Hi list..
I am trying to add g723.1 annex A code in the * as per
instructions in the README and other files.
When i try to MAKE * i get the following error.
So i would be thankful, If anyone can correct me.
I am using GCC 3.2 On REDHAT 8.0 and g723.1 Annex A
Code from ITU.
cc -fPIC -c -o
On Sunday 27 July 2003 23:57, Andy Hester wrote:
Tilghman,
I'm not sure how to use this logic. Would this be for something
like, for example, deleting of forwarding a message that a certain
age?
No, this would be used in something like:
central=US/Central|'vm-received' $[${DIFF_DAY}
Mark,
thank you for your information.
I am in the process of recording voicemail prompts in german. How do I
specify the language for the voice mail messages? I want to offer both
language files, based on the calling party.
Use setlanguage. Then organize the language files by directory e.g.
Hi Andreas,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
at my site it is working exactly as you wrote in your 1st example. How is
your PBX setup? I remember that there is
Mark Spencer schrieb:
Use setlanguage. Then organize the language files by directory e.g.
/var/lib/asterisk/sounds/de
/var/lib/asterisk/sounds/digits/de
Also, say.c will have to be modified to support German style number
handling.
Mark
On Sun, 27 Jul 2003, Peer Oliver schmidt wrote:
I am in the
I have set one up on RH9 but I am sure the process is the same.. its not very well
documented I am afraid..
Basically download the CAPI drivers from the AVM site..
Extract them and then run make and then make install..
Make sure you have the ISDN tools loaded..
Create a file in etc called
Its just like any other codec so it should work in SIP, IAX or any other connection..
Hi,
Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs
running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I
have not found any documentation on
Hi all,
is it possible to go on in the current context after the dest channel hung
up?
For example:
exten = 111,1,Dial,Zap/4
If the originating channel is connected to Zap/4 and the destination channel
(Zap/4) hangs up, both channels will be destroyed.
Is there any option or whatever
Reini Urban wrote:
Mark Spencer wrote:
Use setlanguage. Then organize the language files by directory e.g.
/var/lib/asterisk/sounds/de
/var/lib/asterisk/sounds/digits/de
Also, say.c will have to be modified to support German style number
handling.
Mark
On Sun, 27 Jul 2003, Peer Oliver schmidt
Adam why would you need a channel Bank for the E100P?
Regards
Mark McKibbin
DCS Internet
64 Queen St
Warragul
Victoria3820
Australia
www.dcsi.net.au
[EMAIL PROTECTED]
Ph. 1300 665 575 (Help Desk)
Ph. +61 356 241 120 (Direct)
Fx. 1300 556 595
-Original Message-
From: Adam Goryachev
Peer Oliver schmidt wrote:
Hi Andreas,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
at my site it is working exactly as you wrote in your 1st example. How is
your PBX
Morning Peer,
Am Mon, 2003-07-28 um 09.32 schrieb Peer Oliver schmidt:
Peer Oliver schmidt wrote:
Hi Andreas,
I have asterisk behind my primary PBX connected via ISDN (chan_capi).
Calling out and calling in works just fine, however I can't connect to
my primary pbxs' extensions.
I would like to use the CAPI drivers and I have downloaded the drivers
from the AVM site but when I do a make in the fritz directory it exits
with :
make[1]: *** [tools.o] Error 1
make[1]: Leaving directory `/usr/src/fritz/src.drv'
make: *** [drv] Error 2
I guess that this is because I don't
I've not got a sound card in my RH9 * box and music on hold works great as long as you
have mpg123 in /usr/bin
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: 27 July 2003 20:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] can't get musiconhold to
Do you have more than one language defined within your system, i.e.
international calls received the english prompts, whereby german callers
receive german prompts?
Just as some one said,'The translation itself is art', if we translating the
voicemail for Chinese or Japanese, we have to
I think the quality for music playback on my SIP stuff is pretty good. The
real sound problem is in the voicemail access. I very often get sound
dropouts when * is reporting the number of new or old messages.
Iain
--On Saturday, July 26, 2003 10:39 pm -0500 Mark Spencer
[EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Does anyone know how to configure the spans on the E400P for use with ISDN-E
DSS1 (Ericsson FS standard ISDN30)?
- --
Regards,
Tais M. Hansen
ComX
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
Mark Spencer wrote:
Use setlanguage. Then organize the language files by directory e.g.
/var/lib/asterisk/sounds/de
/var/lib/asterisk/sounds/digits/de
Also, say.c will have to be modified to support German style number
handling.
Mark
On Sun, 27 Jul 2003, Peer Oliver schmidt wrote:
Hi,
I
who has the clock and which side is cpe ?
On Monday 28 July 2003 11:49 am, Tais M. Hansen wrote:
Hi,
Does anyone know how to configure the spans on the E400P for use with
ISDN-E DSS1 (Ericsson FS standard ISDN30)?
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 28 July 2003 12:56, Michael Bielicki wrote:
Does anyone know how to configure the spans on the E400P for use with
ISDN-E DSS1 (Ericsson FS standard ISDN30)?
who has the clock and which side is cpe ?
I was looking for span config, since
Hi list,
anyone know what is going on here?
I don't get any sound from the out clip I get the following when I dial in
to asterisk (after which it just times out):
-- Event [0=[06] Ring] on vpb/1-7
-- Executing Wait(vpb/1-7, 2) in new stack
Read_channel ## vpb/1-7: Setting record
I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot
recieve the the calls from the zaptel interface which is a E100P with
pri signaling.
That is something with asterisk becouse rolling back to version from
06/23/03 using the new libpri and zaptel fixes the problem.
Here is
The ISDN tools are in the package isdn4kutils..
Not sure about the error you are getting when you run make do you have the
kernel-source installed?
I would like to use the CAPI drivers and I have downloaded the drivers
from the AVM site but when I do a make in the fritz directory it exits
Kelvin Chua wrote:
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's the device
fairing?
~kelvin
Can someone send me SIP firmwire for audiocodes 104.
I has h.323 only and it sucks
___
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Thanks for your info but I think I have it working at last. Below are
the steps I took which might help others.
1) Download the PCI AVM drivers from ftp://ftp.avm.de/cardware
2) Download the Chan_capi from http://www.junghanns.net/asterisk/
3) tar -xvzf fcpci-suse8.0-03.09.10.tar.gz which creates
Anton Yurchenko wrote:
the thing seems to be in the chan_zap.c doing a diff from the version
that I have working I see this change:
-
@@ -5607,10 +5609,10 @@
strcpy(pri-pvt[chan]-callerid,
);
John,
See my long-winded commentary:
http://lists.digium.com/pipermail/asterisk-users/2003-April/009797.html
Method 2 is my favorite. Note that this would not be a difficult
feature to add, but it would be tedious since every type of channel
would need to have every Dial exit area
On Mon, 2003-07-28 at 08:36, Peer Oliver schmidt wrote:
Ok, the first three things I did. Unfortunately, I am no c coder. But
the logic to say german numbers is identical to the english logic, ie.
21 = twenty one
11 = eleven
210 = two hundred ten ('and' between Hundred and ten is optional)
Hello all,
I'm trying to get CAPI to work with two B channels (AVM B1 PCMCIA)
on a P4 2GHz (linux kernel 2.4.21) system. All are ok with just one
B channel. But when I open a second B chan, the sound is choppy,
with too long gaps, and the CPU load is too high (~50%).
On the Asterisk's console I
Hi!
Please slow down on that, does the KX-T336 support T1/E1 cards
If yes, where can I find them?, how much does they cost?, do the KX-T336
need any upgrade for that?
See, at my university, we have a KX-T336, 144 lines, 288 extentions, and
we want to install asterisk, but, according to
Armand A. Verstappen wrote:
On Mon, 2003-07-28 at 08:36, Peer Oliver schmidt wrote:
Ok, the first three things I did. Unfortunately, I am no c coder. But
the logic to say german numbers is identical to the english logic, ie.
21 = twenty one
11 = eleven
210 = two hundred ten ('and' between
Is there any option or whatever for preventing the hangup for the
originating channel and go on in the current context ?
Not currently.
Or, is there a way to implement this feature ?
Absolutely. It is only necessary not to return -1 if the hangup was on
the other side. This could be
Also forgot to mention that you should make sure that the isdn and
hisax modules are loaded by doing :
modprobe isdn
modprobe hisax
I now have incoming calls working OK and working on getting outgoing
working.
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Where i can say n for next ?
I can not see an option n in your description in app_dial.c
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Mark
Spencer
Gesendet: Montag, 28. Juli 2003 15:52
An: Asterisk User
Betreff: Re:
Adam why would you need a channel Bank for the E100P?
Regards
Mark McKibbin
As I specified below:
Again, with smaller offices (say 15 or less extensions) the price of a
TE400P card is too expensive, will there be a TE100P card produced and
approved for use in Australia?
Even if it is,
Aehmm... (Mark,)
:-) i think i understand now ... i have to make the n option by myself.
I'am not the best in the english language and i don't know the niceties in
their.
Forgive me,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von
The Nortel 350/390/480 can't be locked just by dialing into an ADSI
service. The phone might not accept programming for that particular slot
anymore, but once the phone is cleared, you can reprogram at will.
The lock is a hardware lock.
To clear the phone, you can follow this procedure. Again,
Klaus-Peter Junghanns wrote:
Hi creator of the first H323 channel ;-)
Hi creator of the only CAPI channel driver :))
does the B1 get its own irq? did you try different versions of
AVM's capi drivers?
Yes, the B1 gets its own IRQ and, yes, I did try different
AVM firmwares
Hi,
The call transfer function in Asterisk seems to work in a way which does not
permit to ATA186 (or any othet hardware phone with only pne line) to have
this feature.
If someone tries to transfer a call to an ATA186 based extension, the call
is transferred to the correspondent voice mailbox,
It's fixed now
On Sun, 27 Jul 2003, Michael Bielicki wrote:
we have now perfect results with yesterdays cvs and the te410p
todays cvs allways thinks that immediate is set to yes in zapata.conf. weird
...
cheers
Michael
On Sunday 27 July 2003 7:12 pm, Mark Spencer wrote:
I put a TE410P
Is there any such thing is a userguide for asterisk from an enduser point
of view ie. what to do to transfer a call etc ? I've looked through all
the official documentation and nothing exists, and trying to install an
ASterisk at a client can't even explain how to transfer a call to another
Dave Alan Caruana wrote:
Even some basic help would be welcome!
Hang out and ask reasonable questions in the Asterisk IRC channel:
irc.freenode.net #Asterisk.
Jeremy McNamara
___
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[EMAIL PROTECTED]
The basic call transfer functions, set with the T and t options to the dial
application and triggered by pressing a # work fine for me. Make sure that
you have set the DialPlan on the ATA 186 so as not to grab the # (ie look
for any # character pairs and change the second character or remove
Hi Iain,
The basic call transfer functions, set with the T and t options to the
dial
application and triggered by pressing a # work fine for me.
I have T and t options in dial application, but how can '#' be used for
transfer.
Escuse my ignorance...
Make sure that
you have set the DialPlan
Good evening Peer,
well, it?s some time ago that I?ve updated my Asterisk. I would say it?s a
little bit outdated:-) (Asterisk is CVS-02/16/03 and Chan_Capi is
0.0.1c-RC4)
But what I?m using is an ISDN Fritz! A1 Card connected to an Agfeo AS141+
PBX on its internal ISDN Bus.
Please tell a little
On Monday 28 July 2003 12:24 pm, Dan wrote:
Hi Iain,
The basic call transfer functions, set with the T and t options to the
dial
application and triggered by pressing a # work fine for me.
I have T and t options in dial application, but how can '#' be used for
transfer.
Escuse my
I meant that attended transfer doesn't work (at least for me) when I'm trying
to transfer call to a different device too.
Let's say I dial from ata186 another h323 endpoint. Put it on hold. Then dial my
cell phone (ata - asterisk chan_h323 - h323/pstn gateway). Then if I hang
up on ata the call to
Hi Michael,
I meant that attended transfer doesn't work (at least for me) when I'm
trying
to transfer call to a different device too.
It works for any other type of IP phones, except ATA186. Tested on Cisco
7960 and X-Lite.
Let's say I dial from ata186 another h323 endpoint. Put it on hold.
Hi,
I have resources available to host a portal
specifically for the Asterisk system, to help correlate documentation, FAQ's and
How To
I am new to Asterisk, and my hardest work is in
locating information on using or configuring the software.
Would Mark, John or any of you feel this would be
Title: Message
Just a
suggestion, but wouldn't it be more appropriate for Digium to host the
documentation?
I
think the missing link here is someone who will write (and illustrate) the
documentation. All of this open source software is great because it's free
- but commercial users and
do Anyone utilize the d-link dg102s gateway ?
I have to set the d-link to use g723 codec with asterisk, but i don´t have
correct parameters for profile in D-link.
If anyone have the correct parameters values for this:
Tx Coding = G723 6.3 kbps
Rx Coding = G723 6.3 kbps
Scott,
I dont disagree.
I'm starting for where you were, and find it very difficult to gather all the nuggets
of infomation together.
The offer is still open, as a central repositiory, etc, unless Digium would like to
take the opertunity?
Damian
Title: Message
Hi,
I know
of a person who would probably write the docs for FREE.
He is
British with a superb understanding of English language.
Anyone
in digium interested, please let me know!
Senad
Jordanovic
-Original Message-From: Damian Flynn
[mailto:[EMAIL
Thanks Wipeout. I ordered a couple of licenses and have them running in the
lab. The codec works pretty good so far.
I noticed that the transmitt packet time of the g.729 codec seems to be
hardcoded at 20ms. Is there anyway to adjust that via a config file? Most
implementations allow you to
I saw that if I add a c to my Dial string as follows:
exten = s,3,Dial(Zap/g2c/18005551212)
That it will not consider that call as answered until the
called party presses #. When the number dialed picks up does a bridge of
the call immediately instead of waiting for the # key.
unsuscribe
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Yes, i've observed the same operation :|, Adam.
I've the last CVS Asterisk, and two softphones (Linphone 1.12 and X-lite
v2 last version), both with speex code active.
When i call from one to another ... ringing ok but ... when try to talk
... the Asterisk go crazy warming out of memory (i
Agreed. We're more than happy to host it. The problem is the writing of
it :)
Mark
On Mon, 28 Jul 2003, Scott Stingel wrote:
Just a suggestion, but wouldn't it be more appropriate for Digium to host
the documentation?
I think the missing link here is someone who will write (and
I'm interested!
Mark
On Mon, 28 Jul 2003, Senad Jordanovic wrote:
MessageHi,
I know of a person who would probably write the docs for FREE.
He is British with a superb understanding of English language.
Anyone in digium interested, please let me know!
Senad Jordanovic
-Original
I don't think confirm is supported by PRI at this time. I would suggest
adding something to the bug tracker.
Mark
On Mon, 28 Jul 2003, Alex Lopez wrote:
I saw that if I add a c to my Dial string as follows:
exten = s,3,Dial(Zap/g2c/18005551212)
That it will not consider that call as
Title: Message
Cor
Blimey! British...with a superb understanding of English language
?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad
JordanovicSent: 28 July 2003 20:51To:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
Hi!
Sure, just look for: Wonder Shaper. It's a HTB based shaper
configuration wich have some very good features, I use a variation of
that here at my College.
http://lartc.org/wondershaper/
It is the page (a simple google search).
Also make sure to uncomment the line tos=lowdelay in every
Here are a few things I would like to see ..
1. In addition to time/date stamps, store/read the caller id info with the
voicemail messages.
2. Have the ability to configure the system to ignore and delete messages
left by a caller that are 3 seconds or less (maybe make this configurable)
Not
On July 28, 2003 02:16 pm, Brian West wrote:
Here are a few things I would like to see ..
1. In addition to time/date stamps, store/read the caller id info with the
voicemail messages.
I am in the process of working on a patch to do that, among other things. If
you like, take a look at
On Monday, July 28, 2003 9:46 AM, Wade Weppler
[SMTP:[EMAIL PROTECTED] wrote:
The Nortel 350/390/480 can't be locked just by dialing into an ADSI
service. The phone might not accept programming for that particular
slot
anymore, but once the phone is cleared, you can reprogram at will.
Ricardo
Have you tested g729 between two endpoints (SIP) for over 5mins?
My experience has been that after 3-4 mins both ends begin to get huge
delays and after a few minutes is impossible to continue the conversation.
HAve you done any testing similar to mine?
- Original Message -
The hardware lock is really just a security code on each/all of the slots.
This security code cannot be changed or erased from the phone. If you don't
have the code, you can't program the phone.
You can also software lock a slot to prevent it from being overwritten by
another program (Bell
On your sip.conf for each sip endopoint set canreinvite = yes.
That way the rtp stream won´t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL
I have been planning to suggest this as well, but I would recommend setting
up a zope site...
If you set it up in zope you can have alot of collaboration very easily.
You could, for instance, designate certain people who have expertise in a
certain config/technology as project coordinator for
Title: Message
Hi
All,
Last week I had
problems installing the TDM DK on my Dell Dimension 2100. Digium support
confirmed that there is an incompatibility problem between the TDM card
(TDM400P) and the motherboard, and that problem will be fixed sometime in
September (which is too late
please remove me from this discussionDoug Reisinger
President, Alpine Broadband
cell 303.522.5434
[EMAIL PROTECTED]
From: "Andy Hester" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Offering an Asterisk Documentation and FAQ Portal
On Monday 28 July 2003 07:19 pm, DOUG REISINGER wrote:
htmldiv style='background-color:'DIV
Pplease remove me from this discussionBRBR/P/DIVBRBRBRDoug
Reisinger
Can the list owner please change the list to reject x-html posts.
Regards...Martin
--
Mitchell's Law of Committees:
Any
Why don't you just plug the e100p into the onramp10-30 why do you need a
channel bank?
Regards
Mark McKibbin
DCS Internet
64 Queen St
Warragul
Victoria3820
Australia
www.dcsi.net.au
[EMAIL PROTECTED]
Ph. 1300 665 575 (Help Desk)
Ph. +61 356 241 120 (Direct)
Fx. 1300 556 595
-Original
Does anyone have any hunt group examples?
phone 1
phone 2
phone 3
message press 1 to leave a message
loop back to phone 1
till the call is answered...
Any examples?
Thanks,
Brian
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I have a number of Welltech FXS devices (running the SIP code) that I'm
trying to register with Asterisk. Has anyone had any success doing this?
If I have the Welltech set in peer-to-peer mode, I can call to and from the
Welltech, to Asterisk, just fine. However, because it's in peer-to-peer
Is there a way in iax to have to endpoints talk to
each other directly (after the call is setup by *) without going through
*. In sip, with * you can do it by
configuring sip.conf with canreinvite = yes.
Check out this bug
http://bugs.digium.com/bug_view_page.php?bug_id=005
its a know problem. I have played with the canreinvite stuff to no end and have never
gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if
it does P2P with my Cisco Phones then it might
Why don't you just plug the e100p into the onramp10-30 why do you need a
channel bank?
Regards
Mark McKibbin
How do you connect your extensions then? If you use the single T/E 1 port
for your onramp, you end up with nil extensions, using a channel bank you
can use the single T/E1 port for
Perhaps the solution is a wiki.
Then the documentation could be a community effort.
Lots of free Wiki sofware avaibable implemented in languages of your choice
See: http://www.c2.com/cgi/wiki?WikiEngines
The most full featured one is:
www.tikiwiki.org
Supports wiki, and you can optionally
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