Displaying 20 results from an estimated 2000 matches similar to: "Easy to Access Telephone Directory AGI"
2004 Jun 29
5
nat problem
hello,
i have trouble with nat + sip outgoing call.when make an outgoing call to a
sip gateway, i have no sound.
i have 2 sip gateway, one is asterisk.
asterisk is on public ip and private ip
other sip gateway is on public ip
phone are cisco and grandstream on private ip on the same subnet as
asterisk.
phone are connected by sip to asterisk (i have try with or without nat=yes)
incoming call
2007 Apr 12
8
test
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<TITLE>Message</TITLE>
<META content="MSHTML 6.00.2900.3059" name=GENERATOR></HEAD>
<BODY>
<DIV> </DIV></BODY></HTML>
2006 Apr 01
2
TO have ringing tone instead MOH
I need to avoid MOH on my asterisk box, so i need to have a ringing tone
when attendant transfer is made, or a call is on hold..
Is there any way to do that.
I did not see a simple way to do that.
Regards
2006 Jun 14
1
SPA-941 Disable call waiting or Disable Call waiting via asterisk
I'm trying to disable call waiting for Linksys SPA-941, but
unfortunately as far as I have seen, there are no parameters on the web
interface regarding this feature. I just want callers to hear the busy
tone when the called party is at the phone. Probably I can accomplish
this by using the "disable call waiting" in asterisk as well, but I have
not been able to find any
2006 Nov 17
5
spc.exe
Does anyone have a copy of spc.exe they could send me?
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2006 Apr 10
5
SPA-941/942 Bulk provisioning
Has anyone got any information on bulk provisioning of Linksys SPA-941/94s?
There is an overview in the admin guide but it refers to a different
provisioning guide that I haven't found anywhere.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>
kerryg@techdatapros.com
2006 Mar 29
3
SMS in Spain (it seems Protocol 2)
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
I am trying to send SMS in Spain using landlines. It seems that
app_sms.c only handles Protocol 1, but Spain and Italy are using
Protocol 2.
I have been searching in Internet without any results... anybody is
sending SMS from Asterisk (or any method) using Protocol 2? (so, it
seems,
2003 Nov 01
13
Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be
directed to. I have no problem to RTFM if I can find the FM...
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under RH8.x and I prefer not to go
back to RH7.3...
BTW, where would I find a useful FM?
David
--
David J. Sussman, MBA
email:
2006 Jun 09
3
Compiling SVN Trunk
I have the same problem on some modules.
For example app_math.so
[app_math.so]Jun 9 18:16:45 WARNING[19001]: loader.c:728
__load_resource: missing mod_data for app_math.so
Any help?. I have been looking , but nothing reasonable found.
Thanks
--
Alberto Sagredo
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2006 Apr 21
5
Separating Asterisk SIP extensions from dialing each other.
This is coming from an * noob. :)
I've got two customers, they both are replacing their phone systems with
VOIP, and we need to retain both their existing dialplans.
One has 5 extensions starting at 100, and the other has 10 extensions,
starting at 100.
Is there a way to have the same extension number twice in the same
asterisk system ?
They will have different incoming DIDs of course.
2008 Aug 12
3
COM32 Implementation Of GFXBoot
What?s the status of the Google Summer of Code?COM32 module implementation of GFXBoot?project? From what has been communicated to me, things have not gone as planned, and it?s currently in limbo.
2005 Aug 10
1
real-time priority
How to list real-time priority in Linux for an application (example
asterisk)?
--
#Joseph
2004 May 07
2
quadBRI & ISDN telephone
Hello,
We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a
ISDN telephone to this nothings happen.
What can I do?
My config files are this:
Zaptel.conf:
loadzone=es
defaultzone=es
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,1,3,ccs,ami
span=3,1,3,ccs,ami
span=4,1,3,ccs,ami
bchan=1,2
dchan=3
2004 May 18
1
How can I dial (0 + telephone number)
I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line.
How can I use Dial command to dial (0 + telephone number) directly?
I used
exten => 10,1,Answer()
exten => 10,2,Dial(Zap/1/0)
exten => 10,3,Hangup
It works, but I need to dial 10 and after the ring tone, the telephone number
How can I do?
2004 Aug 23
1
routing telephone calls via "switchboard/asterisk".
I'm new to this list.
Reading the asterisk handbook pdf (good work) but but still have some questions.
Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri.
We have a dedicated server which is connected to our telephone company.
It makes us able to call ordinary phones via VOIP using Ericsson DRG22.
Would like to make people able to call me - and get a message
"dial 1
2004 Dec 04
0
NewBie Question Modem Telephone -PSTN
Hello, I'm really new on Asterisk.
Is it possible to use a telephone machine connected to a modem as an asterisk voice input output device? I do not need PSTN connection.
The scheme i'm thinking about is;
user -> phone -> modem -> asterisk -> ip -> vice versa.
If it is possible can a user dial another asterisk user via the phone?
I've searched astersik lists but
2005 Jan 24
1
OT: pinout for "standard" telephone headset required.?
Hi,
I have a Cisco 7960G phone for which I know the pinout of the headset socket.
I have a couple of standard telephone headsets which I do not know the
pinout of.
I'd like to connect the two. If I have the pinout of a normal/standard
headset I can
rewire the ones I have to match the cisco.
Thanks
Mike
2005 Jun 01
2
IAX2 analog telephone adapter
Hello All,
I am looking for a IAX2 analog telephone adapter, just want to ask your
views on which ones are bad, good and the best.
Thanks in advance,
Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 67791117 Email :
dinesh@imcb.a-star.edu.sg
WWW: www.imcb.a-star.edu.sg
2005 Sep 30
1
Asterisk and telephone volume
Hello
I am using a Snom 190 and the quality seems OK. Trouble is the volume is
quite low and full volume on the Snom is still too low. Is there something
I can do on the asterisk to increase the volume?
Angus