Displaying 20 results from an estimated 49 matches for "x2120".
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2120
2005 Jul 25
2
Re: Asterisk-Users Digest, Vol 12, Issue 171
...be obtained on eBay for $9-$20, usually. Much cheaper price-per-port,
> although the TDM would give better expandibility.
You mean NON Digium X100P's. Digium no longer sells the X100P. The
cheap ones on eBay are "clone" cards.
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
2005 Jul 27
2
CVS Head No ringing on calling end?
Tie line is type em_w (old school stuff) to TE110P. Phone rings on the
asterisk side, but the calling party does not hear the ring through
sound. If I pick it up within the first two rings it goes through and I
can talk otherwise our old switch drops the call.
Anyhow...here is my config if anyone can shed some light on it. It used
to work with HEAD a few weeks ago.
-Matt
2005 Aug 08
2
Stun support
Hi * users,
I want to know if STUN suport is available with Asterisk.
Kindly let me know. I have posted this also in DEV list but none replied to
me.
thanks,
Somesh
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2005 Jun 30
2
Dial Option A(file.gsm)
Hello,
I am trying to let someone know that is being called from a specified location.
For that, the command:
exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm))
should let the called person hear Anounce.gsm as soon as he/she answers.
(Only calls with prefix 107 are given this notice).
The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every
2005 May 13
3
Poor volume on SPA-2100 due to asterisk?
I just bough a Sipura SPA-2100 to use with Asterisk. When I use the
analog handset plugged into the SPA-2100, the person on the other end
can hardly hear me.
I check the SPA-2100 setup and their is no mic/spk gain control. Is
this a problem with the SPA-2100 or with Asterisk? Any way for asterisk
to compensate for the poor audio level (if the problem is the SPA-2100)?
Thanks,
Mike
2005 Jul 24
11
super high bandwidth codec
I've just gotten off a skype conference call and it pisses me off that
the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
In a situation that you have the bandwidth to share is there something
that I can use for important calls when the situation warrants it?
TIA,
Dean
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An
2005 Aug 05
8
asterisk registered in ser proxy
is it possible to register asterisk in a sip proxy as
if it were a terminal (like a cisco ATA)? how?
Thanx
Jenna ;)
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2005 May 10
1
Asterisk PRI problems (Crashing when full)
We have been running into problems here, we have 2 PRI's when they
fillup, All channels in use, and we dial more calls asterisk becomes
unstable and crashes alot.
We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by
root@localhost on a i686 running Linux
I know I need to upgrade. Is this a know issue??
Kyle
2005 Jun 29
1
Dial ZAP Problem
I'm trying to get this zap dial to work. I want to send DNIS and ANI to
other system (ZAP/g2) at answer, while the caller hears ring (RBT).
I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2
T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should
send DTMF "*ANI*DNIS*"
exten => _XXXX,1,NoOp,${CALLERID}
exten =>
2005 Jun 30
2
[Asterisk-Dev] Developing an Application in Asterisk
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2005 Jun 30
1
Outbound answer on TDM400P
How come an outgoing call using my TDM400P immediately
say the call is answered? I'd like to be able to
detect when the call is actually picked up, is this
possible?
If this is normal with analog cards, does the same
thing happen with T1 cards?
-L
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2005 Jul 01
1
SIPGetHeader application in asterisk-1.0.9
hello
i want to use SIPGetHeader application in
asterisk-1.0.9.
Jul 2 00:04:33 WARNING[19575]: pbx.c:1293
pbx_extension_helper: No application 'SIPGetHeader'
for extension (default, 2000, 1)
Any one using this
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2005 Jul 02
0
(Simple?) ENUM Question
...ot)
*.3.9.3.0.0 IN NAPTR 100 100 "u" "SIP+E2U" "!^+*00393(.*)!sip:\1@pulver!" .
I would like to know if it's possible to have an ENUM entry that would
be the equiv of the Asterisk pattern _21XX (notice no dot).
--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120
2005 Jul 11
1
RTP traffic
Hello.
How can I check if the RTP traffic between two channels is bypassed?
Some * console command?
Thanks.
2005 Jul 20
3
IAXY with DNS name, not IP
Hello All,
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the config, not an ip. This does not seem to work if I try to set it as
such. Has anyone come up with a workaround or solution to this?
I want to be able to put it on the net, or travel with it, plug and go.
The issue is that I am using a
2005 Jul 20
2
Last two digits getting cut off?
We've just setup our A@H server, with our quad port card. Everything works
well so far.
One thing I notice is that when I leave the handset on the hook and dial a
#, all is well. If I pick up the phone and dial, it cuts off at 10 digits,
which is a problem if I need to dial 1+area+phone # (12 digits).
The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a
2005 Jul 26
3
Polycom digitmap question
via google, I found the reference regarding digit maps for the Polycom phones:
http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html
But I don't see any instruction for prepending digits to the number
dialed. Does anyone know how to prepend a digit to the number dialed (from
the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura.
i.e. Say I want to
2005 Jul 27
1
call failed: 499 Not acceptable here
Hi,
Please help me through to overcome this error. I have my asterisk server and a xlite client in 172.16. network. I opened up port 5060 in my enterasys wireless router for my asterisk server and also opened port 8000 and 8001 for my xlite client. When i tried calling a remote system which successfully connected to my asterisk server i'm getting the error as " call failed: 499 Not
2005 Aug 05
2
Zaptel warning
Hi all,
When I was making calls from an IP phone, through a X100P, to PSTN,
the following error was encountered.
-- Executing Dial("SIP/25086937-aa6c", "Zap/1/91713545") in new stack
-- Called 1/91713545
-- Zap/1-1 answered SIP/25086937-aa6c
Aug 6 00:12:53 WARNING[3983]: chan_zap.c:4717 zt_indicate: Don't know
how to set condition 16 on channel Zap/1-1
--
2005 Aug 08
3
FXS - Don't want a Dailtone
Does anyone know of a way to make a standard analog phone plugged into an
FXS port do something other than get a dialtone when you pick it up? For
example, if the phone should automatically ring someone or play a greeting
when picked up without having to enter an extension?
- Robert
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