Displaying 12 results from an estimated 12 matches for "benharoosh".
2004 Oct 04
3
Cisco XML 411 Interface
Hi All,
Did anyone came across a 411 XML service I can feed to the "service"
button with XML?
Some other feed I can manipulate to XML query?
Assaf Benharoosh
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2004 Oct 05
3
C flag in Dial command
...ns.conf:
exten => 114,1,Dial(SIP/114,,C)
It shows in the lastapp:
cdr:
| 2004-10-05 13:16:02 | "112" | | 114 | intern |
SIP/112-3fb6 | SIP/114-0e7a | Dial | SIP/114||c |
6 | 3 | ANSWERED | 3 | | |
|
Assaf Benharoosh
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2005 Jan 07
7
Channel Variable
...now how to get the channel ID on the other side of the
call?
For example: When SIP/50 calls SIP/21, and the call is answered by
SIP/21 I get:
SIP/21-6735 answered SIP/50-b456
${CHANNEL} will show me SIP/50-b456.
Is there a parameter or a workaround to get the SIP/21-6735 part?
Thanks.
Assaf Benharoosh
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2004 Apr 10
1
VoicePulse 1-800 numbers sound problem
To whom it may concern,
When dialing out an 800 number (888,866,877) through VoicePulse IAX
you'll get a choppy sound. This is not due to a problem on your Asterisk
or your line- the bad sound effect occurs in VoicePulse. (just spend
lots of time finding that out)
Assaf Benharoosh
MCP, MCSA, MCSE
ab@franticllc.com <mailto:ab@franticllc.com>
Frantic, LLC.
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2004 Jun 01
2
HDLC
There were a number of messages about using
a digium card as a data t1 (router) kind of thing.
Is there a howto out there or a status that I can see whether
this is feasible or not?
Or is it feasible but very difficult?
--
respectfully, Joseph - (606) 477-2355 x140
------=============
2004 Jul 14
3
Vonage working with asterisk
atlast after working of 7 hours i got voange soft account working on asterisk.
2005 Feb 25
1
WebVMail Woirks but No Audio
Hi Everyone -
I have webvmail up and running, I can see the messages, forward them,
pretty much everything but listen to them.
Here is what I see in my logs:
192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET
/vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default
&password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1"
200 9438
2004 Jul 22
4
VSP? Looking for advice.
...9;s Topics:
1. Re: RAID/SCSI/IDE/SATA and a TE405P (or T100P) card. Should I
expect problems? (Scott Laird)
2. Re: * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
(Walt Reed)
3. Oops - Re: [Asterisk-Users] Asterisk and Linejacks (greg)
4. RE: Asterisk and Linejacks (Assaf Benharoosh)
5. RE: Astricon costs... (Adams, Gavin)
6. RE: Asterisk and Linejacks (daryl@introspect.net)
7. RE: SIP register and unregister events via Manager API
(Matthias Endler)
8. Re: D-Link DPH-80S vs * (Chris Shaw)
9. RE: app_dbodbc URGENT (brian)
10. Re: D-Link DPH-80S vs * (Chris Shaw...
2004 May 20
6
VoicePulse broken?
Is anybody else out there using VoicePulse Connect and having problems
this morning? I just noticed that they have absolutely no contact
information in their website.. just want to make sure I didn't break
something in my asterisk configs.
-fedl
2004 Jul 22
7
Asterisk and Linejacks
I found a message from you to the asterisk users mailing list from 2001. I was
wondering if you got (or still have) an asterisk system working with the
linejack? If so, would you be willing to assist me with mine?
I seem to have things working, and * says that caller ID is coming in, but I
can't get * to actually answer the call.
Thanks,
Greg
--
NetIO.org
2004 Jun 15
7
Voicepulse Down Again?
I can ping it just fine.
I am on gw5.voicepulse.com
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2004 Aug 21
0
Cisco IP Phone- disjoin conference
Hi all,
Does anyone know if it's possible to continue talking to one member of
an initiated conference call on Cisco 79xx ? In other words- disconnect
one of the parties.
Thanks.
Assaf
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