferred?
If someone is transferring a call to me, I see the caller ID of the other person in the office. When the call is transferred, could the caller ID be set back to the caller
2007 Feb 19
2
Transfer Caller ID
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator transfers that
call to another employee, the Caller ID doesn't seem to do what I would
expect it to. I would expect it to show the original caller's ID.
Example:
John calls in from the outside using (213-555-1234) and he calls into
the asterisk system
2008 Jun 27
1
Asterisk, POTS and plain handsets
Hello,
I've spent a couple days searching and posted into the forum with no luck, apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS
line. Also connected to the POTS line are plain telephones, non SIP, just plain
old telephones. When one of the normal handsets goes off-hook,
2003 May 03
1
SIP & Caller ID & outgoing line
Hi all
I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example
in sip.conf:
register => andy@sip.mydomain1.org/1000
register => andy@sip.mydomain2.org/1000
2010 Apr 14
1
Ring Two Extensions Simultaneously with different caller ID values?
Hi All,
We're using Asterisk 1.4, and Cisco phones exclusively (mostly the 7961G, but a few 7911Gs and one 7912G for the time being-all running the SIP firmware image, plus a few analog extensions until the next capital funding cycle).
Each user has a phone at his or her desk, but there are also a growing number of "common area" phones (hallway, kitchen, conference rooms, data
2006 Feb 08
3
Two Lines, Two Businesses
After tinkering with a hand-knitted extensions.conf based on "Asterisk -
TFOT", I've now set up a server with Asterisk@Home and am experimenting
with it. I'd appreciated any advice from the more experienced list members
about which way to proceed.
We (my wife and I) have two separate micro-businesses with two POTS lines
plus fax. I'd like to have inbound calls on the two
2010 Dec 02
3
+ on Caller-ID
I've had this discussion in the office and with some vendors, but no
one has a solid answer, hopefully someone here does.
What is the proper way to format a caller-ID here in the U.S.?
Is it:
+15705551212
or is it
+5705551212
I've always seen it +15705551212, but as I understand it the country
code for the US is 011, which to me would indicate you put
011-570-555-1212 as the callback
2003 Jul 09
17
caller id
Hello,
is it possible to change how are caller id on incoming call from isdn,
capi lines displayed od sip phones ? ( e.g. SNOM ) standard is
1234567@domain.net. I just want only 1234567 to be displayed. is it
possible ?
regards
Marian
--
SUNTEQ s. r. o.
Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic
Tel: +421-46-5430 754 # Fax: +421-46-5439 144
http://www.sunteq.sk/
2005 Jul 26
1
Supervised transfer over SIP to outside POTS lines
Hello all,
I am trying to complete my dial plan and have come up with an
interesting situation. My configuration is set up with 12 xlite SIP
clients on SUSE linux workstation. They are calling out via 10 analog
lines, TE110P->rhino 24 fxo.
It all works and dials out great ... but ... this unit was brought in to
handle the "global" office. So the help desk support on the Suse
2005 Jul 27
0
[PLEASE RESPOND] Supervised transfer over SIP to outside POTS lines
PLEASE RESPOND IF THERE'S A SOLUTION
I am trying to complete my dial plan and have come up with an
interesting situation. My configuration is set up with 12 xlite SIP
clients on SUSE linux workstation. They are calling out via 10 analog
lines, TE110P->rhino 24 fxo.
It all works and dials out great ... but ... this unit was brought in to
handle the "global" office. So the