I actually have a phonejack, not a linejack. So I probably can't help.
What does the linejack config file look like?
You need a context specified (like 'default') that has a 's' extension
(aka start) that answers the call.
Something like:
[interfaces]
context=default
mode=fxo
format=g723.1
dev
Derek Bruce wrote:
depends on which Cisco softphone you are refering to... they have a few
different versions... including an NBX version which will not work with
Asterisk...
- Original Message -
From: "Tim Sailer" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Tuesday,
James Sizemore wrote:
exten => 6500,1,Answer
exten => 6500,2,Wait,1
exten => 6500,3,VoicemailMain2
Or should I say, "Me too!"
Is this the bug for the case in question?
"CSCed48311: Media takes 0.4 sec to be set up"
Thanks.
-Andrew
Yes the problem is that when making outgoing calls, there is
Steve Creel wrote:
On Wed, 10 Mar 2004, John Fraizer wrote:
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty muc
Steven Thomas wrote:
*Hi,*
Has anyone tried this Wireless SIP phone with Asterisk? If so, any
limitations? Thanks.
http://www.bcm.com.tw/product/productIS.htm
This WiFi phone and the Zyxel Prestige that was just mentioned both look
like
the Pulver WiSIP phone. Since that works I would tend
Jonathan Moore wrote:
I have a client that would like to purchase 12 IP phones for an office
environment. We were planning to purchase the Snom 220s, but apparently they are
still not available in the US. The new Sayson 480 also would fit the bill, but
won't be available until April. They have loo
Andrew Gillham wrote:
wrote:
Hello all,
I'm looking for advice for codec that works best for asterisk.
Anyone has real testing with all codecs, specially with G.729. I
have tested with single call on few codecs that come with asterisk by
using IPTraf and the rate as of below:
ul
wrote:
Hello all,
I'm looking for advice for codec that works best for asterisk. Anyone
has real testing with all codecs, specially with G.729. I have tested
with single call on few codecs that come with asterisk by using IPTraf
and the rate as of below:
ulaw 64 Kbps, sample-based Also k
Chris Clifton wrote:
I would buy this, but my 7960 is using g729a on both lines that I'm dialing
out on (to conference), my * installation is licensed for 3 g729 channels.
What codecs are you using ? Is there a conference config in the 7960 that
I'm missing ?
I can make inbound and outbound calls
Jeremy McNamara wrote:
Typical version skewTry linking to the kernel source that is
actually running on the box.
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /
I'm having trouble getting zaptel to work on Debian Testing (Sarge) with a
2.4.22 kernel.
The errors I am seeing with 'insmod zaptel.o' are:
./zaptel.o: unresolved symbol devfs_unregister_R1c83d91a
./zaptel.o: unresolved symbol remove_wait_queue_R1bc53d4c
./zaptel.o: unresolved symbol __pollwait_R8
Senad Jordanovic wrote:
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indi
Steven Critchfield wrote:
Please read past rants about the action you took to create this message.
Hint: You broke the thread by replying to an unrelated thread.
Could all of the thread police please just reply personally to the
offending party?
The amount of people interested in the rant is pr
Pavel Litvinenko wrote:
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple
cheap
analog phones plugged into the FXS ports. I am able to get * to ring
those
phones when a call comes in, but I canno
Robert Murray wrote:
Hi Mark
Did you or anyone else ever find a satisfactory solution to this? Are there any phones which provide voice through the serial connection?
What about the nokia card phone - does it have open source drivers?
Cheers
Rob
On Sat, Jul 13, 2002 at 10:31:53AM -0500, Mar
Dan wrote:
Hi,
- Original Message -
From: "Patrick Cantwell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 19, 2003 6:48 AM
Subject: RE: [Asterisk-Users] Updated iaxComm binaries available for WinXP,
Red Hat 9.0
I am having the same problem.
I'm running it fr
Andrew Joakimsen wrote:
The X100P cards have horrible echo problems. I've heard talk about this
being fixed, but havent seen anything done about it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Larry D. Black
Sent: Saturday, November
Eric Wieling wrote:
"show application disa"
Thanks. I think the reason I have been having so much trouble is that
my initial DTMF is not recognized and I have been hitting it again, holding
it down, etc and my authentication has failed.
My X100P context (now) looks basically like this:
exten
Does anyone have any decent working examples of providing a means for a
remote caller to authenticate and get "internal" access through the PBX?
I would like to be able to call in with my cell (or any other line), enter
a code and then be able to dial just like if I had picked up my IP phone
at hom
Brian West wrote:
Asterisk doesn't seem to support SPEEX all that well. Has anyone had any
luck getting it to work with X-lite?
Speex works perfect with IAX but not that crack headed x-lite stuff.
bkw
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Steve Underwood wrote:
Hi Thomas,
Unless you have a *very* specific need to use G.723.1 for
compatibility with someone else, forget it. It is pretty much an
obsolete product. Licencing is also a pain, as there is not patent
pool for it. G.729 is expensive to licence, but at least it is
relati
Gavin Hamill wrote:
On Mon, 2003-11-03 at 15:14, Eric Wieling wrote:
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
From what I remember when I looked into this about a year ago, this
isn't e
Todd Wallace wrote:
Does anyone know where I can buy SNOM or Cisco (new or used) phones
the cheapest. I need a few
Todd Wallace
Uh http://www.ebay.com/
-Andrew
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WipeOut wrote:
Bartosz Jozwiak wrote:
What is better?
Cisco 7960 or Snom 200 ??
Bartosz
How much do you want to spend and do you really want the name??
Haven't used the Cisco, way too pricey..
Snom's work great..
You have to look at what CODEC support you want also.
My Cisco 7960 phones w
A buddy of mine and I are trying to get his phone to register, but it
complains about not being able to connect to Call Manager.
According to the Asterisk console it has successfully registered.
If anyone has one working can you share the skinny.conf section?
-Andrew
__
[EMAIL PROTECTED] wrote:
I'm still stuck on this. The * is on a private network IP 192.168.0.7
and the 7960 is on 192.168.0.6. When I show 'sip peers' the 7960
appears to be registered although I can't make any calls and still get
the packet retries. I have also checked and re-checked the settin
[EMAIL PROTECTED] wrote:
Does any have a copy of the 30202 7960 firmware?
Thanks
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It is on Cisco's FTP server if you have a CCO account.
-And
Mark Spencer wrote:
For the record, you can turn on the speaker by doing "monitor=1" when you
modprobe wcfxo, e.g.:
# modprobe wcfxo monitor=1
Sweet, this is good to know! It would be nice to have a feature that
kicks the
monitor on for the first 30-60 seconds of a call or something. It wou
Andrew Gillham wrote:
[EMAIL PROTECTED] wrote:
Has anyone had any success using a Cisco 7905G phone with Asterisk?
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Mine works terrific. My
Louis-David Mitterrand wrote:
On Thu, Sep 04, 2003 at 10:56:10PM -0700, Andrew Gillham wrote:
Unless you're hoping to load Linux or some pirate image in the future,
there is no
reason to stay with the old code.
At least I have not experienced any new issues I can attribute to the
upda
Rich Adamson wrote:
I've can now get to the net & sip configuration panel, and I've got
SIPDefault.cnt
file that gets loaded at boot time. One of two 7960's is upgraded to
v4.4, however
the second one constantly reboots (only with v4.4) and never gets to a
usable
status. Using a sniffer trace t
Andrew Joakimsen wrote:
exten => 1000,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
This didn't work - what does the @1000 indicate?
It shouldn't be there, If it's defined as 1000 in sip.conf make your
dial string
exten => 1000,1,Dial(SIP/1000,20,Ttr)
You need 'SIP/[EMAIL PROTECTED]' if you w
Mike Ciholas wrote:
I'm shopping for good deals on Cisco phones. Forgive my
ignorance, but I spent over an hour at Cisco's web site and
Google trying to find a definitive statement as to the
differences between a 7940 and 7940G phone. Anybody know?
Should I prefer the "G" and why?
Related questi
Travis Johnson wrote:
Hi,
Ok, the phones are working and seem to be loading the correct info
from the tftp server. However, I am unable to make them perform any
functions (calling another extension, going to voicemail, etc.). I do
not have any telephony interface installed yet, only a single e
Paulo Mannheimer wrote:
Hi All,
Does anyone have any experience with the ArrayCom VoIP phone?
I bought one a couple of weeks ago, it used to work quite well with *
until I misconfigured one option.
I now cannot make it work anymore, because the phone boots up, doesn't
find a valid SIP gateway,
Joseph Finley wrote:
Rich, you can do **# and go into the Network config and hit **# again, you
should notice the LockPad come unlocked and then you can make changes. If
you upgraded, the default password is "cisco"
Joe
This key sequence does nothing in the newer SIP code. (if it ever worked
[EMAIL PROTECTED] wrote:
Has anyone had any success using a Cisco 7905G phone with Asterisk?
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Mine works terrific. My 7940 (non-G) phones work
Sorry for the late reply on this..
Ben Wern wrote:
Andrew,
Thanks for your help!
I did have the outgoing proxy set -- since I had FWD set up on line 1.
I removed all the FWD stuff, and the outgoing proxy. I altered the
entry to have the qualify, canreinvite, and nat lines and also altered
th
Ben Wern wrote:
I'm trying to get my Cisco 7960 configured to work with Asterisk, with
no luck. I'm sure I'm missing something very easy... since I know
others have this working. I've stepped through Andy Powell's excellent
"Getting Started with Asterisk", and it works for my X-Lite softphone.
Travis Johnson wrote:
Hi,
We are just getting started setting up an Asterisk VoIP server. We are
very experienced with Linux, networking, tcp/ip, etc. However, some
existing sample config files for using Cisco VoIP phones with this
server would be VERY helpful.
Thanks,
Travis
Microserv
_
On Thu, Jul 03, 2003 at 03:57:09PM +, WipeOut . wrote:
> IIRC asterisk by default will not participate in the call between two SIP phones..
> It will help establish the session to the correct UA and then have nothing more to
> do with it unless the call is transferred to another UA in which c
On Wed, Jun 11, 2003 at 07:44:47PM -0600, Dylan VanHerpen wrote:
> >
> >
> Well, I guess you'd have to include a disclaimer not to use it for
> marketing or political purposes ;)
Perhaps an 'abuse' clause is needed.
-Andrew
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