Displaying 20 results from an estimated 1000 matches similar to: "OT: XML Content Manager for Cisco 79XX Phones"
2007 Sep 04
1
Cisco 79xx XML Apps (was: Re: Cisco Directory Format)
Do you know where to find clear developers' guides (with some examples)
for developing apps that run *on* Cisco 79xx phones (especially the
7970)? Examples that can run against Asterisk (not CallManager) with SIP
firmware (not SCCP), and/or LDAP directories (or other open servers)
would be best.
On Sat, 2007-09-01 at 12:00 -0500,
asterisk-users-request at lists.digium.com wrote:
> Date:
2006 Feb 22
2
Cisco 79xx firmware
I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond.
Can anybody tell me where can I buy SCCP and SIP firmware for my phones?
BTW, I'm in Croatia (Hrvatska). I heard that location does matter.
P.S.
My local Cisco reseller wants to sell me
2004 Aug 29
2
AgentCallbackLogin by other means
Hi,
We?re looking at options for logging agents into the system programmatically
via Perl/PHP and I was wondering if anyone else is doing this and if so,
how. We're using AgentCallbackLogin now but would like to set up a web
interface instead. I've been looking at Asterisk::Manager and didn't see
anything relevant and wanted to ask the group before we dove into the
Asterisk source.
2004 Aug 03
2
VoIP experiences with Cable and DSL
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Hi,
A thought occurred to me;
Background;
In the early days of cable, the cable people seemed clueless to things like
over selling bandwidth. But as time went along they got it better and better
under control.
Of course their natural competitor, DSL, created a bigger demand for them to
get things under control.
Today cable is often giving
2006 Mar 15
6
Cisco phones and Linksys SRW224P
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable!
Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot
2006 Mar 01
9
MOH native files
Where can I find alaw, ulaw, gsm, g729 formats for native music on hold?
I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3?
Thank you for your time!
--
Tomislav Parcina
tparcina#lama.hr
2006 Feb 14
5
Call centre - * hang's up
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine.
In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0.
I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help
2006 Feb 09
4
Queue - check agent
I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call.
Thank you for your time.
--
Tomislav Par?ina
Lama Computers Split
2007 Mar 20
3
wrong values in duration and billsec in CDR
Hi to all,
I was looking in google and also in this mailing list, but I dont find the
solution to my problem, so I subscribe me to the list in order to post this
e-mail and find the solution.
This is the scenario:
GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) --------
VoIP Provider ------- Sip Phone or H323 Phone
The problem is that I am generating calls from SIP and also
2006 Feb 13
4
Voicemail - direct call
Hi list!
How to send a call directly to voicemail recording?
When I put this
exten => 313,n,VoiceMail,u221
Or this
exten => 313,n,VoiceMail,b221
In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible?
--
Tomislav Parcina
tparcina#lama.hr
2006 Jan 04
2
Ominiis Asterisk TAPI driver
I have foloved instructions at this web pages
http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call
contacts from Outlook. Now I have few questions. When I place a call, my
phone rings before * tries to dial out. Is it posible that * first dials
out, and when other side picks up, at that moment that my phone rings?
Another question, when I recive a phone call, can that
2003 Jun 17
3
sip.conf
HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register => user:password@host:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks a lot in advanced
michelle
-----
Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam
2006 Feb 27
5
res_features pickupexten
is where anyone who knows what is needed to get the pickupexten (*8)
running ?
gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff
I've activated it in features.conf (default *8) and also tested other
extensions
res_features.so is loaded
show features says:
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind
2005 Dec 28
3
voip-info: Asterisk record calls
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there
is "Example by Mojo". I have done everything he said and I have sox
package installed.
[root@pbx recordings]# sox -help
sox: Version 12.17.7
...
When I open this web page http://10.0.0.26/recordings/index.php I get
this: No Recordings Found
And there are recordings in /var/spool/asterisk/monitor
Do I have to do
2007 Feb 22
3
queue information into db
Hi
the new asterisk 1.4 supports to store queue log information directly
into a database? (like CDR) ?
thanks
2004 Feb 02
6
Transfer
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Hi,
As I've been unable to get app_transfer to work, could someone explain how it
is supposed to work?
Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1
dials ast2 using iax2 and gets instructed to transfer the call to a different
extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing
happens
2003 Dec 27
3
Vocera Communication Badge
Hi there,
yesterday I came across the "Vocera Communication Badge" and now I'd like
to know if anyone here has played with that thing (or even just seen it
in real life), and if a price tag can be found for this device?
Too bad they don't use SIP... ;-(
http://www.vocera.com/
http://www.heise.de/newsticker/data/tol-25.12.03-001/
Cheers, Philipp
** Wireless Specifications
2007 Mar 06
3
Micros-Fidelio - billing in hotel
There is hotel application weary popular in Croatia - Micros-Fidelio.
Now I need to connect Asterisk with this application for purpose of
billing. Thing is that hotel would like to give customer one bill for
every service that he used while he was in hotel.
Has anybody connected Asterisk with Micros-Fidelio? As I understand this
isn't some local developed application, it's something
2006 Jan 05
2
Asterisk CLI | more
What is command when I wona to list something page by page in * CLI?
Something that works like |less or |more.
Have a nice day!
--
Tomislav Parcina
name.surname@email.t-com.hr
2005 Dec 15
3
AoC (Advice of Charge)
Does Asterisk support Advice of Charge? I was told that my Telco sends
me billing signalization that way, and I wonder can I use it?
--
Tomislav Parcina
ime.prezime@email.t-com.hr