Displaying 20 results from an estimated 600 matches similar to: "re: questions about sip requests to asterisk 1.2"
2005 Mar 03
2
Re : Calling card platform
We are using a platform from AmarFone Inc. It great full featured ,
everything you want to run a calling card and does not cost your a lot
of money. Their support is awesome. You can contact them at
sales@amarfone.com.
Ehsanul Karim
2005 Jul 02
3
call forwarding, most basic case
hello all,
i need some help and after trying the wiki i'm even more confused than i was.
i'm trying to set up call forwarding and running into problems...
i want the most basic call forwarding imaginable.
1. caller dials extension (say, 154)
2. dialplan is updated to forward caller's extension (based on
CALLERIDNUM) to voicemail, instead of ringing his endpoint.
3. caller is
2007 Oct 22
1
app_swift issues
Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine, from the command line i can do swift "hello there" -o test.wav and
then
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2005 Mar 23
3
Need some help
Hi all
I have a couple of questions maybe you guys can help me with them
I have sip phones , SER server , Asterisk.
what is the best way to do that (also with accounting and authentication).
which one of those options
1) sipphone -> SER -> ASTERISK -> SER -> PSTN
2) sipphone -> SER ->ASTERISK ->PSTN
on the first option i am trying to return the call to the ser
2007 Oct 12
1
question about PSTN pickup
hi all,
you'll have to excuse the ignorance (i'm a software guy, not a telcom
guy..)
Is there any way to know if a channel has been answered by an automatic
system (like voicemail) rather than a human being?
Specifically, I want to use a .call to make a call on a channel and only do
something if a person answers, not a machine of any kind. Is this even
possible, or is an answered
2006 Dec 11
1
re: L option in dial command
Hello all,
I'm having a bit for a problem with the dial command limit option. I have
the following dial command (executed from inside the a2billing agi)
AGI Script Executing Application: (Dial) Options: (
IAX2/username@voipjet/18005551212|30|HL(60000:20000:00000)0)
Now, from what i read in the wiki, this is supposed to limit me to one
minute (60000 ms), and warn me when there are 20
2005 Jun 22
2
Weird ring back
Hi guys,
I have a weird thing happening sometimes with users calling from a GrandStream phone through Asterisk onto a PSTN.
Sometimes after a user hangs up a call on a GrandStream phone the phone starts ringing after a couple seconds.
When the call is answered there is no one there.
Anyone had this before ?
Kindest regards
David Wilson
_______________________________
D c D a t a
Tel +27 33 342
2006 Jan 30
0
re: help with redirect from SER
hello all,
i have a problem, and i'm tearing my hair out...any assistance is
appreciated. I am trying to redirect from SER to Asterisk, both on the same
machine. In 1.09 I didnt need to set up a peer for SER, just
autocreatepeer=yes, and rewritehostport from SER as below, and asterisk
accepted the requests without a problem. When I updated to 1.23 requests
from SER to asterisk die quietly, no
2005 Feb 23
6
List tips for new subscribers
*spews coffee over keyboard*
- FUNNIEST - THREAD - EVER -
Also one of the most insightful.
Teddy, your gmail invite is on the way.
2005 Sep 10
4
Samba compatibility with NetAPP filers.
Jeremy
There is NetApp simulator that may help you !
Check now.netapp.com
-- Yair
2006 Apr 14
1
asterisk or ser
Hello:
I noticed in few references that asterisk and ser and complementary.
Meaning asterisk handles connections to PSTN and voicemail but SER is better
for routing SIP traffic.
Is anyone using just asterisk for production purpose. Meaning serving a high
number of callers.
Is it mandatory to use SER behind asterisk?
your feedback would appreciated.
-Gaid
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2003 Oct 07
2
I need your help....
Hello,
I have a problem, I can't install the package 'mgu74av2cdf'. I
downloaded the zip file, yet when asked the R console to install it
from a zip file, I got the answer:
"Error in file(file, "r") : unable to open connection
In addition: Warning messages:
1: error -1 in extracting from zip file
2: cannot open file `mgu74av2cdf/DESCRIPTION'
2006 Feb 07
3
No sound on 10% of incoming calls
Hello,
I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring
but I don't hear the caller and the caller doesn't hear me (all IP Phones
have the same problem).
This problem appear also if the call is directly send to the second E1 of
the digium card who is connected to an IVR.
It does not depand on the charge of the server (I have the problem with only
one call).
2005 Sep 29
3
Broadvoice inbound issues
My SIP seems to be configured correctly as I can dial out and my minutes
show up on my broadvoice bill, but whenever anyone calls my broadvoice #
inbound they just get a busy signal. I dont get anything in the logs saying
anything came in from broadvoice at all.
Has anyone had this/simmilar problem with inbound from Broadvoice? Any
suggestions?
Thanks
Neri
-------------- next part
2004 Oct 12
4
A question in R
I started to learn the R language, but I didn't suceed to use an external file.
Let say that I have an excel file called "test1.xls" in the directory
"C:/program files/R/rw2000/external_files" that looks like that:
name mark
yair 80
yosi 70 ...
In the appropriate directory I wrote this:
x<-read.delim("test1.xls")
or this:
2006 Feb 07
2
Asterisk with USB
Hello everybody! I've seen that you can connect your cellphone via
bluetooth, but I've a Motorola V300 and it doesn't have that feature,
so I wish to connect it via USB cable, is it pissible con use my
cellphone with asterisk like that? I 've not been able to find
information on how to do this, I'l appreciate any help.
Thanks in advance!
--
Facundo Ameal.
2006 Feb 06
2
Uniden UIP200 and Asterisk v1.2.4: problem not registering
Hello
We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200
have stopped working ever since.
We can make a call with the UIP200 to any other extensions, but it can not
receive a call. In fact the UIP200 always appears offline:
It does show up in asterisk a few seconds after the UIP200 reboot:
-- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200
but
2006 Apr 24
5
DO NOT REPLY [Bug 3718] New: RSync should verify permission/time before commiting a change
https://bugzilla.samba.org/show_bug.cgi?id=3718
Summary: RSync should verify permission/time before commiting a
change
Product: rsync
Version: 2.6.5
Platform: Sparc
OS/Version: Solaris
Status: NEW
Severity: major
Priority: P3
Component: core
AssignedTo: wayned@samba.org
2005 Sep 26
3
re: DTMF woes, continued
Hi Yair,
Please let me if you managed to fix the DTMF tone issue, which you were
experiencing couple of months ago. If not can you share any advancement.
I'm currently experiencing the same issue, I can make outbound calls but
DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5,
registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set
to rfc2833.