Daniel Nyström wrote:
Is it possible to turn off DTMF recognition (and all transfer services etc.)
pending on CallerID (or FXS channel)?
Some of the FXS channels I will setup soon, is going to work exactly like POTS.
It will be used by people not knowing their within Asterisk.
They will be pretty
Mark Phillips wrote:
Getting nowhere with Digium support. Trying to tell me that their
engineers are working on it and that it could be months.
Ask if you can ship them the box so they can actually reproduce the
problem.
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Paul Rodan wrote:
Using FireFly, all other codecs but G711 Ulaw is selected. But whenever
I place a call, I get:
Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband
DTMF is not supported on codec G.711 u-law. Use RFC2833
Umm, wtf? I thought Inband was ONLY supported o
Me wrote:
Can someone give me a clue as to why I keep hearing DTMF type beeps on
my phone calls. It sounds exactly like someone on the other end is
pushing a key on their phone but they are not!
Has anyone ever heard of this before? It use to happen once in a while,
today it's been happening a
David Shaw wrote:
exten => 510,1,Dial(SIP/510,20)
exten => 510,2,Voicemail,510
exten => 8500,1,VoicemailMain
exten => _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten => _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten => _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN}
David Shaw wrote:
Hello, When I dial out there is a long delay in dialing. Is this normal?
For Analog FXS ports the delay is there because Asterisk has to dial the
DTMF to the line. For any other technology a delay will only happen if
you have a poorly designed dialplan.
Example:
exten => _
Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.
That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Actually the issue seem
MJ wrote:
I'm having problems using the following.
[sip]
exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM})
[macro-test]
exten => s,1,Answer
exten => s,3,Flash
exten => s,3,Dial(SIP/${ARG2},30,t)
exten => s,4,Dial(SIP/${ARG1},30,t)
exten => s,t,Hangup
exten => s,i,Hangup
exten => s,h,Hangup
Paul Rodan wrote:
The BT100's do support conferencing, most SIP phones do. But how does your
Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
VoicePulse?
You basically need to make sure your Asteri
[EMAIL PROTECTED] wrote:
Hi, i cant make a three way call using grandstream phones (BT-100) and
asterisk using sip, is this supported or i need a zap interface?
The BT101 cannot to supervised transfers or 3-way calling.
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Olson, Dana wrote:
Actually, I do care, and I did search Google (albeit quickly) and I did look on
the hardware list as well as the VoIP wiki. Maybe one of the cards listed there
does what I need, but it wasn't listed like the QuickNet cards are. I thought
perhaps the feature list on the site wo
Michael Baird wrote:
It's only one guy who seems to attack each poster for not posting in a
manner of which he approves (there is one/two of these fellows on every
mailing list), don't let him ruin your day, this list is quite helpful
and many guys will give you a good answer without the extra atti
Andrew Kohlsmith wrote:
On January 20, 2005 02:15 pm, Steve Clark wrote:
I am dialing from one zap channel to a second zap channel. Is there a way
to keep the channel I am dialing to from generating a ringback tone.
exten => 1,Dial(Zap/1)
should not generate ringback...
exten => 1,Dial(Zap/1,,r
Olson, Dana wrote:
Are there any cards that work with * that do the VoIP-to-TDM processing
on the cards, with multiple interfaces?
The QuickNet Internet LineJack meets the description I believe, but it
only has a single FXS or FXO. Are there any cards that have more than
one FXS?
It's been a
Michael Greb wrote:
On Mon, Jan 17, 2005 at 09:52:45PM -0700, Joseph wrote:
exten => s,1,Authenticate(X)
exten => s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter "no-password" I can make a call but if I enter any number
instead of word "passcode" it wi
Howard Lowndes wrote:
Will Wait(n) still listen for DTMF input from the caller after there has
been a Background(some-message) prompt, or do I need to use
Background(silence/n) to still listen for DTMF?
The WaitExten and Read applications won't work for you?
There was a bug with codecs for a very long time with Asterisk. In
[general] remove the bandwidth= line (all it does is allow specific
codecs) and disallow=all and allow= for eac codec you want.
Joseph wrote:
When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf
Joseph wrote:
On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote:
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so the voice is
choppy, it is hard to understand anything.
Is it reliable/practical to
Stephane Ricard wrote:
Hi,
I have an external Motorola BitSufR Pro ISDN modem and an ISDN BRI line. Is
that possible to get this to work with Asterisk for dial in/out?
Somebody ever did this? Where should I start?
I guess you start by writing a driver for it.
Start with the chan_modem s
R A wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.
I need to now the name os de file or a specific category link where i can download it.
If you can send me the file is beter ;-)
Cico and Polycom phones both seem to be really great phones
Helder Rogério [MICROREDE] wrote:
Hi!
Is there any way to receive in * server a call from a Terminal adapter in
G.723/G.729 and then convert it to G.711?
I'm wondering this because I can only place all thru Broadvoice in G.711 but
most of customers have ADSL connection with 128k upstream, so the
Shaun Ewing wrote:
On Mon, 17 Jan 2005 10:23:57 -0500, Nabeel Jafferali
<[EMAIL PROTECTED]> wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.
What is IOS? Am I the only one who uses Cisco phones and doesn't know that acronym?
Internetworking Oper
Adam Goryachev wrote:
On Fri, 2005-01-14 at 14:38 -0800, Ben Greear wrote:
Hello!
I am trying to set up multi-link PPP using two T100P cards in one
machine, and 1 T405P card (the 4-port one) in another machine. I have
previously been able to get PPP working between the two T100P cards
in separate
Chris Polk wrote:
Any one have any solution for this?
We need to have the caller id information announced when the phone is answered.
for example
I am sitting at my desk, my phone rings.
I pick it up and hear call from 55 to except press 1 to decline press to
any help would be grately app
Thor Atle Rustad wrote:
I have been running Asterisk CVS for a good while. When I try to
install 1.0.3, asterisk won't start. Below are the last few lines of
output before Asterisk crashes. I ran "make samples" to start with a
fresh setup.
[app_realtime.so]Jan 15 17:42:24 WARNING[19841]: loader.c
Update your CVS
Xu, Duo wrote:
why linux/moduleparam.h is missing in the source? I
saw it in 2.6 source.
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Rushowr wrote:
But then the CLI puts out a message concerning congestion and kicks the call
directly to voicemail.
The problem is that the DIAL fails. You don't have a voicemail problem.
You have a problem with the peer you are trying to dial. The dial
fails, Asterisk considers it a b
Rushowr wrote:
Anyone else ever have the problem of asterisk picking up with voicemail
after one ring on an extension? I'm using free world dial up's IAX2 service,
and I can make calls but received calls get a voicemail pickup after one
ring. No decent answer on google, cannot find anything that se
Paul Rodan wrote:
Capabilities: us - 0x4(ULAW), peer - audio=0x100(G729A)/video=0x0(EMPTY),
combined - 0x0(EMPTY)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Jan 14 17:29:55 WARNING[81922]: chan_sip.c:2820 process_sdp: No compatible
codecs!
What throws me off i
Matthew Boehm wrote:
Eric,
Thank you for explaining this to me instead of being rude and bitching at
me about my lack of GPL understanding.
You caught me in an unusally good mood, that's all.
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Matthew Boehm wrote:
So are you telling me that you cannot use other commercial products in
conjunction with asterisk?
You cannot distribute a closed source add-on (except AGI) for Asterisk
without a commercial license for Asterisk. This is just standard GPL
stuff, not Asterisk sprcific.
___
Howard Lowndes wrote:
I have actually got a bit more cunning that this by using sipgetheader()
and sipaddheader().
The default user name is "asterisk", hard coded in chan_sip.c, so what I
did was to use sipgetheader() to get the From: header, then I cut() it
at the ":" character and the "@" charact
Dhennys Pestana wrote:
I'm trying to find a way to connect two (or more) extensions directly without
being kept in the middle during the conversation but it won't happen.
Asterisk will always stay in the SIP signaling path. It can get out of
the RTP path (only way to really see this is using some
Matthew Boehm wrote:
I got onto IRC last night. Was over 400 people in the #asterisk channel. I
asked a question about PRI stuff. Waited 30 min for response. Nothing. I
left.
Perhaps everyone was just tired of asking questions? I know I get tired
of answering newbie questions from people that don
At the CLI do "show application DISA" While you're at it do a "show
applications" so you know what applications Asterisk has.
Jim Van Meggelen wrote:
Is there a dialtone recording?
I'm thinking :
exten => s,1,background(dial-tone-recording)
exten => s,2,goto(,s,1)
I have not tested this, bu
C F wrote:
I know that according to the docs you are right. I'm just asking you
for a favor, if you have a digium TDM04B test this, and tell me the
result. When I tested it, it didn't work.
I let the call ring for 60 seconds. It was never answered.
fs-1*CLI>
-- Starting simple switch on 'Zap/1-
Paul Fielding wrote:
> Boy, I had a blonde moment back there, I was shooting from the hip and
in looking at my response realize the error.The one thing I am
wondering about, though, is the need for specifying context. I'm not
specifying any context in my mailbox= line and everything works
Paul Fielding wrote:
[EMAIL PROTECTED]
It occurs to me, do you have the numbered extension set the same as the
context name for the phone in sip.conf? For example, in my sip.conf,
the context names for each phone are [7001], [7002] etc. However, this
doesn't necessarily need to be true. If
administrator tootai wrote:
Hi list,
I setup my * to use the SIP 3k as PSTN FXO gateway with auth. I face a
problem with PIN authentification: after introducing the PIN code, I
shall terminate by pressing #, but asterisk takes this DTMF and I get a
"Sorry, this is not a valid extension". How can
C F wrote:
Kelly, when I tried this it didn't work for me. What ever I tried *
picked up. I know in theory this works, but have you tried it?
Asterisk will NOT answer the line unless it's told to by using something
like immediate=yes, Answer, Playback, Background, etc. I suspect you
have immedia
Craig Waddington wrote:
Two Asterisk machines, different CVS, both say "no application" MeetMe,
show application does not show MeetMe, when I browse to /asterisk/apps/
I notice that it is the only app that has not installed?
Do I need to install ZAPRTC first then try to install the MeetMe
applic
Adi Linden wrote:
I can do the dial command like this to give me a 20 second timeout
exten => _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],20)
But this also means that after 20 seconds of ringing it goes on the next
dialpeer. I would like to be able to set the timeout Asterisk wait to
establi
Mark Halverson wrote:
My local Telco uses B8ZSESF and does support PBX customizing ANIs on a per
call basis.
What I need to know is, can I use the SetCallerID command in extensions.conf
to transmit the DID# of the extension making the call with the TE410P or is
there a different one that does supp
Gilad Ben-Yossef wrote:
Samudra E. Haque wrote:
hello, using Asterisk, is there any clever way to provide answer
supervision based upon the received audio only from the FXO interface
(from a public PSTN switch that does not have battery reversal, or CPC).
In zapata.conf use either
busydetecgt=
Dan Adams wrote:
I was wondering, does anyone know if it is possible to have a stream of
audio coming from a Microsoft compressed audio stream fed to the caller
if they are placed on hold and if so how might this be done?
Write a shell script called mpg123, located in /usr/bin or
/usr/local/bin
John Middleton wrote:
Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk
from the digium web server - Whats the CVS command for a 'head'
install ?
Try "make datafiles"
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CClarke wrote:
Dear All ~
I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN & try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain
John Middleton wrote:
Could you please explain or tell me where it is explained the version
and contents of * that is retrieved with CVS.
I am wondering whether there is a change list or something. If you
tell me here I will update the Wiki ;-)
http://lists.digium.com/pipermail/asterisk-cvs/
__
PHP Mechanic wrote:
> Threeway calling is similar. You can make a small impromptu conference
that way with 2 internal phones and an external or 3 internal phones or
even 1 internal and 2 external calls on separate phone lines. All of
these are mixed inside of asterisk and the PSTN is non the wiser
GIBERT Frédéric wrote:
Hello,
I had installed several asterisk, but I every time had a problem with
callerID.
On each phones I don't reveive the first digit.
For example:
Caller 0672083516 called an IP Phone 0123456789. The IP Phone see 672083516
as callerID.
I think there is a patch for it, but I
Samudra E. Haque wrote:
Hello, anytime I make an IAX2 call to another peer, I am collecting CDR records
which are divided into small files, one for each accountholder customer that
makes the calls.
I have records of this nature:
""123456","1234567890","IAX2/[EMAIL PROTECTED]/5","2004-12-30 22:17:
mohammad wrote:
In IAX, because both signaling and rtp ports are uniqe, so Asterisk is always in rtp path. Am I right???/
IAX and IAX2 use the same port for signaling and audio. IAX and IAX2 do
NOT use RTP.
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Serge Schumacher wrote:
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
and when I call 550 I get this error and the MusicOnHold (exten =>
550,4,MeetMe(18|Md)) also doesn't work:
-- Executing Answer("SIP/ses-0730", "") in new stack
-- Execut
Serge Schumacher wrote:
Can it be that the MeetMe application is not installed by default even if
there is a meetme.conf ?
pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension
(from-sip, 550, 4)
If you don't have zaptel installed Astrisk won't build Meetme.
Paul Fielding wrote:
Hmmm I could certainly see that being the issue. If it is the
issue, though, then I think it's something that needs to be addressed.
In my opinion, Digium needs to address it, as well as the whole
provisioning via cli thing. I know Asterisk itself is a CLI oriented
pi
Brent Goran wrote:
For efficiency & reliability, when SIP transmits DTMF as non-audio data,
it uses RFC2833 or INFO.
My question is - (not knowing much about IAX2) - when IAX2 transmits
DTMF as non-audio data - is it also using RFC2833 and/or INFO, or it it
using some other IAX2-specific mechanism
Nicolas FOURNIL wrote:
Hello
I have done the following test-network:
IP-Phone <=> ASTERISK <==> ISDN <> PSTN Phone
(SIP) +
SER
When I'm calling from the PSTN phone to the IP (SIP) phone:
I cannot get
pick the
phone up
DSLink 200E freeze again.
ie. there wasn't any port 5060 on transactions.
I will have this DSL modem on my LAB asap and I will give feedback to
the list.
Thanks
Eric Wieling aka ManxPower escreveu:
On Cisco routers you can do something like "no nat sip fixup 5060"
you might not be able to do it, in which case you
will have to get a different DSL modem.
On Wed, 29 Dec 2004 20:00:28 -0600, Eric Wieling aka ManxPower
<[EMAIL PROTECTED]> wrote:
Rodrigo P. Telles wrote:
Hi Folks,
I've been having troubles with a DSL router (DSLink 200E) and SIP phone
Rodolfo Grave wrote:
Hello.
I'm having a lot of trouble using asterisk behind NAT. This is the
situation:
h323
Asterisk ///---> H323 Switch
NAT Router
I see that my h323 traffic is going out to WAN with my intern
PHP Mechanic wrote:
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash
or hold to work when using asterisk, which means I can't use three way
calling or the call waiting functions. I've tried using combinations of
hook flash button and "*0" on three different phones and I
Megan Willigs wrote:
Hi everybody
in new versions of Asterisk the RTP on SIP pass only througt the Asterisk,
not directly between the endpoints like olders versions.
What happened whit this feature? (reinvite)
Can you help me?
The the two legs of the call are using different codecs then reinvites
Rodrigo P. Telles wrote:
Hi Folks,
I've been having troubles with a DSL router (DSLink 200E) and SIP phones.
When I put any SIP phone (software or hardware) to work behind
that DSL router, it completely freeze.
I ready tech specs of that DSL router and it says that SIP protocol is
supported.
ie. I
Jerry Rasmussen wrote:
Also when I try to dial outbound I get the following errors
channel.c:1920 ast_request: No channel type registered for 'Zap' and
Unable to create channel of type 'Zap' (cause 66). My assumption is I am
getting these errors because Zapata.conf is not being parsed
Or you have
Paul Crick wrote:
So this is doable in the U.S?
That's what I said wasn't it? ;-)
Provided your PRI is set up correctly, you should have a one way speech
patch from called party to calling party upon issuance of an ACCEPT. I
believe it was done this way to allow PBXs to generate ring back, busy etc
Lane wrote:
Hi,
Is it possible with asterisk to deliver a dialtone to a software phone, such
as kphone?
I'm able to dial, but the silence seems to confuse my users :)
SIP phones provide their own local dialtone. If you can get the SIP
phone to call a predefined extension when it goes offhook yo
You also want to add canreinvite=no and type=friend to both entries.
Once you get it working you can try removing canreinvite=no. In [user2]
you want usermane=2 of course.
David Liu wrote:
Hi Vincent,
This shouldn't be difficult. Try the following:
in sip.conf
[user1]
username=user1
secret=pas
Warren Burstein wrote:
I think I've managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways
to do this, nor what the difference is between th
Dorn Hetzel wrote:
On Sat, Dec 25, 2004 at 11:12:22PM -0500, Dorn Hetzel wrote:
I'd like to get VM_CALLERID to include number in addition to name
since often when calls come from cell lines or various other,
the name is just a city, state and the number would be more
usefull. Is there a way to get
Dorn Hetzel wrote:
(a) there are definitely analog DID implementations out there.
not saying they're pretty, but they exist...
(b) are you really sure it's cheaper with only 4 channels to
do a T1? including local loop?
As far as I know, Asterisk/Zaptel does not support analog DID service
Lane wrote:
I just got the new developer TDM11B, but I got some problems with it. Since
Digium is on vacation, I figured I'd ask here first:
I installed the TDM11B, but have not attached any phone lines, yet. I just
want to work with the demo over SIP first.
But here's the story, after instal
Ronald Wiplinger wrote:
I want to record new sound files in different languages, but I need the
text files of the English ones, which I would use as basic.
Since some languages already exists, I believe such a list should be
exist, but where?
See sounds.txt in the Asterisk source code directory.
Venu V wrote:
I am a newbie to asterisk pbx. I got a dialogic card with 2 ports. Can
any one tell whether asterisk supports dialogic cards?
Update me if you have info about the drivers installation and support.
This has been discussed over and over and over again on this mailing list.
See http://ww
Brian Wilkins wrote:
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them
Nabeel Jafferali wrote:
I had the exact same problem, but was in the process of trying to figure it
out myself. I did remove the Asterisk source directory before downloading
the stable version. How do I "remove the modules that are in CVS-HEAD"?
As long as you are only using the stock modules that
Tim Lewis wrote:
Just upgraded to the current stable ver. when I start asterisk with
-vcg I get the following error
[pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258
ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined
symbol: pbx_substitute_variables_varshead
Dec
William Betts wrote:
found the answer in causes.h, but i'd really like to know what this means
Dec 23 17:11:43 WARNING[4845]: channel.c:1921 ast_request: No channel
type regis tered for 'IAX'
IAX2 is the default IAX protocol now.
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Damon Estep wrote:
Has anyone had success with the TDM400 in production? I have multiple
boxes where these cards lock up and the only thing that will fix them is
to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not
matter if it is a FXS/FXO module.
I know this topic has been disc
Fabian Stelzer wrote:
i don't think there are channel driver's for dialogic cards yet...
On Thu, 23 Dec 2004 09:49:54 +0200, Tasos Daskalopoulos <[EMAIL PROTECTED]>
wrote:
Hi there
I have a Dialogic VFX/40 ESC plus installed on Redhat Linux 8.0
and looking for Channel drivers for this Card.
whe
Russell Horn wrote:
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn't mind it hanging up the IAX2 channel and then calling it
again, but it seems that everytime the new call via Zap/2 means no
other calls are possible.
There is ISDN in t
Norman Zhang wrote:
Hi,
I can't receive/send calls with Asterisk. Could someone please give me a
few pointers on my configuration?
Regards,
Norman Zhang
; sip.conf
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=x.x.x.x
localnet=192.168.22.0
mask=255.255.255.0
For one thing
Rich Adamson wrote:
That's an answer to the wrong question. See example below.
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
[voiptalk.org]
;forwards any calls starting with an "8" thru voiptalk.org
exten => _8.,1,Answer
exten => _8.,3,Se
adria vidal wrote:
El 22/12/2004, a las 1:51, Eric Wieling aka ManxPower escribió:
No. Hint is not supported in 1.0.x. Only in CVS-HEAD developement
version of Asterisk.
--Eric
running fine for my in 1.0.3 release and snom 190
adrià
I sit corrected
Alex Brecher wrote:
I still don't get why we don't move over to a web based forum ? I can set
one up on a dual athlon server with 4Gb of memory if you guys are interested
Because then all the client side filtering that many people do to reduce
the amount on messages they have to read would not wor
Peer Oliver Schmidt wrote:
Thanks. I tried it, but no success. Do you know if the hint extension
does work with 1.0.2 stable?
No. Hint is not supported in 1.0.x. Only in CVS-HEAD developement
version of Asterisk.
--Eric
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Greg - Cirelle Enterprises wrote:
from the asterisk messages log:
Registration from '' failed for '192.168.70.25'
the only place I can see extension 40852 linked to the ip is in the
phone's configuration.
pedantic=yes
Take out pedantic=yes and see if it makes any difference.
__
Philipp von Klitzing wrote:
How about this variable? :-)
${SIP_CODEC}: Used to set the SIP codec for a call
That only works for calls going OUT from Asterisk. It does nothing for
incoming calls. By the time the dialplan is called the codec is already
set.
___
Matthew Boehm wrote:
rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client
if possible
All of the calls I am recieving are comming in from PRI. And I am sending
them all out via SIP as 711 to our VoIP carrier/terminator.
If the client is PRI, how can I turn off RFC3389?
rfc3389
Remco Barende wrote:
On Mon, 20 Dec 2004, Eric Wieling aka ManxPower wrote:
Remco Barende wrote:
Hi list!
Just wondering, why is * sticking with an mpg123 version from the
stoneage?
Gentoo comes with 0.59s-r8 and this version doesn't even start.
Ik know I could forcibly unmerge mpg12
Remco Barende wrote:
Hi list!
Just wondering, why is * sticking with an mpg123 version from the stoneage?
Gentoo comes with 0.59s-r8 and this version doesn't even start.
Ik know I could forcibly unmerge mpg123 and install the old version but
I guess some day newer versions will have to be supporte
Matt Schulte wrote:
Now that I finally have someones attention :) I can explain the rest.
The problem is I'm sending calls out through SER (with digest of
course). So in my sip.conf I have:
[sipfarm]
insecure=very ; Because SER inbound doesn't know how to auth
host=blah.blah.net
type=peer
context=i
Matt Schulte wrote:
So what's the work around? Have faxes come from a diff IP?
Well have them come into a different user/friend at least. The IP can
be the same if you are authenticating on username/secret rather than IP.
___
Asterisk-Users mailing l
Matt Schulte wrote:
I asked this question once before with no answer. Hopefully someone can
help me as I cannot see a way to do this. I am wanting to differentiate
inbound calls voice from FAX. The purpose of course voice gets g729 and
FAX gets 711 (ulaw). The problem I'm having is everytime it mat
Wilson Pickett wrote:
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch
between the phone and Asterisk. For most phones you want to use RFC2833
for both the phone and for the entry for that phone in sip.conf.
Yep, and the BT will only work right with certain codecs. I think it's
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
hi
any chance of making asterisk support these?
http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-3835624908
8.htm
According to the manufacturer, they already do:
http://www.ipvolution.com/
NOTE: I have very few FACTS to back this informa
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch
between the phon
Matthew Boehm wrote:
Hey gang,
Getting ready to run some test bills for customers. Most SIP phones have
both an extension and a DID. If a person calls a DID asterisk redirects the
call to the right extension:
exten => 8005551212,1,Goto(companyA-internal,3022,1)
The problem is, that if someone call
mohammad wrote:
I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail
and Call Transfer.
How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/
This is what I call one of the "dirty little secrets of SIP". On SIP
phones (and H323) all the call control
Nabeel Jafferali wrote:
This is somewhat related to my other query on the list regarding NAT
traversal.
I have heard many times that IAX is "NAT-transperant". I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port
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