Don't know if this helps, but my installed 4-port fxo card has the
rj11 jack closest to the pci edge connector as zap/4, and the rj11
away from the pci edge connector as zap/1. The board is installed
and working, so can't look at much more.
> I have a TDM400P with 3 fxs an
Lars,
I could be quite wrong, but I think you only need a 'timing' source if you
want to use trunking over IAX. You can still use IAX without trunking if
you don't have any sort of timing device.
-Chris
On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote:
>Dear Sirs,
>
>Anybody ever tried running
On Sat, May 22, 2004 at 03:07:14PM +1000, Christopher Lee wrote:
> > Okay, doesn't mean so much to me, but it might help someone.
>
> Nor I. I've posted the results to the RT ticket I've already got open with
> Digium support in case it helps them as well.
>
> > > (gdb) x/5i $eip
> > > 0x4044e862
Lars Boegild Thomsen wrote:
Dear Sirs,
Anybody ever tried running SIP up against Voicepulse? On their
http://connect.voicepulse.com they claim they support both SIP and IAX, but
I can't seem to get SIP running. I have as mentioned before on this list -
huge problems getting any timing devices run
> Okay, doesn't mean so much to me, but it might help someone.
Nor I. I've posted the results to the RT ticket I've already got open with
Digium support in case it helps them as well.
> > (gdb) x/5i $eip
> > 0x4044e862 : fcomi %st(5),%st
> > 0x4044e864 : jbe0x4044eb40
> > 0x4044e86a : fsubr
Who do you use now?
David Hickman
TSG Computer Consulting - Auctions
314-865-4752 x2
On May 21, 2004, at 8:49 PM, Brian Cuthie wrote:
SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I could make IAX calls and see
On Sat, May 22, 2004 at 01:05:23PM +1000, Christopher Lee wrote:
> Hi Andrew,
>
> Here's the results:-
Okay, doesn't mean so much to me, but it might help someone.
>
> [snipped out most of the above symbols messages]
> Reading symbols from /usr/lib/asterisk/modules/format_g726.so...done.
> Load
Hi Andrew,
Here's the results:-
[snipped out most of the above symbols messages]
Reading symbols from /usr/lib/asterisk/modules/format_g726.so...done.
Loaded symbols for /usr/lib/asterisk/modules/format_g726.so
Reading symbols from /usr/lib/asterisk/modules/format_g729.so...done.
Loaded symbols f
On Sat, May 22, 2004 at 12:29:11PM +1000, Christopher Lee wrote:
> Hi,
>
G'day,
>
>
> I've been trying to get the G.729a beta codec running with my remote
> Asterisk box that talks IAX2 to my local Asterisk box.
>
>
>
> Digium fixed the problem I was having in registering the beta codec,
It very well could be email [EMAIL PROTECTED] and see if maybe they
are aware of this issue.
bkw
- Original Message -
From:
Christopher Lee
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 8:29 PM
Subject: [Asterisk-Users] G.729a beta
codec on old Pentiums
Hi,
I’ve been trying to get the G.729a beta codec running
with my remote Asterisk box that talks IAX2 to my local Asterisk box.
Digium fixed the problem I was having in registering
the beta codec, so that now works fine. I’ve removed the old
codec_g729b.so from /usr/lib/asterisk/mod
I have a TDM400P with 3 fxs and 1 fxo ports. I need to know which phone
connector corresponds to which module and also which port number. If we
are looking at the card with the PCI connector at the bottom, fxs/o
modules at the top and the phone jacks on the left - do the phone jacks
start at th
Let me get this straight - you moved on to a different provider because the
calls did NOT show up on your bill? :) :) :)
Just kiddin' :)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Brian Cuthie
> Sent: 22 May 2004 09:50
> To: [EMAIL PROTECTED]
> Su
SIP used to work fine with VoicePulse. But the funny thing is I could
never detect any signs that they were doing call accounting. I could
make IAX calls and see them show up in the CDR and the $$ deducted from
my account balance. But when I made SIP calls they appeared, by all
measures, to be
> cvs [login aborted]: connect to cvs.digium.com(65.38.23.22):2401 failed:
Well - since your DNS resolve correctly there are three possible options
really:
1. You have tried just at a moment where Digium's CVS server was down (I
have never experienced it to be down, but they could have been upgra
Dear Sirs,
Anybody ever tried running SIP up against Voicepulse? On their
http://connect.voicepulse.com they claim they support both SIP and IAX, but
I can't seem to get SIP running. I have as mentioned before on this list -
huge problems getting any timing devices running on some of my machines
Excellent! I'll give it a shot.
-brian
brian k. west wrote:
This time stamp issue is all gone.. now if everyone will just UPDATE!
bkw
- Original Message -
From: "Andres" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 4:14 PM
Subject: Re: [Asterisk-Users] RTP timest
Oops, Forgot the line to create the pvc0..
sethdlc create #
where # is dlci ##.
At least this is what I have gotten
M.
On Friday, May 21, 2004, at 08:22 PM, Michael A Rowley wrote:
Your system is using frame relay mode: ansi, timing none.
so I think nethdlc should be fine, but your sethdlc sho
Hey Guy,
I am battling with this, but from what I have figured out, yours seems
to be similar to mine... Read the "generic-hdlc.txt" file in
/usr/src/linux/Documentation/network/ directory. It was actually very
helpful for me.
Here's my two cents. :)
Your system is using frame relay mode: ans
Watch the CVS logs the update was put in stable TODAY for a timestamp issue
in chan_iax2.c
bkw
- Original Message -
From: "Andres" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 5:22 PM
Subject: Re: [Asterisk-Users] RTP timestamps
> brian k. west wrote:
>
> >This
On Fri, 2004-05-21 at 18:22, Andres wrote:
> It is not gone. We updated one more time all our Asterisk boxes l0 days
> ago (fresh checkout). And the problem was also evident (no other
> providers involved in the tests, just calls between all our boxes). We
> quickly put in place the rtp.c hac
Take a look at:
http://www.neuro.gatech.edu/users/cwilson/cygutils/dll-stuff/index.html
Cheers,
Rich
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joshua Colp
> Sent: Friday, May 21, 2004 4:24 PM
> To: [EMAIL PROTECTED]
> Subject:
I figured so, but searching for "upgrade" and "kernel" returned so many
irrelevant hits I threw in the towel.
;P
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> brian k. west
> Sent: Friday, May 21, 2004 7:23 PM
> To: [EMAIL PROTECTED]
> Subject:
All,
I am trying to configure hdlc support on T100P Digium card - everything
seems ok... but it just does not work at all.
I was able to compile all drivers, the light on the card becomes green
when I plug T1 link but I even can't ping default router IP... there
is no data coming back to me,
Yes you will need to recompile zaptel also. This is comonly talked about on
the mailing list.
bkw
- Original Message -
From: "Nik Martin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 5:17 PM
Subject: RE: [Asterisk-Users] Asterisk upgrade on production box
> I
brian k. west wrote:
This time stamp issue is all gone.. now if everyone will just UPDATE!
bkw
It is not gone. We updated one more time all our Asterisk boxes l0 days
ago (fresh checkout). And the problem was also evident (no other
providers involved in the tests, just calls between all our
Is it normal for asterisk to have to be recompiled when you upgrade your
kernel? I fetched a new kernel yesterday, taking me from 2.4.25 to 2.4.26
I built yesterday, and rebooted this afternoon. After upgrading, none of the
asterisk modules would load. I assume they are dependent on the kernel th
Check your sip.conf
Make sure the dtmfmode is set the same as the phone.
I had this before.
Usually to dial an IP address you have a keystroke before you enter the
address.
I think on a Grandstream phone you press the menu button then the IP
address.
Dave
-Original Message-
From: [EMAI
This time stamp issue is all gone.. now if everyone will just UPDATE!
bkw
- Original Message -
From: "Andres" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 4:14 PM
Subject: Re: [Asterisk-Users] RTP timestamps
>
> >
> >VC: My phone is broken: I get no audio.
>
Brett Nemeroff wrote:
Hi All,
I'm tring to do some DB operations before and after a call. I see the
'g' option in dial to continue in context if the destination hangs up,
but what if the originator hangs up?
Basically I do a DB get/put before the call is placed. After the call is
completed I want t
On Fri, May 21, 2004 at 04:52:17PM -0400, Brian Cuthie wrote:
> 8) show channels to make sure nobody's using it
>
> 9) stop now
"stop gracefully" may be preferable
> 11) make install
It is possible to use "make install DESTDIR=/somewhere"
The install won't break anything even if for some reaso
Title: Message
Hi
All,
I'm tring to do some
DB operations before and after a call. I see the 'g' option in dial to continue
in context if the destination hangs up, but what if the originator hangs
up?
Basically I do a DB
get/put before the call is placed. After the call is completed I wan
VC: My phone is broken: I get no audio.
TAC: Show us a network trace.
VC: (presents my ethereal traces, with the non-counting RTP timestamps)
TAC: (laughing) NEXT!!!
I have not read RFC1889 (RTP) in detail, but I am positive that the
timestamp field was put there for a reason. Sure, maybe Cisco
Ok Elmer... just watch for those Wascally Wabbits.
bkw
- Original Message -
From: "Joshua Colp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 3:48 PM
Subject: Re: [Asterisk-Users] Asterisk Cygwin Port.
> The cygwin has me Brian... please - I must follow the wh
The cygwin has me Brian... please - I must follow the white rabbit instead!
AHHH!
- Joshua Colp.
- Original Message -
From: "brian" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 6:28 PM
Subject: RE: [Asterisk-Users] Asterisk Cygwin Port.
> I know who you
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joshua Colp
> Unfortunately yes. If these had worked it would have been much easier.
>
> - Joshua Colp.
Well, drat. I couldn't find it with a quick scan of what I've got on hand.
You c
I know who you are NINNY!!! :) Give up now before it kills you... its going
to be like that "Dawson Creek Trapper Keeper Futura S2000" on South Park and
take over your life and start to Hybrid with things...
Trapper Keeper ready for Hybrid
WE ARE TRAPPER KEEPER WE ARE ONE
bkw
> -Origi
Kingston claims 300K:
http://www.kingston.com/products/MKF_2603CF.pdf
http://www.kingston.com/literature/MKF_591EliteProCF.pdf
Sandisk Industrial claims 300K-2M "industrial", but only 100K "extended"
cycles (whatever that may mean):
http://www.sandisk.com/industrial_cf_card_specifications.html
H
Check the dial-plan of your SIP device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, May 21, 2004 3:57 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Making a SIP call
If someone could point me in the right directio
Hi,
Unfortunately yes. If these had worked it would have been much easier.
- Joshua Colp.
- Original Message -
From: "Dr. Rich Murphey" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 6:15 PM
Subject: RE: [Asterisk-Users] Asterisk Cygwin Port.
> > -Original M
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: None
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk Cygwin Port.
>
> Hello Everyone,
>
> For the past few days I have been working on porting ast
If someone could point me in the right direction I would much appreciate
it. Here is my problem:
My directions for my sip phone says to dial an ip address 12*34*65*78#.
When I dial that into my phone my asterisk server is only picking up some
of the numbers in the above example it would pick up 6
I'd like to propose a change to the voicemail remove script found in the
contrib directory of the asterisk source
Currently the find command looks like so
system('find '.$dir.'/'.$context.' -name msg.??? -mtime +'.$age.'
-exec rm {} \; -exec echo Deleted {} \;');
I'd suggest it be changed t
I'm sure there are better ways, but I usually do the following:
1) make sure my current source directory is named something else (see
step 3)
2) fetch cvs head
3) mv asterisk to something like asterisk_cvs_head_5_21. This keeps all
the old source trees around so that I can easily roll back to an
> What is the best way to upgrade a production asterisk box? make upgrade? I
> don't want my configs messed with, and need the process to go as smooth as
> possible. I fetched and built a new kernel last night, but haven't rebooted
> into it. I'll do that tonight, and then want to quickly upgra
What is the best way to upgrade a production asterisk box? make upgrade? I
don't want my configs messed with, and need the process to go as smooth as
possible. I fetched and built a new kernel last night, but haven't rebooted
into it. I'll do that tonight, and then want to quickly upgrade to th
Need to anounce that Im using sip to h323!
Is there any beter solution to do this ?
.
> Can you tell us in details what the problem is (or I didnt understand)?
> if the problem is on call forwardin you have to add the following line
> on the context you are using:
>
> exten => _0.,1,Dial(h323/${E
works god here.
cd /usr/src
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
> Fabio Donaggio wrote:
>> I have some problems with the download of Asterisk-addons.
>> > [EMAIL PROTECTED] src]# export
>> CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
>> [EMAIL PROTECTED] src]# cvs
I run the 65.38.23.22 mirror and its working fine... It has in the past went
silly but I have since corrected that error!
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Fran Boon
> Sent: Friday, May 21, 2004 2:37 PM
> To: [EMAIL
Fabio Donaggio wrote:
I have some problems with the download of Asterisk-addons.
> [EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
[EMAIL PROTECTED] src]# cvs login
Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot
CVS password:
cvs [login aborted]: connect to
Hekuran Doli wrote:
exten => _0.,1,Dial(h323/${EXTEN:[EMAIL PROTECTED]) so all calls
starting with with "0" will be forwardert to the gatekeeper`s IP adress
(gatekeepers-IP-address) Note:I use this for international calls so If I
want to dial 37744387555 I use 037744387555.
If you are using a gate
Can you tell us in details what the problem is (or I didnt understand)? if
the problem is on call forwardin you have to add the following line on the
context you are using:
exten => _0.,1,Dial(h323/${EXTEN:[EMAIL PROTECTED]) so all calls
starting with with "0" will be forwardert to the gatekeeper`
On Fri, 2004-05-21 at 12:39, George Pajari wrote:
> > A modern CF card has about 10,000 write cycles before it starts
> > failing.
>
> Don't know where you're buying that crap but quality CF modules have a
> per-sector lifetime of between 300,000 and 2,000,000 write cycles. On top of
> this, good
On Fri, May 21, 2004 at 11:57:29AM -0500, Roger said:
> Look at asterisk bug 1688 for more info.
Interesting, but I didn't see any discussion of the "blocked/anonymous"
versus "unknown/unavailable". It is highly desirable to handle the two
differently. This is probably specific to the Zap interfac
There were reasons for not giving access to the "admin area" via the
list, which contains the files needed to configure the UIP200. Since
the first phone was shipped from uniden 2 days before I received mine,
they created a username and password for me so I could get access to the
materials I nee
Hello,
I'm writing an AGI script that receives an incoming call, records the
caller's name, places an outgoing call and plays the name back then asks
if the call should be accepted or sent to voicemail.
I know I can use Dial to call another number and bridge the call, but I
need a much more adv
Hello,
i want to use asterisk as a gateway for H323-Phones.
But i cant get it work.
I'm using a gatekeeper on another computer. My IP-phone is registered there.
Does anybody can sent me an oh323.conf and extension.conf as examples?
Thanks in advance
Erik Bastian
--
NEU : GMX Internet.FreeDSL
A
On Fri, 2004-05-21 at 05:45, [EMAIL PROTECTED] wrote:
> Hello Everyone,
>
> For the past few days I have been working on porting asterisk to the win32
> platform using Cygwin. Asterisk is already capable of being loaded itself, but
> the modules are presenting a problem. On Windows a DLL can not
> A modern CF card has about 10,000 write cycles before it starts
> failing.
Don't know where you're buying that crap but quality CF modules have a
per-sector lifetime of between 300,000 and 2,000,000 write cycles. On top of
this, good quality CF modules have very sophisticated wear levelling
firm
Why don't you configure the IP address instead of leaving it 0.0.0.0?
While that should work, it's always a good idea to be specific about
your bindings.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stefan-Michael. Günther (in-put GbR)
Sent: Friday, May
Yes, I'm very familiar with Brian's "review". "Review" != "Tips or Tricks".
To quote from it: "Once I gained access to the appropriate admin guide, I whipped up a
few
of the configuration files on my TFTP server"
How did he gain access? What were the config files? Basic examples? You see,
he pla
Hi All
In Order to place a "call Me" button on a
webpage
which would you use ?
[A] Call
Me
or
[B] mailto:"sip:[EMAIL PROTECTED]">Call
Me
Then what s/w is required on the users end to make
the call so that it rings via an * server
to a sip phone (ip phone) registered with the *
Hi,
I have a strange error, that I haven't yet found in another posting on the
archive. Here's my sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incom
Ok - to solve the problem and mark incoming calls w/ Unknown if the
callerid is unavialable or block, as opposed as them showing up from
asterisk. I put the following in the general section of my sip.conf
[general]
callerid=Unknown
Look at asterisk bug 1688 for more info.
--
Rock River Internet
Given what we just went through this morning with iax/gsm, there is something
different about how your 7960's deal with the uneven timestamps. Therefore,
your tests with capi are not likely to be reflective of Vic's problem either.
Wish we know what was different.
> It doe
So what isn`t there a diference between the g729a and b? If they have
g729b and I g729a should it be a problem?
Best Retards
Hekuran
> g729a and g729b are stream compatible... the only thing that differ is
> how the stream is formed. Low complexity vs Medium to high complexity.
>
> bkw
>
>>
> PS
> Someone mentioned about some other problems with 2.05e. What kind of
> problems are they ?
> For me it would be important to know.
The biggest one I know of relates to the speakerphone. When you have the
phone set to ring on speakerphone but use headset to talk, an incoming call
will bump
g729a and g729b are stream compatible... the only thing that differ is how
the stream is formed. Low complexity vs Medium to high complexity.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hekuran Doli
> Sent: Friday, May 21, 2
Hello Everyone,
For the past few days I have been working on porting asterisk to the win32
platform using Cygwin. Asterisk is already capable of being loaded itself, but
the modules are presenting a problem. On Windows a DLL can not access any
functions of the program it was loaded from, which
It does look like stable can send the wrong timestamps but that wont fix the
capi issue as it will have to be updated also. But in my testing it didn't
mess with the audio.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of brian
>
Title: RE: [Asterisk-Users] Dial Plan Format Strings
I
currently have about 40 users up on Asterisk and it is working great. One issue
I have though is the inability to conference calls/3-way calling on my SNOM
200 phones. Whenever I press the CNF/TRAN button on the phone, it just
drops the
Ok that fixed it. But why all of a sudden did it start doing this after
I updated? Anyidea? It had been working fine for a few months.
Kyle
Philipp von Klitzing wrote:
Hi!
On my SIP softphone, when I stop speaking the audio stops. So if im not
talking I cant hear the other person.
FAQ!
X
Philipp von Klitzing wrote:
Hi!
exten => s,1,gotoif,"$[${CALLERIDNAME} = \0]?2:3";
Do this:
GotoIf($[foo${CALLERIDNAME} = foo]?2:3)
Got a syntax error,
Tried
gotoif,($[foo${CALLERIDNAME} = foo]?2:3)
Still no dice.
--
Rock River Internet Roger Grunkemeyer
202 W. S
bkw what can I do if my provider is using g729b? Im using the new one but
dosen`t seems to work correctly!
> The one on the website is the only one you need. If its not working for
> you contact digium.
>
> bkw
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>>
Care to find me on IRC and let me look at it and see if I can fix it?
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Vic Cross
> Sent: Friday, May 21, 2004 9:04 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] RTP tim
On Fri, 21 May 2004, brian wrote:
> Get ethereal traces where we can see timestamps and we can figure it out.
I did. See a post a few days ago (in the AArgh thread). But I'm the
"chan_capi problem" guy... :(
> Other than that I can't reproduce the problem, trust me I HAVE tried.
Appreciate i
> That's not even close to an acceptable solution. Try this for
> perspective...
> A new service provider is getting ready to provide iax termination service
> to asterisk users throughout the world. Lots of systems and agreements in
> place. There is "no" usable cvs code to start the offering. You
Hi Chris,
Here is a copy of the extensions.conf logic I use to accomplish this.
exten => 300,1,Playback,sipextn
exten => 300,2,Read(DSTNE||8)
exten => 300,3,Playback,sipip
exten => 300,4,Read(DSTNI||15)
exten => 300,5,Playback,sipport
exten => 300,6,Read(DSTNP||4)
exten => 300,7,Cut(DSTNI1=DSTNI,
The one on the website is the only one you need. If its not working for you
contact digium.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hekuran Doli
> Sent: Friday, May 21, 2004 7:52 AM
> To: [EMAIL PROTECTED]
> Subject: Re:
> > Problems (just for bkw):
> > 1. Stable does not include the "major" iax2 fixes that were corrected last
> > month 2. Head has a iax2/gsm sequence number issue
> > both result in choppy audio when transcoded to sip on Cisco phones.
> >
> > As has been noted MULTIPLE TIMES on this list, there is
> I presented a stopgap solution to this list earlier this week (TDMoE
> between
> two * boxes to give you clean RTP timestamps to the Ciscos connected to
> *1,
> and *2 to the world)
>
That's just silly to have to do this. Collect me some timestamp info so we
can see this invalid timestamp non-s
> You're not the only one with this frustration / show stopper, and
> regardless
> of what bkw says, its a serious problem that for whatever reason, a few
> arrogant people refuse to acknowledge.
Get ethereal traces where we can see timestamps and we can figure it out.
My phones aren't doing this.
Hi Chris,
FWIW I have a very crude version of what you are looking for. I've been
using it since last July for testing purposes.
It plays a beep, you enter the sip extension number aaa, then it plays
another beep, you enter the ip address bbbcccdddeee, then it plays another
beep, you enter the por
Chris Stenton wrote:
Olle,
I looked at this a month or so ago and what you outline worked for anonymous
sip phones but it did not work for anonymous * sip connections. It would not
go to the context in general, you had to have a valid extension for some
reason.
Add a match-all extension in the co
> Problems (just for bkw):
> 1. Stable does not include the "major" iax2 fixes that were corrected last
> month 2. Head has a iax2/gsm sequence number issue
> both result in choppy audio when transcoded to sip on Cisco phones.
>
> As has been noted MULTIPLE TIMES on this list, there is no realisti
> > As you already know, the timestamp problem with cisco phones dropping
> > packets has been associated with at least two channels.
> >
> > Although iax & capi are getting hammered, the real issue seems to be
> > that no one has opened a high sev TAC case to fix the root problem.
>
> If I was
Olle,
I looked at this a month or so ago and what you outline worked for anonymous
sip phones but it did not work for anonymous * sip connections. It would not
go to the context in general, you had to have a valid extension for some
reason.
Chris
- Original Message -
From: "Olle E. Jo
On Fri, 21 May 2004, nicolas wrote:
> I want use iLBC and have following in mind, please help me is it possible ?
>
> ISDN <-(ALAW)-> * <-(ALAW)-> SNOM
> SIP <-(iLBC)-> * <-(ALAW)-> SNOM
>
> 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC
On Fri, May 21, 2004 at 11:24:04AM +0200, Philipp von Klitzing said:
> Hi!
>
> > exten => s,1,gotoif,"$[${CALLERIDNAME} = \0]?2:3";
>
> Do this:
> GotoIf($[foo${CALLERIDNAME} = foo]?2:3)
Not knowing how the low-level caller ID works, how do you differentiate
between "anonymous/private" where the
On Wed, 19 May 2004, Rich Adamson wrote:
> As you already know, the timestamp problem with cisco phones dropping
> packets has been associated with at least two channels.
>
> Although iax & capi are getting hammered, the real issue seems to be
> that no one has opened a high sev TAC case to fix
Anyone written an extension that will take a 12 digit number, convert it to
an IP address so that you can make a sip call to it.
Chris
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To UNSUBSC
I use this setup for users to set call-forwarding remotely to another
extension.
exten => *76,1,Read(extfrom,fwd-ext-from)
exten => *76,2,Read(extto,fwd-ext-to)
exten => *76,3,DBput(CF/${extfrom}=${extto})
exten => *76,4,Hangup
Marv Horst
Kekin Dand wrote:
Philipp,
I already have that call-forwar
Sorry, didn't see the attachment up until now, will try it out later
today... Thanks a lot.
Duane wrote:
Maron Kristófersson wrote:
I guess a http scraper would be a legal way of mass-configuration.
Anybody created such a script and is willing to share?
Not my script, and I can't remember where
Thanks, I've been searching but can't seem to find the link, do you
remember the name of the thread?
The phone does the request for the cfg.txt as weell as cfg, but the
problem is that the format of that file seems unknown. There is some
information in the wiki with a suggestion of the format
Yes. That is what the two lines look like. It has been the same error
since those were changed to get rid of the PSTN-1 variable.
Scott
- Original Message -
From: "Leo Ann Boon" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 20, 2004 7:24 PM
Subject: Re: [Asterisk-Users
> I rather think this is a soundcard issue - try a different brand in
> *both* computers or call a hardphone with your softphone.
>
> Cheers, Philipp
I think you are right because this lag is different when using different
sound cards.
Navnit
___
Aster
You have to remove the the /var/lib/va-certificate if you want another
license. Can you mail me your codec_g729b.so couse the one from the digium
ftp dont work.
>
> Hi all,
> Are there any way to clean codec_g729b license ffrom Asterisk. I
> would like to clean a license to install other mor
Hi!
X-Lite: Menu --> Advanced settings --> Audio --> Silence
set keep transmitting after silence to 1 or something like that
Cf
- Original Message -
From: "Philipp von Klitzing" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, May 21, 2004 11:24 AM
Subject: Re: [Asterisk-U
Just checked it and its working...as of this time, 05:51 EDT
At 12:52 5/20/2004, you wrote:
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
Hi,
> > > enough get redirected to human consultant. There should be
> > > possibility for supervisors to connect to ongoing conversation.
> > > Expected traffic will not exceed 30 concurrent calls.
>
> Look at "ZapBarge" for the listening-in. As usual the Wiki is
> your friend. Also I assume
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