I've just upgraded to the latest CVS head, and my outbound calls stopped
working. I traced it back to the line
exten = s,9,CallingPres(${ARG2})
It seems as if this application is now missing.
I tracked back the changes and found in 1.415 of chan_zap.c the code was
removed because it was
Wilson Pickett wrote:
Now here's a thread I've been waiting to see! I have had issues with
what is considered to be a decent phone, the siemens DECT line.
Fortunately, the problem is just callerID which although annoying,
isn't mission critical, and we are in Europe. Still, the USA isn't the
Hi, ALL:
I'm almost give up the oh323 too.
I compiled the asterisk-oh323 for several times(ten or more).
In my first time, I got pwlib 1.6.6 from CVS of openh323.org.
But it seems a little buggy,so I failed.
I downloaded Janus's patch version, and followed its steps. It seems
OK when I compile
The reason i am using 1 scenario is because of routing, authentication and accounting. If i can use these things in asterisk i will use it.
1)What is the best way to do that, through extensions ?
2)When the call coming i can check the phone number and if the number should go to pstn i will
Asterisk wrote:
I've just upgraded to the latest CVS head, and my outbound calls
stopped working. I traced it back to the line
exten = s,9,CallingPres(${ARG2})
It seems as if this application is now missing.
-= Info about application 'SetCallerPres' =-
[Synopsis]:
Set CallerID Presentation
Hi,
I had try to set up the call routing for asterisk to
interwork with cisco AS5300 and found thatAsterisk only support codec
g711alaw and g711ulaw. For the other codecs (g723, g729, gsmfr), the
callswere disconnected with cause value 63 (service option not
available) or 127
I actually got rid of my phone service.. no more pots line in my house.
But, I miss my call screen macro. Any way to do this with a SIP
channel? (Obviously the parking isn't the problem, but rather recording
their name). I set it up so they should only have to record their name
once provided
These OPTIONs packets are what we send for qualification, a scheme that
can also be used for NAT keepalives. You turn them on by adding
qualify=yes in the [peer] section of sip.conf
With qualification on, we regurlarly measure the latency between
Asterisk and the client and decide whether the
I didn't read the message because I wasn't an intended recipient :( I
hope it didn't say anything great.. It was about asterisk right?
C F wrote:
I love this message, just the perfect one to get some people on the
list pi**ed off. The subject is *very* descriptive. The disclaimer
just a
g729 is a commercial codec and requires a license to use. You can
purchase a license for use with Asterisk. See
http://www.voip-info.org/wiki-Asterisk+G.729+licensing
g723 is also a commercial codec but Asterisk does not support it other
than in passthru mode.
I have not heard of gsmfr.
What
Many thanks.
Julian.
Trevor Peirce wrote:
Asterisk wrote:
I've just upgraded to the latest CVS head, and my outbound calls
stopped working. I traced it back to the line
exten = s,9,CallingPres(${ARG2})
It seems as if this application is now missing.
-= Info about application 'SetCallerPres' =-
G729 requires a licence from Digium.
Umar
On Thu, 24 Mar 2005 16:37:15 +0800, raymond [EMAIL PROTECTED] wrote:
Hi,
I had try to set up the call routing for asterisk to interwork with cisco
AS5300 and found that Asterisk only support codec g711alaw and g711ulaw.
For the other
I've noticed that my * box isn't logging all that it use to / should.
I'm running version Asterisk CVS-v1-0-03/07/05-22:42:03, prior versions
would log everything including connections to voicemail and such.
This version of (or more likely my configs) seems only to be logging
certain things.
A few ideas to help the Iaxy little device work better:
at the very first startup (or with a switch), finding the LAN and a
phone connected, it could call it and say it's IP (like 'one' 'nine'
'eight' 'dot' and so on).
Maybe with a little improvment it could also be called from it's
telephone
Ive got the same
problem. MusicOnHold works if I use something like:
Exten =
,1,MusicOnHold()
but if I try to answer a
call and then transfer or put on hold the call, I get no music.
Does anyone have any idea?
Bye,
Gianluca.
Da: Kanishka
Somaratne
On Wed, Mar 23, 2005 at 06:55:28PM +0200, Mark Elkins wrote:
Last time I tried - there were a few problems...
I had a few random crashes, higher delays and echo with the EICON. I
replaced it now with an HFC. The EICON on isdn4linux was however
a bit better than the AVM C4 with CAPI.
1 -
Hi all,
Is * able to do the difference between Fax and voice, and then adapt the
treatment of the call ?
An example ?
Thx
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To
google asterisk fax
On Thu, 2005-03-24 at 11:53, Guy Decarpentrie wrote:
Hi all,
Is * able to do the difference between Fax and voice, and then adapt the
treatment of the call ?
An example ?
Thx
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Yes all ports have been forwarded on the iptables section at top
UDP/5060, UDP/4569, UDP/5036, UDP/1:2, UDP/2727
Doing a simple telnet to these ports non of them are open, even from
inside the LAN, so the issue is on the asterisk box rather than the
forwarding I think.
On Wed, 23 Mar
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how to receive and mail a fax, now i want to know how to detect
if the call is a fax or a voice call, and reroute the call if it's a
voicecall, and mail the fax if it's one.
Thx
On Thu, 2005-03-24 at
Hello!
I'm searching for a way to call ATA (IAX or SIP) that is not registered
with any server or proxy.
Is it possible to make such a call from a softphone to an ATA just with
IP? Something like (sip:// or iax://)[EMAIL PROTECTED] (where
210.12.34.45 is ATA's public ip)?
Regards,
CuPoTKa.
On Thu, 24 Mar 2005, Paul Goodyear wrote:
Yes all ports have been forwarded on the iptables section at top
UDP/5060, UDP/4569, UDP/5036, UDP/1:2, UDP/2727
Doing a simple telnet to these ports non of them are open, even from
inside the LAN, so the issue is on the asterisk box rather
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how to receive and mail a fax, now i want to know how to detect
if the call is a fax or a voice call, and reroute the call if it's a
voicecall, and mail the fax
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how to receive and mail a fax, now i want to know how to detect
if the call is a fax or a voice call, and reroute the call if it's a
voicecall, and mail the
Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit :
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how to receive and mail a fax, now i want to know how to
detect if the call is a fax or a voice
Hi,
I use the windows client SJPhone (http://www.sjlabs.com) to make direct SIP
calls. To call my * box I just enter: sip:[EMAIL PROTECTED]
SIP:10.100.0.201 to call my SIP handset.
Good luck,
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of CuPoTKa
exten,fax,1,Dail(
On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit :
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit :
google asterisk fax
Well, i know how
sorry
exten = fax,1,Dail
On Thu, 2005-03-24 at 12:53, Altus Snyman wrote:
exten,fax,1,Dail(
On Thu, 2005-03-24 at 12:45, Guy Decarpentrie wrote:
Le jeudi 24 Mars 2005 11:22, Forrest W. Christian a écrit :
On Thu, 24 Mar 2005, Guy Decarpentrie wrote:
Le jeudi 24 Mars
I have had a 7905G lying around for the past 8 months and have just had
time to start setting it up with asterisk. I started following the
links on the wiki about where to download the firmware from cisco. I
signed up to cisco so I could apparently download the new firmware for
the phone, but
ah, you learn something every day :)
I will have a look at the asterisk conf files next week, thank for the info.
Paul.
On Thu, 24 Mar 2005 11:13:57 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Thu, 24 Mar 2005, Paul Goodyear wrote:
Yes all ports have been forwarded on the
Hi,
I'd like to setup oneway connection - so asteriskB can place calls on
asteriskA and be safely authenticated with rsa keys. I just don't get any
response on asteriskA.
I've generated pair of keys: name.key, name.pub and put them on both servers
- is it right to only have name.key on
On Thursday 24 March 2005 11:10, Greg wrote:
I have had a 7905G lying around for the past 8 months and have just had
time to start setting it up with asterisk. I started following the
links on the wiki about where to download the firmware from cisco. I
signed up to cisco so I could apparently
On Thursday 24 March 2005 11:28, you wrote:
how do i do that?
If you want to steal it, ask on the list. I'll not help you.
On 24/03/2005, at 9:19 PM, Bob Goddard wrote:
On Thursday 24 March 2005 11:10, Greg wrote:
I have had a 7905G lying around for the past 8 months and have just
had
Hello all
I have come to Asterisk with no previous telco experience.
As I will be playing with Asterisk really soon, I would like to have
some pointers as to some tutorials in telco that could help me get into
all this. I am quite a beginner, don't forget :)
Thanks a lot!
Best,
fred
I am new like yourself Fred. Good luck!
On Thu, 24 Mar 2005 12:45:58 +0100, Fred Blaise [EMAIL PROTECTED] wrote:
Hello all
I have come to Asterisk with no previous telco experience.
As I will be playing with Asterisk really soon, I would like to have
some pointers as to some tutorials in
I just found that you need a support contract that will set me back
AU$19. I will buy it after the easter break as the sales offices are
closed. Can anyone send me the sip firmware off list at all so I can
get the phone working over the weekend?
Regards,
Greg
On 24/03/2005, at 9:19 PM, Bob
The biggest tip anyone can give you is to simply read the wiki and
don't give up if it doesn't work straight away
I started out by getting someone to write a basic dial plan for me and
then backed it up and modified it... Not a bad starting point but not
everyone is willing to pay to save a
Michael Sanders wrote:
Hi, How do I connect two Analog PBX together with Asterisk.I want two simultaneous voice channels between sites.
My (simple) solution was to put an * server at each site with an extension from the PBX connected to an * FXO port. Users dial the regular pbx ext to
I haven't experienced this problem with Asterisk but I have with
other products. Those products updated their support for DTMF to
match the state of the RFCs and thereby created an incompatiblity
with existing UAs. It sounds like this may be the case here. I'd look
at both chan_sip.c files and
Hi ,
I have 3 ISDN BRI and 4 analog line .
I would like a smal ofice with 30 exension.
Can you give me it is possibile to work together isdn and analog in a same pc (PBX).
Which isdn and analog card aou recommand ? Is there any support for these card ?
Thaks.
Do you Yahoo!?
Yahoo! Small
Hi all,
Im running Asterisk since two days, and its
really one of the phatest software available on the net!!! Respect!!! I have
connected Asterisk as a call manager for a cisco gatekeeper. Everything works
fine internal, but if I want to ring to a PSTN over another call manager, which
Good day all
I have a snom 220 with the extra keypad
When more than one call comes in none of the extra lines on the phone
lights up or anything.You hear the beep in you ear but no way of picking
it up.I tied 4 different firmware versions.On was a very old one,with
actually worked but is gave echo
I don't see how it could be firewall issues. I have firewall ports 4569
and 5036 open for UDP traffic to and from the Asterisk server.
Yesterday we conducted a conference call that lasted several hours
without a drop, just periodic dead spots for a few seconds. Other
calls disconnect entirely.
I am using TDM FXO (4) with one of my server , in middle east and there
internet not so good, every time its has some packet loss happend. but speed
is good. quite enough for 4 port with ILBC. my problem is i setup the same
thing with same config in several country like singapore, bangladesh
I too have heard of people persuading a vonage tech to provide the
password to log into and factory reset their device, but I get the
impression that it is an uncommon occurrence.. you'd be lucky, basically.
I have an ATA-186 that Vonage unlocked for me. They used to just charge
$20 or so (on
1) When an incoming call to my DID number is initiated, a prompt is played so
that the caller
can enter an extension number or
zero for the operator. However, at least 30%-50% of the time the digits that
are entered from
the touch tone phone is slightly
different from what is received by
Dear Asterisk Users,
if I do a : /usr/sbin/asterisk -r -x restart gracefully , asterisk
just quits without any message. Any idea ?
(debian 3.1 with asterisk packages from unstable :
1.0.7-BRIstuffed-0.2.0-RC7k)
/sebastian
___
Asterisk-Users mailing
Faxes I found are mostly unreliable if the padding is not set correctly on
the analog trunks.
The example below shows how to set the mailbox per timezone. Also what is
your system timezone set for? the tz option sets the zone.
;4200 = 9855,Mark
Spencer,[EMAIL PROTECTED],[EMAIL
Hello All,
This could be more a routing/linux problem, but I'm wondering if can be
handled somehow within *.
I have a client that has a contract with a VoIP provider. This contract
limits number of phones registered using same IP to 3. There are 3 ways to
handle this:
1. Few ATA connected to 3
I have an x100p card is this padding something that can be set? I'm
off to google to see if I can find some fax padding. :)
Henry Devito wrote:
Faxes I found are mostly unreliable if the padding is not set correctly
on the analog trunks.
The example below shows how to set the mailbox per
On March 24, 2005 08:08 am, Rich Adamson wrote:
Then try the following in zapata.conf:
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
as a starting point for each fxo channel.
Does echotraining *improve* echo cancellation at all? All I've ever found it
to do is help the
Dear all
After googling unsuccessfully, a question :
In which circumstances should I be using SER ?
i.e. what does SER give me that * does not ?
Thanx
PaulH
--
Paul Hewlett (Linux #359543) Email:`echo [EMAIL PROTECTED] | rev`
Tel: +27 21 852 8812 Cel: +27 72 719
Paul Hewlett wrote:
i.e. what does SER give me that * does not ?
SER gives you the ability to process thousands of calls per second. Because
asterisk is NOT a SIP Proxy (which SER is) Asterisk cannot process thousands
of calls per second.
-Matthew
Well this is true.. how reliable is that though? I know even with
dialup we SOMETIMES will miss a call accounting packet because they
are sent UDP
On Thu, 24 Mar 2005 00:17:31 -0500 (EST), Greg Boehnlein [EMAIL PROTECTED]
wrote:
On Tue, 22 Mar 2005, Matt wrote:
Hi,
The reason I
First, ensure the TDM card has been defined to match the telephony
standards for each country where the card is installed.
Bad echo on these outside North America is likely to be because of
impedance mismatch. Echo cancel isn't likely to properly fix it.
I was surprised at the lack of info
All, I get this whenever trying to dial to a peer when the peer
registered to another server. I'm basically trying to use realtime to
check for the peer and dial it.
Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial(SIP/brak-f69f,
IAX2/brak-test/107) in new stack
Mar 24 09:16:47 DEBUG[4527]:
Hullo :) Can someone help me untangle a bit of a mess?
I'm trying to set up a demo * server to show off how useful it can be to our
business (as an IVR system and VoIP backup if our ISDN30s fail). I've not
been able to get NT mode working with our InterTel Axxess PBX, so I've
resorted to using
1) When an incoming call to my DID number is initiated, a prompt is
played so that the caller
can enter an extension number or
zero for the operator. However, at least 30%-50% of the time the
digits that are entered from
the touch tone phone is slightly
different from what is received by
It is in the zapata.conf file RXGAIN and TXGAIN this effects all calls voice
and fax.
- Original Message -
From: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 24, 2005 7:58 AM
Subject:
Indeed.. there is no $40 cancellation fee unless you fail to return
their ATA.. then they charge you and it's yours... what you think
those devices are free?
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Steven Critchfield wrote:
On Wed, 2005-03-23 at 17:50 -0800, Sean Kennedy wrote:
I am trying to compile spandsp on my asterisk server, and it keeps
failing out with the following
t4.c:38:21: tiffiop.h: No such file or directory
In file included from t4.c:41:
spandsp/t4.h:62: error: syntax
As does mine. I have multiple streamsperson is on holdstreams.
They hang upnever drops the connection. Never lost music. Cept for
once when the stream server went down.
-Ronan
On Mon, 2005-03-21 at 15:24 -0500, Matt wrote:
I'm not having this issue... YES... it DOES stop when
hello list;
When the call manager opens two channels with the 'originate' cmd, is
there a way to bridge them together later? Is there a command like
'Action: bridge' with the two channels in parameters, in the manager, or
elsewhere?
I couldn't manage to find it on the wiki. Should I use the
hi all, was wondering if someone could assist with a slight problem
i'm having. I have asterisk setup with extensions 101 to 109 and am
using xlite, grandstream budgetone, polycom ip500 and a couple of
other phones. the problem is:
1. only the xlite extension (107) can receive calls.
2. all
Matt Schulte wrote:
Mar 24 09:16:47 NOTICE[4527]: Unable to create channel of type 'IAX2'
(cause 3)
Doesn't seem like a RealTime issue to me. Did you add that user into the
flatfile and test that way?
-Matthew
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On Thursday 24 March 2005 16:14, Matthew Boehm wrote:
Paul Hewlett wrote:
i.e. what does SER give me that * does not ?
SER gives you the ability to process thousands of calls per second.
Because asterisk is NOT a SIP Proxy (which SER is) Asterisk cannot process
thousands of calls
Paul Hewlett wrote:
Thanks Matthew. From this I would infer that SER is to Asterisk as
Squid is to HTTP ?
Thats actually not a bad analogy. I'm terrible with analogies. But yes, SER
is a pure SIP proxy server; meaning it knows nothing about the audio portion
of a call. SER can recieve a call,
Dana Olson wrote:
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote:
6) Configuration requires Linux, as opposed to a web browser or
something more standard.
I compiled iaxprov on Cygwin, works nicely. There's somebody on this
list that made a Windows version to
On Thu, 24 Mar 2005 08:09:19 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Dana Olson wrote:
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote:
6) Configuration requires Linux, as opposed to a web browser or
something more standard.
I compiled iaxprov on
I currently have the following outbound-local config in my setup
I can call SOME of the numbers (like 337, and 998, and
323).. but when I try to dial say like 601 I get a 404.. any
thoughts, I can't see any difference in the config.
Also, I seem to be able to dial any number
Hello!
I am in the middle of having a vanity toll-free DID set up. It's been 13
days now (9 business days). This is the first time I'm doing this, and
I'm not sure of the process. There has been a very weird progression of
changes on my number, from fast-busy, to a message saying that I'm
On Thursday 24 March 2005 14:19, Gavin Hamill wrote:
Hullo :) Can someone help me untangle a bit of a mess?
I solved my own problem as per usual - it seems this only happens after I post
to a busy mailing list...
For reference, here's what works for me ... I expect there are much more
elegant
On Wed, Mar 23, 2005 at 13:00:57 -0600,
Bruno Wolff III [EMAIL PROTECTED] wrote:
What I am wondering is if I could use the TDM431B to divide our phone lines
into 3 groups to use as an intercom, but still have them work normally
for answering calls and calling outside.
It looks like I do
How about scanning for it's mac address?
http://ipscan.sf.net/ipscan.exe
--
http://www.umich2.com
Digium doesn't label the MAC address on the device, unless it's such a
fine print that no one can read it. I believe this has been said a few
times in the conversation.
Folks,
Please forgive my ignorance. I think that what I am asking must be so
obvious that no one bothers to write it down. But I don't know the
answer so...
I want to set up * with one incoming VOIP phone number. If someone calls
me and is talking to me on that phone number, how does a second
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What do you get for an output from the CLI? Is the 9 being stripped?
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 24, 2005 9:16 AM
Subject: [Asterisk-Users] Question on
Problem:
Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that
Asterisk can detect and use. It worked in 1.0.5, but has not worked
since. This has been verified on SoundPoint IP 300's and SoundPoint IP
600's.
Workaround:
It used to be that for DTMF to work, I had to set the
On Thu, 2005-03-24 at 10:50 +0100, Marc SCHAEFER wrote:
On Wed, Mar 23, 2005 at 06:55:28PM +0200, Mark Elkins wrote:
Last time I tried - there were a few problems...
I had a few random crashes, higher delays and echo with the EICON. I
replaced it now with an HFC. The EICON on isdn4linux was
Art,
Some VOIP ITSP's (all?) support multiple incoming calls. * picks up the
second call, and sends the caller to voicemail.
Art Zemon wrote:
Folks,
Please forgive my ignorance. I think that what I am asking must be so
obvious that no one bothers to write it down. But I don't know the
answer
Read this!
http://www.automated.it/guidetoasterisk.htm#_Toc49248757
rgds
Rikard
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Art Zemon
Sent: den 24 mars 2005 16:35
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Newbie Voicemail Question
Flatfile meaning iax.conf? Yes..
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 24, 2005 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote:
Mar 24
Try playing with these;
exten = s,n,DigitTimeout(3); Set Digit Timeout to 3 seconds
exten = s,n,ResponseTimeout(20); Set Response Timeout to 20 seconds
Many Regards.
Danny Froberg
On Thursday 24 March 2005 09.24, Joel Jn-Francois wrote:
1) When an incoming call to my DID
I too have heard of people persuading a vonage tech to provide the
password to log into and factory reset their device, but I get the
impression that it is an uncommon occurrence.. you'd be lucky,
basically.
I have an ATA-186 that Vonage unlocked for me. They used to just
charge
$20 or
Hi Noah -
I got everything to load via ftp. The phone appears to correctly boot
from the config files. I also put the latest firmware there and the
phone sucessfully loaded it.
For some reason, the phone and * don't see each other. This is the
part
that confuses me. Any clues as to why the
Hello,
We are using Asterisk-1.0.3 version. Grandstream works
fine with this version. When trying to use a Sipura 2000 we the 488
error and also Answering with non-codec capability 0x1
(telephone-event) message.
also in the sip show peer for Sipura
Codecs : 0x0 (nothing)
Codec
Jesse Guardiani wrote:
Hello,
I've installed speex 1.1.7 and asterisk 1.0.6
from Gentoo's Portage and I'm experiencing
asterisk crashes whenever I try to make a
connection from my X-Lite client under wine
to asterisk using the speex codec.
I know speex is being attempted because SPX
Okay sorry for the empty message bodies earlier problem with Outlook, I
guess is you have digital IDs as I do and they expire, even if you override
the signature option it clears all sent textwindows...ph
Anyway, what I was trying to relay to the group was that ZyXEL has come out
with
Tim Pushor wrote:
Some VOIP ITSP's (all?) support multiple incoming calls. * picks up
the second call, and sends the caller to voicemail.
Tim,
Ah... thank you. No wonder nobody bothered to write about it!
-- Art Z.
--
Art Zemon, President
Hen's Teeth Network http://www.hens-teeth.net/
Hey gang,
I'm trying to setup the ability to dial a SIP user via their email address.
I'm using SJPhone as my tester UA, but most clients will be using XTen Pro.
I added an SRV DNS entry into our zone, and it returns:
; DiG 9.2.1 SRV _sip._udp.cytelcom.com
;; global options: printcmd
;; Got
Because it seems if I dial 9 before the number all of my dial rules
get ignored... but I'd like to avoid the 9 anyway.
On Thu, 24 Mar 2005 11:08:02 -0500, Matt [EMAIL PROTECTED] wrote:
Ok.. apparently if I dial 9601 then it works.. and 9 is set as my
outside line digit... but I seem to be
Hi all,
My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls
over ISDN2.
Now I want to implement receiving incoming faxes into my setup so I did a
google and some reading on the wiki.
I got the spandsp 0.0.2pre10 package compiled and installed, patched
asterisk's apps
On Thu, 24 Mar 2005, Matthew Boehm wrote:
Paul Hewlett wrote:
Thanks Matthew. From this I would infer that SER is to Asterisk as
Squid is to HTTP ?
Thats actually not a bad analogy. I'm terrible with analogies. But yes, SER
is a pure SIP proxy server; meaning it knows nothing about the audio
This is a known issue with livevoip.com service. It's my opinion this
is really a design issue within asterisk, but Mark disagrees.
Your are correct - I do not agree with Mark but, he has never replied to
any emails about this.
The problem is * must answer the incoming iax call from livevoip in
Ok.. apparently if I dial 9601 then it works.. and 9 is set as my
outside line digit... but I seem to be able to dial just the phone
numbers without a 9 (which is how I want it) for all the other
ones
What do I need to do to remove 9 as a dial-before digit for outside line access?
On
On Wed, 23 Mar 2005, Joseph wrote:
My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it
works in version 1.0.5 (was working with 1.0.3).
I'm using SPA-3000 and dtmfmode=inband
I noticed the exact same behavior w/ my upgrade to 1.0.7 using Polycom
SPIP phones w/
On Thu, 24 Mar 2005, David Gomillion wrote:
Problem:
Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that
Asterisk can detect and use. It worked in 1.0.5, but has not worked
since. This has been verified on SoundPoint IP 300's and SoundPoint IP
600's.
Workaround:
warning - shameless plug
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-Kerry
-Original Message-
On Thu, 24 Mar 2005, Matt wrote:
Well this is true.. how reliable is that though? I know even with
dialup we SOMETIMES will miss a call accounting packet because they
are sent UDP
What does it matter? You can't get the information out of just a Start
packet. All you'll know is that a
Then try the following in zapata.conf:
echotraining=800
echocancel=yes
echocancelwhenbridged=yes
as a starting point for each fxo channel.
Does echotraining *improve* echo cancellation at all? All I've ever found it
to do is help the canceller converge faster. i.e. if the echo
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