Displaying 20 results from an estimated 2000 matches similar to: "Using switch =>"
2003 Jun 27
2
Making calls from snom 100
Hello,
I`m trying to make a call from the snom 100( SIP mode) but whatever
number I dial I get a 404 error from Asterisk. Here are my configs and a
dump from "sip debug" . But if I make a call from a Zap line (see
extension 2382031), it rings the snom phone
sip.conf:
------------------------------------------------------------------------------
;
; SIP Configuration for Asterisk
2003 Jul 28
2
"immediate=yes or Compleate recieved" with intcoming calls with new CVS
I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot
recieve the the calls from the zaptel interface which is a E100P with
pri signaling.
That is something with asterisk becouse rolling back to version from
06/23/03 using the new libpri and zaptel fixes the problem.
Here is an exept from the config:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension
2003 Nov 28
4
call waiting disable in sip
Hello,
is there a way to disable call waiting in sip? I`m using grandstream 101
and even when the phone is in use I hear ringing in the headset. It is
pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Dec 12
2
Dlink DG-104SH
Hello,
Anybody has it working with asterisk? Could you share your experience (
good/bad)
Thank you
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Dec 02
2
incominglimit stuck in app_queue
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are inuse. typing reload on the console
resets this and they are again available for working.
anybody
2003 Nov 24
3
strange SIP authentication/authorization behaviour
When I have an ip hardphone username setup in my sip.conf :
[109]
type=friend
username=ipphone9
secret=bla-la
host=dynamic
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
defaultip=172.20.0.139
mailbox=109 ; Mailbox for message waiting indicator
callerid=ipphone9 <109>
callgroup=1
pickupgroup=1
and this user has a wrong password then calls are denied, but
2003 Dec 23
2
Asterisk + CRM
Hello,
Anyone aware of any CRM products projects that intagrete with *? Or that
integrate with any telephony products? Is there some open API for such
integration, or are they all proprietory?
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Nov 28
4
Mute button in Grandstream?
Hello,
Has anybody been able to get the Mute button work on grandstream? it
simply does nothing. Only Hold is avalable, which is not that good.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2003 Dec 25
1
return of the transfer to a busy number
Hello,
Can such thing be done through dialplan , that say I transfer a call to
an extension but it is busy, so that this call returns back to me.
Thanks
--
Anton Yurchenko<phila@dg.net.ua>
Digital Generation
2004 Mar 15
1
smbpasswd trying to add instead of replace attribut
I have some weird error with one of my samba installation.
When modifying samba password using smbpasswd, samba seems trying to add same attribute (instead of delete and add again), pls see the "MOD" from log file (from different domain) :
UNSUCESSFULL
Mar 15 17:10:53 hurricane slapd[27056]: conn=29489 op=1 MOD dn="uid=pwreka,ou=people,ou=purwakarta,dc=indorama,dc=com"
Mar 15
2003 Jun 23
1
Setting up the E100P
Hello,
I have an E100P, and in the zaptel.conf I have:
span=1,1,0,ccs,hdb4,crc4,yellow
fxsks=1-10
the light on the card is green( BTW what do all those states of the card
that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or
for the card?)
in the asterisks` zapata.conf I have:
[channels]
context=default
switchtype=euroisdn
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
2003 Dec 29
1
transfer with MGCP
Hello,
I`m try to make the attended transfer work Dlink DG-104S via FLASH, when
somebody calls my phone I pickup and press flash to get a second line to
call another extension. When I press flash I hear no dialtone, and only
a long and then small beep. When I try to dial digits I hear again those
long+short beeps, but the extension dialed is not ringing. If I pres
flash again I get back to
2003 Jun 25
6
snom 100 and GSM codec
Anybody has figured out why asterisk + snom have such bad quality using GSM?
When I use GSM I see such messages dumped on asterisk console:
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect
process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect
process 2 frames
WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect
2012 Oct 29
1
XML namespace control
Hello,
I am working with a database system from which I can retrieve these kinds of user defined fields formed as XML ...
<udf:field unit="uM" type="Numeric" name="facs.Stain final concentration">5</udf:field>
You can see in the above example that "field" is defined in the namespace "udf", but that the "udf" namespace is
2006 Jan 26
4
extension to extension dialing
Sorry for all the newbie questions. I really appreciate everyone's help
today.
Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken
2004 May 04
1
Asterisk and windows h.323 gatekeeper calling problems...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper....
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
dtmfmode=rfc2833
gatekeeper =
2003 Aug 20
1
IAX <> IAX trunking... DP cache?
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one
another using IAX/IAX2 trunks.
I've managed to get a semi-functional NAT Firewall working as a PBX
(with Asterisk running directly on the firewall itself), but there are
issues with bind()ing to various interfaces which is causing outbound
SIP issues.
To get around these issues, the idea is to do something like
2011 Feb 18
2
no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
2006 Feb 27
7
TDM400P digium card
Okay everyone -
I'm moving away from using sipura 841 phones. I'm starting to test with
Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but,
for now we have a digium tdm400P with 4 analog lines coming into it. So
my question is will upgrading the IP phones with my existing digium
tdm400 card be enough to satisfy my users ? or is it really a combo
deal needing to
2006 Aug 27
1
Simulations in R during power failure
Hi everyone,
I recently ran a simulation on a computer using R that was hooked up to a UPS. There was one time when the power was out for length and the computer shut down. I was worried that I had lost the simulation, but upon booting the machine up, I heard the processor kick in. It sounded like the simulation had resumed.
Does anyone have any experience with this? Because I live in