hi,
As i understand (correct me if i am wrong) R2 signalling consists of 2 parts: Line Signalling (supervisory signalling)and the InterRegister Signalling (call control signalling).
Now i am testing this in a lab with an Analyzer. TheIdle CAS ABCD bits are 1000. Thelatest problem that i see is
Hi,
I was thinking if it was possible to get this list as news ?
It would be much easier that 'hotmail-account'
/HHA
_
Scope out the new MSN Plus Internet Software optimizes dial-up to the max!
I'd like to get some feedback from users of USB headsets as to what they
like/dislike about the unit they own (manufacturer/model number). I'm
looking to buy some. Is there already a thread somewhere or a review (I
tried to find one with no luck), discussing this topic?
Any
Maik Schmitt wrote:
Has somebody got it work at all ?
I mean data calls (ISDN 64k) through asterisk.
Yes. Works fine here with a PRI from DTAG and an Ascend.
Maik,
so you have a PRI from DTAG into asterisk and an Ascend access server as
a PRI extension where users can dialin from the PSTN
On Fri, Jan 23, 2004 at 07:00:49AM +, Hans-Henrik Andresen wrote:
Hi,
I was thinking if it was possible to get this list as news ?
http://www.gmane.org offers many mailinglists as a newsfeed.
Even VoIP stuff such as * and SER.
You can also read/search the mailinglists via the web
Hi,
-Original Message-
from 64.76.148.186:2427Verb: 'ntfy', Identifier: '6001', Endpoint:
'aaln/0/[EMAIL PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Jan 22 18:05:11 NOTICE[49159]: chan_mgcp.c:1102
find_subchannel: Gateway
'ap1' (and thus its endpoint 'aaln/0/0') does not
Hi Siggi/Jan,
If so, there's still a load version conflict (although I've
never seen a
7960 or 7940 care about the version communicated through SCCP):
On the phone, press Settings, then 4 for load information.
watch out for the App-Load-ID. On my 7940, this is
P00305000300. Yours
is most
On Thu, 22 Jan 2004 at 23:36, Ken Alker wrote:
I'd like to get some feedback from users of USB headsets as to what
they like/dislike about the unit they own (manufacturer/model
number).
I've got a Labtec Axis 712 stereo USB headset. They also produced it
in a mono version (Axis 711). Both
Ariel Batista wrote:
I have 2 Snom 200 and would like to get them to work properly with
Asterisk. With the Firmware 2.02t I am able to use the phone. But only
one line configured. With there newer firmware 2.03o it will not allow
me to make calls. But I can get calls on the unit. Again the
Greate - it works.
Thank you
/HHA
http://www.gmane.org offers many mailinglists as a newsfeed.
___
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Ariel Batista wrote:
clipcomm people?
Well, I was/am looking for a device with PSTN FXO backup.
www.dlink.com does one like that, but is way too expensive.
I found information on a D-link DVG-1120S (Sip unit.) It has 2 FXS
ports and on FXO port. This would make a nice small office
Hi,
I make some more tests and the results are a little bit strange...
My testbed consists of an active card and I used linphone as client. The
results are as described here in my previous posts. But today I tested
with xTen Lite as client... And it works. I take some sniffer traces and
the
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC
-Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES
I could not see anything there that is working.
Even the normal SIP connection. (Gives noisy output in the phone)
(Doesn't support stun) It is not NAT friendly.
FXO is utter waste option on this.
I have tried with the filter as what i read in the previous email in the
list, that doesn't works.
Hello,
I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
telephone on (no
Does that connects VoIP to PSTN or only on Fail Over, means it changes from
choosing the PSTN line instead of VoIP line?
Kannaiyan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 9:44 AM
Subject: RE: [Asterisk-Users]
Thanks. I solved this problem using a cross-cable.
Daniel
CW_ASN - Gus wrote:
Please send your zaptel.conf to see what's going on.
- Original Message -
From: "Daniel Bichara" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:38 PM
Subject:
Samuel Jimenez wrote:
Hi,
Assuming that the problem *is not* soft settings I would recommend you to
verify that the channel mapping is the same in your adapter as in your
E100P. Some times they do not match, and the D channel does not arrive
where expected at each end.
Hi Sam,
My
I have reported to clipcomm, but they were on holidays until end of this
month.
Kannaiyan
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:29 PM
Subject: RE: [Asterisk-Users] Standalone FXO device
Kannaiyan Natesan
M.A. Ali wrote:
hi,
As i understand (correct me if i am wrong) R2 signalling consists of 2
parts: Line Signalling (supervisory signalling) and the InterRegister
Signalling (call control signalling).
Now i am testing this in a lab with an Analyzer. The Idle CAS ABCD
bits are 1000. The latest
Hi!
I'm somewhat new to * and haven't encountered the MySql connectivity.
We use MySql and I was wondering what is achieved by installing this addon?
See:
http://www.voip-info.org/wiki-Asterisk+billing
There are also - less established - ways to manage sip.conf,
extensions.conf and
Hi!
1. The spec calls for a 24 analog line system with a fairly sophisticated
response matrix using SQL into Oracle text-to-speech (among other
things). Is Asterisk in the same class of product as Websphere, or is it
for a more straightforward voicemail office environment?
Not knowing the
Hi!
There are 60*60*24 seconds in a day 9E6 per messages/day
means a little over 100 messages per second sustained 24x7.
I really doubt it.
I know little to nothing about mailman, but this is where smart smtp
engines (esmtp?) and tools like LISTSERV come in that do everything they
can to
On Thu, 22 Jan 2004, Jeff Gustafson wrote:
[...]
no, its not necessary required. in this case, check that the contents of
OS79xx.TXT if they match with your current version.
I didn't have that file because I thought it would make things worse.
:) I took the number from Settings -
Can anyone give an idea how much does it cost if we want to buy the Licensed
asterisk source code?
I hope asterisk has two type of licenses,
1. GPL
2. I can buy and develop software on my own.
Am I right ?
Kannaiyan
___
Asterisk-Users mailing list
Maybe you have to update Asterisk (see
http://bugs.digium.com/bug_view_page.php?bug_id=732). snom is now a
little bit picky about line-ID (see the discussion in the bug).
CS
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ariel
Is Voicemail2 still able tohave
connectivitywith mysql.
I know the example was in voicemail.conf where you
specified dbhost,dbname,dbuser... etc.. , but dont
see any more in the latest CVS download. I am aware
of the asterisk-addons feature. I have installed the add-ons and am using Mysql
Kannaiyan Natesan wrote:
Can anyone give an idea how much does it cost if we want to buy the
Licensed asterisk source code? I hope asterisk has two type of
licenses,
1. GPL
2. I can buy and develop software on my own.
Am I right ?
Kannaiyan
Get in touch with www.digium.com .
Hi,
I am new in asterisk.
Is it possible to use it with common dialup modem to connect
ptsn to the server?
Thanks
Regards,
Soragan
On Thu, 22 Jan 2004 20:27:28 -0300
CW_ASN - Gus [EMAIL PROTECTED] wrote:
Maybe Telefonica (the same from .ar) is not big enough!
By the sight Telefónica in Brazil is not very serious, in Argentina
offers ISDN in all country, for all kinds of teleservices... I'm sure
of that.
In
I am sure mostly anything is possible with
asterisk, but I would definitely recommend you buy a X100P if you want to
connect to PSTN., check it out at their website
at http://www.digium.com/index.php?menu=wildcard_x100p
for the price its worth all the troubleshooting you'll have to go
so you have a PRI from DTAG into asterisk and an Ascend access server as
a PRI extension where users can dialin from the PSTN through asterisk
with modem *and* 64k data calls?
Exactly.
That's exactly what I am looking for!
Did you notice any loss of performance or reliability compared to
Why doesn't someone just ask Markster how many are on the list?
Inquiring minds want to know...
Chris Albertson wrote:
Maybe the words nine million was not ment to be taken
literally. What if he said about a gazillion Then we'd
all be arguing if gazillion == 1x10^14 or 1x10^16
Have you ever
Have you ever set up mailman on a Linux system? 9,000,000
would be a real trick setup not something you'd do with a
standard PC. You have a rack full of mail servers, raid disks
and the works. Lets do some math:
There are 60*60*24 seconds in a day 9E6 per messages/day
means a little
When you see 'X100P' cards for that price you can be assured that they
are either used Digium (who would want to part with their X100P??) or
more likely (as in this case and most cases) manufactured by a
third-party. I don't know if they are technically 'pirate' cards, but I
wouldn't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
How can I configure the TE410P card to act as master instead of slave?
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.3 (GNU/Linux)
It's been added to the wiki:
http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script
MATT---
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Friday, January 23, 2004 12:38 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom Reboot Script -
Do not compile openh323 and pwlib from cvs.
Use the versions described in the README of chan_h323 so (in
channels/h323 dir).
Good luck!
Doichin Dokov
Mike Bentley wrote:
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH
[EMAIL PROTECTED] wrote:
Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request:
Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1
That error means asterisk cannot transcode to GSM Make sure you have
not mistakenly noload'ed codec_gsm.so or havent' disallowed it
Hi All,
I've been having a hard time getting the AbsoluteTimeout function to work.
Is this Function working in for SIP? I've search all the messages in the
news letters and tried what was suggested and still have not gotten it to
work. Below is a portion of my extensions.conf. I've also been
Geert Nijpels wrote:
Ariel Batista wrote:
I had the same problem, so I emailed SNOM. After a quick and clear
reaction from SNOM, the following turns out:
I have an SRV record set for Asterisk using both TCP and UDP, because
I was first experimenting with SER and that SIP proxy DOES support
Key,
I've been playing with the Grandstreams for some weeks; one good way to see
the registration messages it to monitor the network with Ethereal. (packet
sniffer). You'll see the SIP messages coming and going, with complete
decoding. This works pretty much as predicted when using VOCAL.
Mike Bentley wrote:
Hi can anyone help me with this
g++ -g -c -o ast_h323.o -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN
-DNDEBUG -DDO_CRASH -DDEBUG_THREADS -pipe -Wall -fPIC
-Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT
-D_GNU_SOURCE -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS
Ariel,
I have 2 Snom 200 and would like to get them to work properly with
Asterisk. With the Firmware 2.02t I am able to use the phone. But only
one line configured. With there newer firmware 2.03o it will not allow
me to make calls. But I can get calls on the unit. Again the 2nd line
I use it in that way, it works very well:
exten = s,4,AbsoluteTimeout,600
miklos
- Original Message -
From: Wes Marderness [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 12:33 PM
Subject: [Asterisk-Users] SIP Absolute Timeout
Hi All,
I've been having a hard
Here's the follow-up article to the first article I published on Asterisk:
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html?page=last
This one covers getting Zap hardware installed, and also covers
integrating an IPCSP (IP Communications Service Provider - aka: long
distance via
I think this has been fixed since 0.5.0 their was a problem with timeout's
and native bridges. Might wanna update.
bkw
On Fri, 23 Jan 2004, Wes Marderness wrote:
Hi All,
I've been having a hard time getting the AbsoluteTimeout function to work.
Is this Function working in for SIP? I've
I have 8 lines coming into an existing PBX system and am looking for a cost
effective way to replace the existing system with Asterisk. We need some of
the features in Asterisk, including its ability to support remote offices
(long distance savings).
At first glance this appears to require a
Hi
I sugest you to make a reset and switch off the phone before upgrade.
It solved many problems for me.
Miklos
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 11:32 AM
Subject: Re: [Asterisk-Users] Snom 200 phones not
John Todd said:
United States:* +1-800-...
+1-888-...
+1-877-...
+1-866-...
via: Telesthetic/Local Exchange Carriers of Michigan
JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has
8xx dialing into
Building * on a machine with a minimal install of Mandrake, worked fine
on non minimal install now I get this:
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
If anyone can help me figure out what package I might have missed out
when installing mandrake,
FYI and to whom it may concern, I have made Debian
packages of Asterisk et. al. You still need to build
a new kernel and the zaptel modules from source, but
Asterisk and libpri are manageable with dpkg.
The debs as well as mirrors of the source distribution
are here:
The Packet8 8x8 DTA-310 that I have ran SIP when I was using it.
On Thu, 2004-01-22 at 23:11, Jeremy Jones wrote:
Hello folks,
I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no
dialtone can't get it to ring. My mgcp.conf says:
;
; MGCP Configuration for Asterisk
;
On Fri, 2004-01-23 at 09:30, Darren Martz wrote:
I have 8 lines coming into an existing PBX system and am looking for a cost
effective way to replace the existing system with Asterisk. We need some of
the features in Asterisk, including its ability to support remote offices
(long distance
Title: RE: [Asterisk-Users] 8 lines - best approach
One solution that we're investigating is using a gateway product instead of a channel bank.
There's a couple to choose from...take a look at http://www.voip-info.org/wiki-VoIP+Gateways
Sounds like you want one of the FXO devices. We
Title: RE: [Asterisk-Users] Debian Packages and Mirrors
Note that there are also asterisk packages in the standard Debian repositories
http://packages.debian.org/cgi-bin/search_packages.pl?keywords=asterisk=names=1=insensitive=all=all
v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in
Key Aavoja wrote:
Hello,
I have a problem with asterisk and Grandstream BudgeTone-100.
With default configuration everything works (in anonymous mode and fixed
IP), but if Im trying to enable registering, it dos not work.
I used 'sip debug' and verbose level 10, nothing happens if I switch
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
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To UNSUBSCRIBE or update options visit:
Hi
All
I have a problem with initialization of asterisk
using my rc.local file. when i call asterisk from the prompt it works well but
don´t in the initialization...
I have in my file that comands:
touch /var/lock/subsys/localmodprobe
zaptelmodprobe wcfxosafe_asterisk
I read in
This one does mgcp... It's been used in conjunction with a hosted pbx
system called Centile that 8x8 now owns. If there's a firmware image
anyone knows of to make these do sip, I'd rather do that. But for now,
mgcp help is what I need.
Jeremy
-Original Message-
From: [EMAIL PROTECTED]
Title: RE: [Asterisk-Users] 8 lines - best approach
Two Voicetronix Openline 4 port FXO cards would do
the trick. They run about $550 each.
John
- Original Message -
From:
Kostur,
Andre
To: '[EMAIL PROTECTED]'
Sent: Friday, January 23, 2004 10:40
AM
Subject:
Actually no. If you look at the model number of the Dlink box you will
notice that last letter M. This designates MGCP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Thursday, January 22, 2004 10:46 AM
To: '[EMAIL PROTECTED]'
Just upgraded from 0.7.1 to the latest CVS version yesterday. This introduced
a slew of warnings on startup. About 20 or 25 of the first, then 5 of the second:
Jan 23 10:58:49 WARNING[8201]: chan_sip.c:446 __sip_xmit: sip_xmit of 0x80ef064 (len
461) to 0.0.0.0 returned -1: Invalid argument
Actually no. If you look at the model number of the Dlink box you will
notice that last letter M. This designates MGCP.
Ok, that said - primus is pretty bad at technical support since they have
just avoided any technical questions I have had so far and given me a
canned sales response. I have
BT broadband voice uses ATA-186s configured as MGCP devices.
I think asterisk supports MGCP. I want to configure MGCP with asterisk to
connect to my BT Broadband Voice.
Do you have any idea relating to that.
Kannaiyan
___
Asterisk-Users mailing
Some of you may remember seeing my issue using SIP for incoming calls
from the PSTN:
http://voip-info.org/wiki-Asterisk+cisco+FXO
i.e. all incoming calls arrive in the default 'bogon-calls' context.
Well, I tried again using H.323 get exactly the same result (both for
chan_h323 chan_oh323)
On Thu, 22 Jan 2004, dkwok wrote:
Just got GS 101 phone and plugged into the network.
Got ip setup however, the following problems arise:
1. when dialing an extension, I cannot further send any key tone to
Asterisk.
2. there is no sound coming from the other end.
[gs]
canreinvite=no
Thanks for all your responses on this, but on and off line... I got my
questions answered, at least for now.
E.A. (Ted) Thomas, CEO
eThomasGroup.biz
Collaboration for Business
502-802-6130
__
I am a collaboration consultant doing some research for a major client who
has a
Kannaiyan Natesan said:
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
___
No, and not sure of their rates but http://www.telappliant.com/ has good
rates, voice quality and is easy to interface to Asterisk.
Robert
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870 number from http://www.speak2world.com but they
charge for it.
Kannaiyan
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 6:44 PM
Subject: Re:
Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550
each.
Their web site does not mention asterisk drivers. Is this card supported?
Any idea how it compares to a pair of external Mediatrix 1204 Sip FXO boxes?
___
Do you have to continue to use the existing handsets? You should look at
replacing the existing phones with SIP phones.
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
hi everybody...
have you checked the asterisk backports from www.backports.org? I'm
currently building my asterisk system and i think i will use these debs as
I've successfully used alot of debs from backports.org in almost every
production-server we have.
don't know the quality of the asterisk
Title: RE: [Asterisk-Users] Debian Packages and Mirrors
hi everybody...
http://www.backports.org has asterisk 0.7.1
for woody ;))
bye
- Original Message -
From:
Kostur,
Andre
To: '[EMAIL PROTECTED]'
Sent: Friday, January 23, 2004 5:45
PM
Subject: RE:
On Fri, 23 Jan 2004 15:49:31 -0300
CW_ASN - Gus [EMAIL PROTECTED] wrote:
In Brasil, Telefonica offers ISDN, but it's a diferent comercial
service (if you want voice and data in your E1), and it's more
expensive. If you only want voice, the only choice is R2.
Very weird, in Argentina the
Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO
4-port gateway?
The archives tend to suggest the box is not very straight forward, and possibly
lacks some basic pstn interaction features.
Thinking about trying one in place of a pair of x100p's (functioning fine
Kannaiyan Natesan said:
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870 number from http://www.speak2world.com but they
charge for it.
Kannaiyan
Don't think so but sometimes free isn't free. Depending on calling
patterns it might actually be lower cost
My CLEC just called and asked if we will support the MI2 protocol on
our proposed T1 circuit. I think this is for CallerID name. Will the
T100P support this?
Thanks,
Mike
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I've just noticed that if you start a call by writing a file to
/var/spool/asterisk/outgoing the cdr created on termination logs the call
placed to the local extension - not to the destination in the PSTN. Hence
there is no record of the PSTN number dialled. I guess most people want to
log
I downloaded asterisk and was trying to compile fresh, It end up in
error, Any help appreciated.
cvs checkout asterisk
cd asterisk
make clean
make
END UP with following error, (Previously I was able to compile without
any errors. After a make clean stopped compiling.)
gcc -shared -Xlinker -x
On Fri, Jan 23, 2004 at 08:45:07AM -0800, Kostur, Andre wrote:
v0.1.11 in stable, v0.5.0 in testing, v0.7.1 in unstable (unless you're not
on an i386)
Ah, I didn't realize 0.7.1 was in unstable -- I run
mostly testing here.
What do you have different in your packages?
Nothing in
It is supported. chan_vpb is the Voicetronix driver and I believe the
asterisk file that deals with this is vpb.conf. Look in this month's
mailing list archives for one user's (successful) experiences with this
card.
I have no idea about the comparison to Mediatrix.
You do have the sip phones
Hi ! thanks for the answer..
I use rh9...
I think with an interrupt problem, any startup will fail, may it be
manual or automatic during startup.
but.. you think that there is a problem in the interrupts at all? i don´t
understand.
regards
Miklos
- Original Message -
From: Karsten
Rich Adamson wrote:
Two Voicetronix Openline 4 port FXO cards would do the trick. They run about $550 each.
Their web site does not mention asterisk drivers. Is this card supported?
The drivers are for Linux. Yes the card is supported. See vpb.conf.
Any idea how it compares to a pair of
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:07 PM
Subject: Re: [Asterisk-Users] UK BT Interface with asterisk?
Kannaiyan Natesan said:
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870
On Fri, 2004-01-23 at 00:33, Martin Bene wrote:
Hi Siggi/Jan,
The Error Verifying Config Info Message doesn't have anything to do with
the real problem. I also get that message, possibly because I don't keep a
device specific config file (SEP000D65707B78.cnf.xml) or
DISTINCTIVERINGLIST.XML on
Works okay but user interface is a little like using RegEdit to program your
router.
In the version of software the one I have it lack some security features and
I am unable to find any DMTF controls
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list
Is it so hard to put X100P as a ethernet device?
I have been trying FXO devices, but gets me luck.
Kannaiyan
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 7:40 PM
Subject: [Asterisk-Users] Mediatrix
Go for inter-fone products. it can both support sip and h323.
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 2:40 PM
Subject: [Asterisk-Users] Mediatrix 1204 sip experience?
Anyone had any
Title: Troubles with the System Attendent Patch.
Dear all,
I have spent some time tying to get the system attendant patch to work;
http://bugs.digium.com/bug_view_page.php?bug_id=214
I get no errors patching the system and the function runs, but I keep getting the following error;
For people who are using GS 101, what do you think the sidetone
generated by the phone.
I find mind a bit annoying. It has a delay and you notice it as an echo.
The volume of the sidetone is also quite hight. I am distracted when
both caller and called party talking over each other
Sorry, it's the NI2 protocol. A previous poster said the N2
protocol was supported by the T400P. Is NI2 the same as N2? Does
NI2 mean switchtype=national?
Thanks
Michael Welter wrote:
My CLEC just called and asked if we will support the MI2 protocol on
our proposed T1 circuit. I think
If you are not users from mysql database then you can disable in the
makefile.
For this,
USE_MYSQL_FRIENDS=1
change it to
USE_MYSQL_FRIENDS=0
You won't get that error.
Alternatively you can install mysqlclient library to compile it without
errors.
Kannaiyan
- Original Message -
There is no CDR for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:27 PM
Subject: [Asterisk-Users] Back to front logging for calls
I was getting that error today as well.. I just checked out a new CVS
and it seems to be compiling now though. Try it again.
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Hi all,
I have
a g.723.1 file and my voice devices support this codec, I need to playback this
file in asterisk , I stored it in the directory /var/lib/asterisk/sounds/ but when I
executte the command in the extension.conf (exten =
100,1,playback(file.g7323) the call hang up, my voice
It's NI-2
Yes it does.
Alfred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Welter
Sent: Friday, January 23, 2004 12:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MI2
My CLEC just called and asked if we will support the MI2 protocol on
UI for switch config allows you to generate scripts for setting if you need
them. I found that to be useful. They can be easily configured from remote
if you have the UI software. There are features for caller id, but I have
not used them yet.
Wes
-Original Message-
From: [EMAIL
I'm not sure I understand your english here. I have two x100p's working just fine,
but I've got a couple more pstn lines I'd like to connect up. I probably could
put another one in the system, but I'd rather use a 4-port external gateway that
works well if such a thing exits at a reasonable price.
Hi
I have updated from CVS about a week ago and got the externip working
with FWD for outbound calls., but I'm having problems with inbound
calls, I don't think they are even reaching the Asterisk box even though
I have forwaorded 5060 and the rtp range specified, another thing I have
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