it works well with chan_capi.
Marian Danisek wrote:
Hi,
does anybody know if primus isdn cards - they support capi under linux,
provided by own driver are usable with asterisk together with capi
channel driver ?
http://www.primuxisdn.de/primux/index.htm
regards
Marian
On Fri, 2003-10-03 at 03:30, Chris Albertson wrote:
I checked out gnophone from CVS and I'm trying to build it.
I got as far as getting a ./configure built and that to
build the makefiles and then I find compile poblems in the source.
Leads me to thing maybe 0.2.5 is still a work in progress.
hi there ..
I asked sometime ago regarding getting a Budgettone
working with Asterisk over G729.
My system is quite simple, Asterisk server with 1 G 729 license
installed, and 10 Grandstream phones. Only one of them needs
G729, because it's on a remote link via an ADSL bridge. The
rest run
Hi,
Unless there has been a recent change, you can't set codecs in the sip.conf
on a per-context basis. The way to do what you want is to have the following
in the [general] area:
disallow=all
allow=ulaw
allow=alaw
allow=g729
Then, set all the codec preferences on the g729 phone to g729. That
many thanks!
it's all pretty much a hit miss process
becaues of Asterisk's notorious lack of documentation..
I will try this out at my client tomorrow,
fingers crossed.
cheers
Dave
- Original Message -
From: T Aksoy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003
If we avoid g729, what options do we have for SIP/Budgetone/Grandstream when
communicating to * via the Internet and still have something comparable to
GSM?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter
Sent: Thursday, October 02, 2003
Hi,
Here is a bit of a long shot..
Does anyone have a gsm sound file that says somthing like All our
(outgoing) lines are currently busy, Please try again later.?
Or anything close?
I would record it my self but I hate hearing myown voice on the prompts.. :)
later..
Hello all, I'm new to this list and starting with Asterisk.
Have any of you have tried a SIP client (like Microsoft messenger) to
sent text messages and voice through an Asterix server?
Is this possible or the Asterix server simply can't manage this kind of
traffic?
Regards,
Luis
Luis Vazquez wrote:
Hello all, I'm new to this list and starting with Asterisk.
Have any of you have tried a SIP client (like Microsoft messenger) to
sent text messages and voice through an Asterix server?
Is this possible or the Asterix server simply can't manage this kind
of traffic?
Hi list ,
I am able to compile the old version of * PBX from
the CVS dated 20 Sept 2003 when the mysql support was there ..
But i have not been able to find out that hoe the
Voicemail2 will work with mysql database. Means Is there any file like
cdr_mysql.conf file where we have to specify
Hi!
I have some question about the use of codecs in sip.conf
I have that lines in sip.conf:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
when i use show codecs:
localhost*CLI show codecs
1 (1 0) G.723.1
2 (1 1) GSM
4 (1 2) G.711 u-law
8 (1 3) G.711
Hello all again,
Last time you helped by suggestiong that monitor will record by
telephone conversations - I have added this to my config - but where
does it save the files?
Nick
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Nick Knight wrote:
Hello all again,
Last time you helped by suggestiong that monitor will record by
telephone conversations - I have added this to my config - but where
does it save the files?
Nick
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[EMAIL PROTECTED]
Hello all,
I have a question about the pbx spool and extensions.
After I put a sample.call in /var/spool/asterisk/outgoing,
Asterisk calls out and jumps to context ringback.
[ringback]
exten = s,1,ResponseTimeout(5)
exten = s,2,DigitTimeout(5)
exten = s,4,SetVar some variables
exten =
I will join the other list, but as a statement from the
industry We actually utilize it for our existing client base,
it is not for direct cold-calling of new clientel. I am sure that most
ppl don't like to have the automated/manned systems from many companies
calling them, but
Hi everybody!
I am trying to do a H.323-SIP Gateway and someone have told me that
asterisk would help me. Has this software this functionality? If it has, so
what must I do to make that everything works ok?
Thanks a lot for your answers!
___
Hello,
I've seen various suggestions thrown around for hardware when people ask,
but can we all agree on some basic hardware recommendations for a few basic
setups(and post them on a website) to make it easier for new people to avoid
some of the hardware/software pitfalls when they are setting up
I have a www.linuxrouter.org fw/nat would luv
to test this
-Original Message-From:
sip [EMAIL PROTECTED]To: [EMAIL PROTECTED]
[EMAIL PROTECTED]Date:
October 2, 2003 8:34 PMSubject: [Asterisk-Users] THE
NAT-MARE IS OVER test volunteers needed
5volunteers
There is no need to use oh323. If you look in
/usr/src/asterisk/channels/h323 then you will find that there is already an
h323 implemenatation present (chan_h323). You just need to follow the
instructions and it works great.
Tan
telappliant.com
-Original Message-
From: [EMAIL
no /var/spool/asterisk/monitor - although there is a /var/spool/asterisk
clippnig from extensions.conf
exten = 870582,1,Wait(1)
exten = 870582,2,Monitor()
exten = 870582,3,Dial(${EVERYONE},10)
exten = 870582,4,capiCD(${NUMBER})
exten = 870582,5,Hangup()
Nick
Hi!
I have downloaded asterisk... and I have installed it how I continue? I
don't know anything about the software and I don't really understand how it
works please any help will be ok.
Does it exist some manual?
Thanks a lot!
___
you'll find that the context is being overwritten.
look in chan_iax.c line 1628 and chan_iax2.c line 1645 (or within
3 lines of each)
there is a sprintf that is stuff the context, if you comment
those out, it should work again.
Disclaimer: i have NO CLUE what else this BREAKS!!!
Dave Weis
On Friday 03 October 2003 08:06 am, listas iPfone wrote:
I have that lines in sip.conf:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
when i use show codecs:
localhost*CLI show codecs
1 (1 0) G.723.1
2 (1 1) GSM
4 (1 2) G.711 u-law
8 (1 3) G.711
On Friday 03 October 2003 07:54 am, Sip Rtp wrote:
I am able to compile the old version of * PBX from the CVS dated 20
Sept 2003 when the mysql support was there ..
But i have not been able to find out that hoe the Voicemail2 will
work with mysql database. Means Is there any file like
Has anybody else out there had a problem with
transfers not being detected?
Occasionally I will want to transfer somebody, so I'll
hit the # key and instead of the Transfer application
starting, the # tone is played.
My hardware is T100P connected to an Adtran TA 750.
I have relaxdtmf=yes in
Hello,
We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We have
a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The
fritz card is configured to use chan_capi, and we can make calls SIP-SIP
SIP-PBX extension PBX extension-SIP all successfully, we have assigned
Below are some links that should point you in the right direction. Assuming
you don't have any IP Phone on hand I would recomment starting with to
computers with softphone(one example
http://www.eutecticsinc.com/download6/DISK1/IPP200_SJSoftPhone.htm) and
have them talk to each other.
Hi Dave,
try :
Dial(CAPI/${CALLERIDNUM}:${EXTEN},10)
(make sure you have a msn= line in capi.conf that allows
this for ${CALLERIDNUM})
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30
I have an IconnectHere account
with a Inbound number and have setup the sip.conf to register and am recieving
the call but When I answer the call it disconnect. I have tried sending the call
to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon
as I accept the
Steve,
I don't have any real experience in DSP methodologies, although I have picked up
on the high-level theories in my research. However, I am *very* strong-willed
in the Where there's a will, there's a way category. :)
Here's my current thought:
Sphinx is an open source STT library that
Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager
image?
I want to start playing around with the chan_skinny addition, but it seems the .exe's
from cisco want to open a connection to a SQL server or CallManager (which I don't
have).
Dustin,
It's quite a pain to get those without a CallManager...
However, there are some tools for extracting the compressed files from
an InstallShield image and I have successfully done so with those files
in particular and was able with some tweaking to get a phone back to
Skinny without
I've moved back and forth using the OS79XX.TXT file on the TFTP server.
Copy the bin file, modify the OS79XX.TXT to have name of the firmware
image and power cycle the phone.
The only issue with this process is that once you have the phone running
the firmware you want, you need to clear out the
I have a cisco 7940 with the following sip.conf config:
[Desk1.1]
type=friend
secret=**
defaultip=192.168.1.14
insecure=no
mailbox=102
callerid=Desk1.1
qualify=500
canreinvite=no
context=extensions
host=dynamic
group=2
I do not get message waiting indicator (mwi) on this phone. Is the
Hi all!
Are there some people who have already implemented a SIP - H.323 Gateway? I
am trying to do so... but I don't know how. Please if anyone can help me...
Thanks a lot for all your answers.
Mireia
___
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[EMAIL
I would love to see this kind of information.
Hello,
I've seen various suggestions thrown around for hardware when people
ask, but can we all agree on some basic hardware recommendations for a
few basic setups(and post them on a website) to make it easier for new
people to avoid some of
Ah - just got it finished!!
In case anyone else has this problem, here's what I done:
1) Extract the cmterm- file from cisco in order to get the data1.cab (I used
linux's cabextract)
2) Extract the two files 'P00305000200.bin' 'P0030500200.sbn' of the data1.cab file
(I used i6comp for
I think you need [EMAIL PROTECTED], where voicemailcontext is
your voice mail context. (I'm assuming you're using VoiceMail2.)
Hope that helps...
Jared Smith
On Fri, 2003-10-03 at 12:01, Babak Pasdar wrote:
I have a cisco 7940 with the following sip.conf config:
[Desk1.1]
type=friend
Is there anything special needed to load up a TDM10B card??
I got the card today.. Took it from the box, put it into a PCI slot..
connected the power to the card and booted the PC..
I have removed the X100P to avoid confusion and I have the following in
the config files..
in /etc/zaptel.conf
Hi!
I´m thinking inan incoming number from
ICH
please share your sip and extensions.conf files off
list, it will help me a lot.
miklos
- Original Message -
From:
Glenn
Dalgliesh
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003 2:17
PM
Subject:
Hi!
Anybody have experience using asterisk and 3com voip systems?
Miklos
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
We have the 3com NBX100 here at work..if anybody has ANY info on
integrating * and the NBX I'd LOVE to hear about it. I'm not very happy
with the NBX but loving what I'm learning about *.
Hi!
Anybody have experience using asterisk and 3com voip systems?
Miklos
Jared,
Thanks for the response. I have tried
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
none of these worked for me.
I have enclosed part of my entensions.conf as well as my voicemail.conf
Hopefully this will provide pertinent information.
Again thanks for your help.
Bbaak
Hey all...for whatever reason my caller id doesn't appear to be working.
My setup is simple (Wildcard FXO and thats it) and I'm just expecting
the Caller ID to show up on the console.
I'm seeing this:
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238
Is it showing up on /proc/pci? It should be a tigerjet.
Yes. I put the other card back in (production machine) but over the
weekend I'll get the card in there and capture the output of lspci.
If the card shows up in /proc/pci then your motherboard *must* be
supplying 3.3V somehow (unless
This issue was resolved by adding the @context in the voicemail.conf
file for the extension to the mailbox=XXX command.
[EMAIL PROTECTED]
Thanks so much for your help.
Babak
On Fri, 2003-10-03 at 14:43, Babak Pasdar wrote:
Jared,
Thanks for the response. I have tried
[EMAIL PROTECTED]
On Fri, 2003-10-03 at 13:28, Chris Hirsch wrote:
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
(Ring/Answered)...
I've only seen this message when using callprogress=yes and/or
busydetect=yes. Set them to no.
--
Sample configs and more:
I'm still seeing this:
*CLI -- Starting simple switch on 'Zap/1-1'
NOTICE[262161]: File callerid.c, Line 238 (callerid_feed): Caller*ID
failed checksum
NOTICE[262161]: File chan_zap.c, Line 4314 (ss_thread): Got event 2
(Ring/Answered)...
My zaptel.conf:
fxsks=1
loadzone = us
defaultzone=us
many thanks!
it's all pretty much a hit miss process
becaues of Asterisk's notorious lack of documentation..
I will try this out at my client tomorrow,
fingers crossed.
cheers
Dave
- Original Message -
From: T Aksoy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 03, 2003
Hi,
Here is a bit of a long shot..
Does anyone have a gsm sound file that says somthing like All our
(outgoing) lines are currently busy, Please try again later.?
Or anything close?
I would record it my self but I hate hearing myown voice on the prompts.. :)
later..
You can probably patch
; SIP Configuration for
Asterisk;[general]port =
5060
; Port to bind tobindaddr =
0.0.0.0
; Address to bind tocontext =
sipinbound ;
Default for incoming callsregister =
1410344:[EMAIL PROTECTED]/1410344
--=-=-=-= extentions.conf-=-=-=-=-=- have also
tried sip phone same
Hi
Somebody can tell me wich card i need to use that suport em signal, and
is posible to connect this card to any PBX
Thanks
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
www.diguim.com
Wildcard T100P - Single t1
http://www.digium.com/downloads/product_sheets/T100P.pdf
Wildcard TE410P - 4 port - I believe support independant config of each T1
http://www.digium.com/downloads/product_sheets/TE410P.pdf
Wildcard T400P - 4 port -
Is there a good way to detect FAX and Modem on a call that is established
and then take some sort of action? What I have is a situation that all
calls going out through an asterisk system are being recorded. Some of those
calls are internal fax machines or modems. When monitoring is turned on
Tilghman Lesher wrote:
On Friday 03 October 2003 07:54 am, Sip Rtp wrote:
I am able to compile the old version of * PBX from the CVS dated 20
Sept 2003 when the mysql support was there ..
But i have not been able to find out that hoe the Voicemail2 will
work with mysql database. Means Is there
Ask the producers to implement iLBC:
http://www.globalipsound.com/products/iLBCfreeware.php
It is free and from my experience one of the best codecs available.
Grandstream promised to implement it in the future :-). X-lite and
kphone ( http://www.wirlab.net/kphone ) support it.
Jan.
On 03-10
On 03-10 12:45, Luis Vazquez wrote:
WipeOut wrote:
Luis Vazquez wrote:
Hello all, I'm new to this list and starting with Asterisk.
Have any of you have tried a SIP client (like Microsoft messenger) to
sent text messages and voice through an Asterix server?
Is this possible or the
I would love to see this kind of information.
I was talking a while back about a registry of live systems out there, kind
of like the Asterisk version of the Linux Counter. It's still on my list of
things to do, I've scribbled some notes so far, just haven't got round to
doing it..
What if you separate the fax machine channels to diffrent contexts that
don't call application Monitor ? It's for outgoing calls and for incoming
calls if you have certain extensions for faxes you can call StopMonitor
application.
regards
Martin
On Fri, 3 Oct 2003, Nicholas Romero wrote:
Is
Hello -
Here's my first impression review of the first SIP 802.11 phone. I
got my hands on the first one sold, so that perhaps makes me the
first person to have a real 802.11 SIP phone commercially in the US
interworking with Asterisk. Whee! Can someone point me to other
commercially
So, is Astrisk being changed to an OSI-compliant license without the
anti-patent clause?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter
Sent: Thursday, October 02, 2003 2:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help with
nice review!
how about call hold transfer ?
they works? and what type of transfer is supported?
Matteo.
Il sab, 2003-10-04 alle 01:09, John Todd ha scritto:
Hello -
Here's my first impression review of the first SIP 802.11 phone. I
got my hands on the first one sold, so that perhaps
The anti-patent clause was dropped ages ago.
Mark
On Fri, 3 Oct 2003, Uriel Carrasquilla wrote:
So, is Astrisk being changed to an OSI-compliant license without the
anti-patent clause?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan
Hi List,
I am sorry that I may bring the old question to the community. My question
is
1. How can we determine if asterisk is working normally or not ? what kind
watchdog process do we have at this moment ?
2. In case the running asterisk is mulfucntion, is there any available way
to auto
On Fri, 2003-10-03 at 18:09, John Todd wrote:
Hello -
Here's my first impression review of the first SIP 802.11 phone. I
got my hands on the first one sold, so that perhaps makes me the
first person to have a real 802.11 SIP phone commercially in the US
interworking with Asterisk.
dsp.c has silence detection that works quite well for detecting end-of-voice
silence. It is used to allow only a certain amount of silence at the end of
voicemails, for instance. See app_voicemail2.c on how to use it, specifically
the function play_and_record(). Note that the silence threshold
That would be a little more ideal but the way in which the faxes pass though
the system the both the port/channel and the destination are dynamic. There
are some special situations that I have some modems that are dialed enough
that they are specified. These specific destinations go to another
Off the top of my head maybe something like this?
while(1)
asterisk
wait `cat /var/run/astrisk.pid`
sleep 1
end
Looking at the above it could be improved, like checking
for the /var/run/astrisk.pid file before the wait and
bailing out if not found.
--- [EMAIL PROTECTED] wrote:
Hi
I found it. but that webite is chinese BIG-5. take care.
http://www.mpn.com.tw/index-big5-PRODUCT.html
and that already released by Fujitsu.
http://www.net-2com.com/jp/product/hw/wireless_ipphone/
mack_jpn
On Fri, 03 Oct 2003 19:03:10 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
On
After some months of Make updates, I have just deleted my Zaptel and
Asterisk source directories and done cvs checkout 's of asterisk and
zaptel, in order to clean up the trees.
After re-installing, I am finding that when dialling into an X100P, that
Answer is now answering on the second
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Has been fixed.. This is what happens when you don't pay your bills..
:(
J
-Original Message-
From: PJ Welsh [mailto:[EMAIL PROTECTED]
Sent: Friday, 3 October 2003 12:52 AM
To: [EMAIL PROTECTED]
Subject:
When I place a call using Iconnecthere
or Nikotel as my sip provider, I hear no ringback when making a call. I do see call progress in the console. Does anyone else have this problem or
offer any suggestions? Thanks, Kevin
On Fri, 2003-10-03 at 20:11, Masakazu Nakano wrote:
I found it. but that webite is chinese BIG-5. take care.
http://www.mpn.com.tw/index-big5-PRODUCT.html
and that already released by Fujitsu.
http://www.net-2com.com/jp/product/hw/wireless_ipphone/
Actually, this is the one I was
Glenn Dalgliesh wrote:
I have an IconnectHere account with a Inbound number and have setup the
sip.conf to register and am recieving the call but When I answer the
call it disconnect. I have tried sending the call to from * to a
Softphone, Pingtel, and FXS port and all result the same. As soon
Mark Spencer wrote:
Hello Asterisk users and developers! I wanted to let you all know that
Digium is hiring for a full-time technical support position. This job
includes:
* Benefits
* Working with Open Source Software
* Lots of opportunity for advancement and growth
* Working directly with
[EMAIL PROTECTED] wrote:
Hi List,
I am sorry that I may bring the old question to the community. My question
is
1. How can we determine if asterisk is working normally or not ? what kind
watchdog process do we have at this moment ?
2. In case the running asterisk is mulfucntion, is there any
ChanIsAvail returns the channel ID plus -session.
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a warning, but dialing a SIP channel
completely errors out.
-- extensions.conf snippet-
;
May I ask the reason for the Tech Support person needing to be in
Huntsville? Wouldn't it be simple to have the person Virtually There
with Asterisk?
Chris
On Fri, 2003-10-03 at 17:50, Mark Spencer wrote:
Hello Asterisk users and developers! I wanted to let you all know that
Digium is hiring
May I ask the reason for the Tech Support person needing to be in
Huntsville? Wouldn't it be simple to have the person Virtually There
with Asterisk?
I thought about that. In principle they could be anywhere, but in
practice, it's often more productive when people are together and you can
take out usecallerid=yes in zapata.conf
Martin
On Sat, 4 Oct 2003, Richard Scobie wrote:
After some months of Make updates, I have just deleted my Zaptel and
Asterisk source directories and done cvs checkout 's of asterisk and
zaptel, in order to clean up the trees.
After re-installing, I
maybe just have a 'first line' TS that's vitual, if that does become an
option.
lord, knows that's been done before.
Mark Spencer wrote:
May I ask the reason for the Tech Support person needing to be in
Huntsville? Wouldn't it be simple to have the person Virtually There
with Asterisk?
I
Hello Asterisk users and developers! I wanted to let you all know that
Digium is hiring for a full-time technical support position. This job
includes:
* Benefits
* Working with Open Source Software
* Lots of opportunity for advancement and growth
* Working directly with Martin and Mark
*
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