Very nice work.
I wonder if mark will include that into the main
cvs tree. There're many patches till now that're
interesting (like that, or pauline i4l patches),
but the risk is to have a lot of confusion.
Mark, will you merge that ;-) ?
(i'm thinking to put up a little website,
with a reposito
Hello everyone,
I would like to have soneone who knows how to use SQL method to update
asterisk's conf files, without disrupting the ongoing calls.
Can someone give me some sample about using SQL to update conf files.
Thanks
George Lin
___
Asterisk-
I am using an ATA-186 connecting to an asterisk SIP gateway. When I
dial out through it (via a PRI) to a real number, I notice that I hear
a fake ringback tone. For example, if I call my voicemail, which
answers without a ring, I still hear a bit of ringback when I call via
SIP.
In fact, if I ca
Hello,
* from cvs today, Wed Mar 12 about12:00, don't add SDP 'a' parameter in SIP
messages. Any calls between SIP-devices via asterisk.
SDP body:
v=0
o=root 27982 27982 IN IP4 192.168.50.8
s=session
c=IN IP4 192.168.50.8
t=0 0
m=audio 59430 RTP/AVP
And log from cisco AS5300
Mar 12 13:42:39
Which version of pwlib and openh323 libraries should I have to successfully
compile
chan_h323?
AND not get seg faults?
Best regards,
Krzysztof Bujak
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 10, 2003 8:45 PM
Subject: Re: [
pwlib v1_4_10
openh323v1_11_6
Am Mit, 2003-03-12 um 14.19 schrieb Krzysztof Bujak:
> Which version of pwlib and openh323 libraries should I have to successfully
> compile
> chan_h323?
> AND not get seg faults?
--
___
Jens-Erik Hansen
Global Informatio
I wondered about using streaming content from the internet .. some is available for
free and some site allow custom music lists.
I thought it would be cool to access audio from elsewhere in the world and found
that Reuters (the news source for a lot of media stations) provided streaming audio.
Th
On Tue, 2003-03-11 at 23:02, Jim Archer wrote:
> Hi all...
>
> I have been shopping around and noticed that licensed music on hold music
> can be a bit expensive if you want to assemble a variety of types. Does
> anyone know of an inexpensive source?
There are a handful of sites collecting mus
On Wed, 2003-03-12 at 06:48, Brian Johnson wrote:
> I wondered about using streaming content from the internet .. some is available for
> free and some site allow custom music lists.
>
> I thought it would be cool to access audio from elsewhere in the world and found
> that Reuters (the news sourc
Just got this from my TechTV newsletter:
PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE
You don't have to download illegally to get great music on the Web.
http://cgi.techtv.com/memberservices/newsletters?click=18285&release=2560
Karl Putland ([EMAIL PROTECTED]) wrote:
>
>On Wed, 2003-03-12 at
On Wed, 2003-03-12 at 07:09, Brian Johnson wrote:
> Just got this from my TechTV newsletter:
>
> PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE
> You don't have to download illegally to get great music on the Web.
> http://cgi.techtv.com/memberservices/newsletters?click=18285&release=2560
Free t
> Mark, will you merge that ;-) ?
>
> (i'm thinking to put up a little website,
> with a repository of all * patches. tell me
> if anyone is interested into that)
Usually, on a patch of this size, it takes time to merge because I have to
carefully study and understand all the changes that are bein
> Who is generating this ringback? The ATA or asterisk? What if I call
> a non-suping number with a "the number has been changed" recording?
> Will I never hear it because audio will never be cut through without
> answer supervision?
Find out by doing a trace. If you're using callprogress, the
Good to know that distinction
William X Walsh ([EMAIL PROTECTED]) wrote:
>
>On Wed, 2003-03-12 at 07:09, Brian Johnson wrote:
>> Just got this from my TechTV newsletter:
>>
>> PATRICK'S PICKS FOR FREE AND LEGAL MUSIC ONLINE
>> You don't have to download illegally to get great music on the Web.
>>
Hi Lubo,
I appreciate your email to help with this issue, but I don't understand
your message. I assume your comment about format=slinear is to use
format=slinear in phone.conf instead of format=ulaw. If so, how does
this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
Gregg
Hi,
At 10:30 12-3-2003 +0100, you wrote:
(i'm thinking to put up a little website,
with a repository of all * patches. tell me
if anyone is interested into that)
Yes, this is -very- usefull.
How about a cvs repository :-)
Best regards,
Florian
___
Aste
Title: Message
When dialing a port
on my ATA-186 I get:
== Spawn extension
(default, s, 1) exited non-zero on 'SIP/ata1-1-0c77' --
Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
stack -- Executing Dial("SIP/ata1-2-4fc0",
"SIP/ata1-1|30") in new stack -- Cal
hi alaw (and now also ulaw) folks,
version 0.1.0 is out now. thanks to Bicster for beating
eicon capi to support ulaw and echo cancellation!!
http://www.junghanns.net/chan_capi.html
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
Hi Gregg,
I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
mount without any problems. The quality is perfect and everything is OK
(only some little problems sometime).
But today morning, with the NEW CVS version update of asterisk I found
that SIP(G723/ulaw) and iconnec
On Wed, Mar 12, 2003 at 05:35:44PM +0100, Michiel Betel wrote:
* from today cvs ?
May be same problem about my posting.
Asterisk don't send 'a' parameters SDP body SIP message to called SIP device.
I got same error (488) when using today cvs version.
Try cvs update -D '1 day ago'
On my * box it
Hello all,
I'm using iconnect with LineJACK/PhoneJACK/PhoneCARD and G723.1 codec
from about 1 mount without any problems. The quality is ok and
everything is OK (only some little problems sometime ... when the format
in phone.conf isn't slinear, but format=g723.1 I have only ONE way audio
(the
This issue should be fixed now.
Mark
On Wed, 12 Mar 2003, Michiel Betel wrote:
> When dialing a port on my ATA-186 I get:
>
> == Spawn extension (default, s, 1) exited non-zero on 'SIP/ata1-1-0c77'
> -- Executing Macro("SIP/ata1-2-4fc0", "stdexten|6200|SIP/ata1-1") in new
> stack
> -- Ex
Please check latest CVS. This issue has been fixed and was related to the
dynamic payload merger.
Mark
On Wed, 12 Mar 2003, Lubomir Christov wrote:
> Hi Gregg,
>
> I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
> mount without any problems. The quality is perfect and e
Hello,
I am having a problem with Asterisk that I just cannot get fixed...
When I call in to the main number I have to wait until well into the
second message shown in the extensions.conf snippet below to enter an
extension number. If I enter digits really slowly sometimes it will
work during
Playback is not interuptable, use Background.
On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote:
> Hello,
>
> I am having a problem with Asterisk that I just cannot get fixed...
> When I call in to the main number I have to wait until well into the
> second message shown in the extensions.con
I still haven't been able to get fax detection going,
but I came across something: when I execute "zap show
channel 47" one of the parameters shown is "Fax
Handled: no". I assume that's a reflection of
something in zapata.conf, but I don't find anything
there. Should it read "...yes" in order fo
Florian Overkamp wrote:
> Hi,
>
> At 10:30 12-3-2003 +0100, you wrote:
>>(i'm thinking to put up a little website,
>>with a repository of all * patches. tell me
>>if anyone is interested into that)
>
> Yes, this is -very- usefull.
>
> How about a cvs repository :-)
it would also be nice that a si
A bug that I've been meaning to report...
When you call someone and have the remote person put
you on hold (both are Cisco 7960/SIP recipients), when
they come back off of hold they can hear me, but I
cannot hear them... sounds like one of the audio
channels is not restored properly..
I'll be h
I am using background, the pbx-invalid stuff should (if DTMF
recognition is working correctly) not get played.
On Wednesday, March 12, 2003, at 01:30 PM, Steven Critchfield wrote:
Playback is not interuptable, use Background.
On Wed, 2003-03-12 at 12:19, Brian J. Schrock wrote:
Hello,
I am hav
I had updated CVS this morning and it broke me being able
to call the voicemail extension from my SIP/Cisco 7960 phone
it won't receive DTMF digits...
Restored back to Mar 10 2003 and it worked just fine...
___
Asterisk-Users mailing list
[EMAIL PROTEC
This is two SIP/7960 phones... when one is put on hold, the
music on hold doesn't come to the other. I have the official
mpg123 code installed (not the mpg321...). I do have the zaptel
driver installed since I have a Wildcard FXO card in there
for PSTN access...
zapata.conf:
musiconhold=
A fake web Cvs repository is already on,
checkout http://vmail.espia.it/horde/chora
Unfortunately, due to the limitations of cvs,
there's only 1 revision, that's 1.1 , and only
1 author, that's the user I upload as (matteo).
I update it on a daily basis (07:00 am Rome Time).
I use it to checkout i
Fax handled will be set to yes when zaptel detects the fax CNG tones on
the line. At that point it will try to switch the call to the fax
extension
James
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
> I still haven't been able to get fax detection going,
> but I came across something: when I
On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote:
> I am using background, the pbx-invalid stuff should (if DTMF
> recognition is working correctly) not get played.
>
My users here have complained about similar problems. We've noticed it
most often on outside callers with cell phones, but one
>I still haven't been able to get fax detection going,
>but I came across something: when I execute "zap show
>channel 47" one of the parameters shown is "Fax
>Handled: no". I assume that's a reflection of
>something in zapata.conf, but I don't find anything
>there. Should it read "...yes" in
It's dynamically changed to "Yes" when the fax gets detected on this
channel.
regards
Martin
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
> I still haven't been able to get fax detection going,
> but I came across something: when I execute "zap show
> channel 47" one of the parameters shown is
I know there was discussion at one point of signal loss through multiple
d/a and a/d conversions
what speed does it connect at through the extension ? (assuming analog line in)
Tim - can you show us your config as an example ?
At 11:29 AM 3/12/2003 -0800, you wrote:
>I still haven't been able
You may try to add
relaxdtmf=yes
just before channel => 4 in zapata.conf
regards
Martin
On Wed, 12 Mar 2003, Brian J. Schrock wrote:
> I am using background, the pbx-invalid stuff should (if DTMF
> recognition is working correctly) not get played.
>
> On Wednesday, March 12, 2003, at 01:30 PM,
Same for me , when I call from one sip fxs gw phone to the snom one.
I can hear audio only on from the sip gw and not from the snom.
Thery're using only alaw/ulaw (only accepted in sip.conf).
Yesterday all was working.
>From sip to zap or viceversa is all ok.
Need a debug?
Matteo
> -Messag
how does your cisco send DTMF?
Mark
On Wed, 12 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
> I had updated CVS this morning and it broke me being able
> to call the voicemail extension from my SIP/Cisco 7960 phone
> it won't receive DTMF digits...
>
> Restored back to Mar 10 2003
there is a relaxed dtmf mode that may help.
Mark
On 12 Mar 2003, James Hines wrote:
> On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote:
> > I am using background, the pbx-invalid stuff should (if DTMF
> > recognition is working correctly) not get played.
> >
>
> My users here have complained
I have a fork for this thread. With our asterisk system we have to enter my
touchpad keys a little more carefully than on other systems. Is there
parameters to adjust this behavior.
Brian, this is probably your problem...
>> exten => s,1,Answer
>> exten => s,2,Wait,1
>> exten => s,3,DigitTimeout,
Exactly! Every test I have done has been with a cell phone! I assume
everyone is still perplexed by this?
On Wednesday, March 12, 2003, at 02:12 PM, James Hines wrote:
On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote:
I am using background, the pbx-invalid stuff should (if DTMF
recognition is
I tried that, same thing. Cell phones horrible, but landlines work fine.
On Wednesday, March 12, 2003, at 02:59 PM, Mark Spencer wrote:
there is a relaxed dtmf mode that may help.
Mark
On 12 Mar 2003, James Hines wrote:
On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote:
I am using background
On 2003-03-12 at 12:43, you wrote:
> I had updated CVS this morning and it broke me being able
> to call the voicemail extension from my SIP/Cisco 7960 phone
> it won't receive DTMF digits...
I noticed this last night and found I could fix it by adding
dtmfmode=rfc2833
into each extension defi
On Wed, 2003-03-12 at 14:32, Martin Pycko wrote:
> You may try to add
> relaxdtmf=yes
> just before channel => 4 in zapata.conf
Thanks! This has solved the problem for the test phone! I will try my
cell phone from home tonight, but I suspect the problem has been solved.
Just out of curiosity, is
rxgain and txgain are in db.
We have a similar problem which is even more noticeable since we divert
calls by receiving on one fxo card #1 and sending out on fxo card #2. I
can't seem to find a properly working solution for the attentuation which is
taking place.
For your issue, try setting txgai
I cranked them up around 15 and now the voice levels appear to match the
levels for the automatic voice (like the voice mail and the directory).
Doing that seems to distort the caller id so Asterisk can't decode it. I'm
still experimenting. Thanks!
--On Wednesday, March 12, 2003 8:55 PM +
>Tim - can you show us your config as an example ?
[inBound]
exten => s,1,Answer
exten => s,2,setmusiconhold,default
exten => s,3,DigitTimeout,5
exten => s,4,responsetimeout,20
exten => s,5,BackGround,officemenu ;/var/lib/asterisk/sounds
;... other exts
;Fax Test
exten => fax,1,Dial,Zap/6
--
I doubt it's a signal loss problem. It's a simple
circuit, connected through our local Meridian, so it's
fax-to-Meridian [A]
Meridian internal [D]
Meridian to Channel Bank [A]
Channel Bank to Asterisk [D]
And I know that the fax works okay; it's a production
unit that gets used every day.
Finally, I have NATted ATA-186s working with Asterisk (thanks to
all who made this happen)! My final troubles were with the firmware
version in the 186 -- if you have troubles with this (as I did), make
sure you have the newest firmware in the 186.. Otherwise it just won't
work.
Now for
>And I know that the fax works okay; it's a production
>unit that gets used every day.
>
>As for the conf's, I've tried several things. Here's
>my best guess so far:
>
>from zapata.conf:
>
> context => incomingfax
> channel => 47
>
>and from extensions.conf:
>
> [incomingfax]
> exten => s,1,An
Are you running latest CVS? I think we addressed this issue a few days
ago with the hit='f' thing.
Mark
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
> I doubt it's a signal loss problem. It's a simple
> circuit, connected through our local Meridian, so it's
> fax-to-Meridian [A]
> Meridian
On Wednesday 12 March 2003 21:25, Jim Gottlieb wrote:
> On 2003-03-12 at 12:43, you wrote:
> > I had updated CVS this morning and it broke me being able
> > to call the voicemail extension from my SIP/Cisco 7960 phone
> > it won't receive DTMF digits...
>
> I noticed this last night and found I co
Brendan,
Can we catch up tommorrow before 3pm or around middle of the day on Tuesday
next?
Peter
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Probably you should do dtmfmode=inband in the general section.
Mark
On 12 Mar 2003, Matthew Farley wrote:
> Finally, I have NATted ATA-186s working with Asterisk (thanks to
> all who made this happen)! My final troubles were with the firmware
> version in the 186 -- if you have troubles wit
> with latest CVS rfc2833 DTMF are not detected even when explicitly stated in
> sip.conf.
Need to do some debugging. This could be yet another side effect of the
merger of Ross's code. From now on big contributed patches will need to
have OK's from at least 3 people on the list who are testing
Hi All...
I read the discussion from last December about the various options for
detecting distinctive ringing, but I could not find what the final decision
was.
Could someone please point me at an example of how to configure Asterisk to
recognize distinctive ring?
Thanks!
Jim
_
Hi All...
Is it possible to change the ringing pattern of an extension? I would like
to make my extension ring differently based upon what number (distinctive
ring or different channel) I am being called from.
The best use of this I think would be to let internal calls ring
differently than e
I just posted something earlier regarding music on hold not working.
I've got to believe that it is something I have done or am not doing but
have yet to figure out what.
Anyway have you checked your message and debug logs?
--
Walt Davis
www.waltdavis.net
> This is two SIP/7960 phones... when
version is 'Asterisk CVS-03/11/03-09:57:33'
we can regist to wcom in two ways.
first.
register => masakazu:[EMAIL PROTECTED]
* send REGISTER, but no response from wcom.
second. quit * and change the way with number. like this.
register => 9706052:[EMAIL PROTECTED]
and REGISTER again.
in thi
look at the 'r' option for the Dial command. Example exten =>
333,1,Dial(Zap/1r2)
--Mike
On Wednesday, March 12, 2003, at 04:03 PM, Jim Archer wrote:
Hi All...
Is it possible to change the ringing pattern of an extension? I would
like to make my extension ring differently based upon what n
Hi folks,
I am looking for a way to actually have a live feed for a music on hold
channel.
Basically this is for a local radio station feed so people can ring to
get the lastest report... Sort of like the US vhf weather channels
ANy ideas please ??
Gary
.
Anyone know if it is possible to load your own XML scripts on to the
phone, bypassing the Cisco CallManager? I am still waiting for my
phone to arrive, but I have been playing with Cisco's phone services
emulator, and that doesn't seem to like anything I pass to it.
If it is possible, anyone w
On Wed, 2003-03-12 at 18:35, Gary wrote:
> Hi folks,
>
> I am looking for a way to actually have a live feed for a music on hold
> channel.
>
> Basically this is for a local radio station feed so people can ring to
> get the lastest report... Sort of like the US vhf weather channels
>
> ANy
On Wed, 12 Mar 2003 19:25:16 -0600, Steven Critchfield wrote:
>On Wed, 2003-03-12 at 18:35, Gary wrote:
>> Hi folks,
>>
>> I am looking for a way to actually have a live feed for a music on hold
>> channel.
>>
>> Basically this is for a local radio station feed so people can ring to
>> get the l
Perhaps a clue ... with an established call, like:
snom200<>*<>t400(daggressive_ecan_enabled)<>zhone<>speakerphone
when the snom numberpad key is pressed the dtmf tone is not continuously
robust but the volume is attenuated after the first ~1/10 sec for a moment
and sort of resembles a double
Hello
The option relaxdtmf=yes improved the accuracy of received DTMF digits
on cell phones but did not make DTMF detection 100%. Increasing the
rxgain to 15 does improve the accuracy almost to 100%, as long as you
let the first few milliseconds of the recording play. The problem now
is that o
Hi All...
We have found that the caller ID information presented to some one we call
from Asterisk using iconenct is not predictable. The caller ID will be
unavailable or else deltathree. I have in sip.conf:
[iconnect]
type=friend
username=41306756
password=2264
host=natrelay.deltathree.com
c
Two issues-- is anyone using Asterisk as a gatekeeper with
cisco 7960 phones and cisco gateways? Experiences, thoughts,
etc appreciated. If anyone has moved from/to ser to/from
Asterisk, I would be interested in hearing experiences...
We have been trying to get message waiting indication workin
That is basically what I see as well. I do not see any response coming back
with the SIP debug. Could it be a problem with the first header line
XXX Need to handle Retransmitting XXX:
(or is that something generated by asterisk).
Just a thought. I am not a protocol expert so I am just curious.
What mode are you running the Phone in? SIP, MCGP, or SCCP (Skinny)
You mentioned Call Manager so I will assume SCCP. If that is the case I
do not know.
However if you are running it in SIP, All you have to do is set
# XML URLs
services_url: ""; URL for external Phone Services
d
> it does need to service multiple lines at the same time.
>
> Also as a background music (on loudspeaker type phones) would be nice.
>
> We are currently playing with festival, so the caller can select which
> area they want details for and we hourly download and massage the
> hourly updates avail
The friend would only happen if the "From: " was iconnect. Unfortuantely
SIP does not differentiate a user from Caller*ID. The only way to make the
peer match would be if we matched the peer based on IP address.
Mark
On Wed, 12 Mar 2003, Jim Archer wrote:
> Hi All...
>
> We have found that the
Try this. Make a "pipe" called "live.mp3" then use the normal
Asterisk music on hold function for play the pipe.
Next you will need a very simple copy type script to read
dev/audio filtr it through an MP3 encoder and write it to the
pipe.
Somehow you fork/exec the script just before connecting
In principle you could use a named pipe in the filesystem right? Then you
don't have to fork/exec anything (man mkfifo)
Mark
On Wed, 12 Mar 2003, Chris Albertson wrote:
>
> Try this. Make a "pipe" called "live.mp3" then use the normal
> Asterisk music on hold function for play the pipe.
>
> Ne
> Is anyone using MWI on 7960's when Asterisk is ONLY being used
> for voicemail, and not for a gatekeeper?
>
> If anyone has MWI working successfully with 7960's, would it
> be possible to get a dump of a successful NOTIFY message that
> turns a light on/off?
Just put "mailbox=1234" in your sip f
It used to work just fine and I can only think of three possibilities:
1) Moved asterisk onto a different server and that server does not have a
sound card?
2) Rebuilt and hardened asterisk on a Mandrake 9.0 distro, so perhaps I
left out some modules or something.
3) Upgraded from the developer
working examples would be appreciated ;-)
On Wed, 12 Mar 2003 23:32:05 -0600 (CST), Mark Spencer wrote:
>In principle you could use a named pipe in the filesystem right? Then you
>don't have to fork/exec anything (man mkfifo)
>
>Mark
>
>On Wed, 12 Mar 2003, Chris Albertson wrote:
>
>>
>> Try t
Did you remember to move to mpg123 and not use mpg321 which is often
aliased to mpg123?
Mark
On Wed, 12 Mar 2003, Walt Davis wrote:
> It used to work just fine and I can only think of three possibilities:
>
> 1) Moved asterisk onto a different server and that server does not have a
> sound card?
On 2003-03-12 at 09:44, you wrote:
> > Who is generating this ringback? The ATA or asterisk?
> Find out by doing a trace. If you're using callprogress, then you should
> see a 180 Ringing sent to the ATA when we detect ringing on the FXO. If
> you're not using call progress, then we should no
Hi,every one!
I would like to know if Asterisk
could be used to build a IVR with complex flow? To provide a complex sample
would be appreciated.
Regards.
john
Will mark merge this into CVS or do I have to apply this to my own code
checked out of CVS? Also, which version of CVS code do I patch this
against?
On Tuesday, March 11, 2003, at 10:00 PM, Fettahlioglu, Mahmut wrote:
Hello everyone,
I was working on implementing several changes to asterisk
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