On 23 Mar 2006, at 23:48, Mike Dent wrote:
Hi,
which OSX softphone do you use that supports IAX2 protocol with
Asterisk?
I like LoudHush a lot:
http://www.loudhush.ro/
It is a very simple client, but looks great and works well. My only
complaint is that the ring tone it generates when
On 3/24/06, Alyed Tzompa [EMAIL PROTECTED] wrote:
Think a zaptel recompile is just what you need.
Alyed
i've tried but i get some error when the module wtc2xx is loaded...
maybe i've got to rebuild libpri?
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Any reason whz additional classess are necessary for AstTapi? How to
make that secure? ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Tichy
Sent: Wednesday, March 22, 2006 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Ganbold Tsagaankhuu wrote:
Hello,
I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.
My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN
boldsoft*CLI
Hi all
I want to use conference in Asterisk. I configure a
conference room in meetme.conf (as conf = 600,1234)
and extensions.conf as (exten =
600,1,MeetMe(600,i,1234)) . When i call the extension
600, i have the following message in the asterisk
logs:
WARNING[7758]: pbx.c:1688
Hi all I wonder if anyone can give a little insight into this;
[EMAIL PROTECTED] 2.2, HP Proliant gl3, sangoma A101,Cable and Wireless,
ISDN30. 10 zactive channels
Incoming calls work fine no problems tested 43 DIDs all working.
zaptel.conf
# Global data
loadzone= uk
defaultzone
You forgot need and please
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Hi,
since we do not have a nice common numbering plan like (XXX) XXX for
national phone numbers here in Germany, the dialplans usually contain lines
like this
exten = _0X.,1,NoOp(Dial outwards etc.)
If you use such context with overlap dial (DISA, ZAP), it takes a while for
Asterisk to
Hi,
I have unpleasent short audio gaps when a
perl based agi scripts starts.
Thus, I now started to put all those things in C programmed
daemons for fast-agi.
Anyway I'm looking for another mean, which would help me
more quickly.
I noticed, that all agi scripts are running with system
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if
I
hang up during Playing 'conf-onlyperson' my machine freezes. It's not a GSM
Enconding problem as I suspected first, this happens with every encoding.
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
conf-onlyperson. This has been seen
*CLI show version
Asterisk 1.2.1 built by root @ pbx on a i686 running Linux on 2006-02-02
09:34:1 6 UTC
Hmm, so maybe a * 1.2.5 bug?
P.S.
I see that you are using language de, maybe you should look at that
direction...
Nope, Brent confirmed it also happens with his installation with
On Mar 23, 2006, at 3:48 PM, Mike Dent wrote:
Hi,
which OSX softphone do you use that supports IAX2 protocol with
Asterisk?
There is a new one called JackenIAX that is working stunningly well for
me. It's still beta, but it's way better then Iaxcomm.
Marty
Mike Dent wrote:
Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?
Idefisk from asteriskguru.com works very well.
--
Benoit Merouze
Network Software Developer
[EMAIL PROTECTED]
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i would suggest Astribank-8 of XorCom. it is a dedicated Asterisk compliant solution.
you can look at : www.xorcom.com.
On 3/24/06, Curt Shaffer [EMAIL PROTECTED] wrote:
Is anyone out there using FXS channel banks to connect analog phones to Asterisk? If so do you have brand recommendations?
Hi there!
Which 2 Port ISDN Card for P2P do you recommend?
Regards,
Marcus
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|** realität ist da wo der pizzamann herkommt **|
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Hello,I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands)From the moment i switched all inbound calls are terminated after aproximatly 1 minute.The provider tells me it's not their issue sinceI have no other configuration than all their other users.What can I do.I
On Fri, March 24, 2006 12:01, Asterisk said:
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no
AK == Andrew Kohlsmith [EMAIL PROTECTED] writes:
AK There is no mechanism in place for the DB to tell Asterisk that a
AK row changed and that the cache is invalid. If you are using the
AK cache in Asterisk you must manually clear out the peer entry to
AK get the new value, or simply wait for the
On Thursday 23 March 2006 21:14, stoffell wrote:
On 3/23/06, Henning Holtschneider [EMAIL PROTECTED] wrote:
I've got Asterisk 1.2.4 running with two Junghanns QuadBRI cards using
the qozap driver from bristuff 0.3.0-PRE-1l. One of the cards is running
in TE mode, the other one in NT mode.
probably a DTMF issue. Try changing it. Font have the
link here. Go to voip-info.org and search for DTMF
type.
--- Mohammad Salaque [EMAIL PROTECTED] wrote:
Dear List,
I am facing another strange problem . some of my
envisions like to use
other prepaid card (whatever they found in market)
Nope,It's not a firewall problem.I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.It replaces the inside IP adresses from the * server in the SIP frames by the outside IP adress and creates pinholes for the udp streams.I have severalSIP connections (SIPphone, SIPGate, IPtel,
because your phone is prob. set to a diffrent context
--- Larry Alkoff [EMAIL PROTECTED] wrote:
Luigi Rizzo wrote:
On Thu, Mar 23, 2006 at 01:18:15PM -0600, Larry
Alkoff wrote:
It _appears_ that the only way to create valid
[context] is by a
context = line in sip.conf.
Is there
Join the club. I just put a block on my cc. They
terminated me pretty fast :)
--- Ronald Lewis [EMAIL PROTECTED] wrote:
After months of BroadVoice ignoring my trouble
tickets for dropped calls,
delayed termination, etc., I'm throwing in the
towel. While they have
credited $19.95 to my
Hi !
What is now the difference between a:
reload - (cli command reload).
restart - (I assume the application asterisk is restarted. o.k starting
from new)
sip reload - (cli command sip reload). Is sip reload part of the
reload command ?
Please confirm:
Which is the correct command when
snip
They have both worked reliably for me, although
frickin' comcast
sometimes has very poor latency, which they admit,
but fail to do
anything about.
/snip
From what I know DSL is more reliable when it comes to
VOIP. Depending on hoy many channels you use you can
get a basic DSL line. If
That's it. 'sip reload', reloads the sip.conf, 'extension reload' (or
extensions?), reloads the extension.conf and so on. The 'reload' command do
it all at once.
Regards,
Filipe Mordhorst
Joinville - SC - Brasil
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I don't think there's any kind of (significantly small, anyway) limit. I
have over 300 users at one site in voicemail.conf and no issues there.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, March 22, 2006
Thanks Sean!
Fortunately, I don't think I will have to worry about passing extensions back and forth between Asterisk and the Legend. But, I'm glad to know that it is possible, if the situation arises. We don't have that many users. My concern is just making sure that the two can coexist for a
RS == Roger Schreiter [EMAIL PROTECTED] writes:
RS Is there any mean to let AGI scripts run in a lower priority
RS (except starting a new shell from the a short initial AGI script)?
You can start the script with renice 15 $$, or whichever value you
prefer. If the hickup happens because of the
I have search wiki, asteriskguru, chan_sccp and some other site's for
information's how to upgrade, and make Cisco 7970 IP phone to work with
asterisk on SCCP firmware.
I'm sure that there are users on this group that have working Cisco 7970 phone.
Please send me some information's how to do
On Mar 23, 2006, at 11:05 AM, Ronald Lewis wrote:
After months of BroadVoice ignoring my trouble tickets for dropped
calls, delayed termination, etc., I'm throwing in the towel. While
they have credited $19.95 to my account, they refuse to credit
anything more, despite ALL of the problems
On 3/24/06, serge messa [EMAIL PROTECTED] wrote:
Hi all
I want to use conference in Asterisk. I configure a
conference room in meetme.conf (as conf = 600,1234)
and extensions.conf as (exten =
600,1,MeetMe(600,i,1234)) . When i call the extension
600, i have the following message in the
I am trying get true ANI from my provider into asterisk. I have found a few
patches that supposedly accomplish this on Mantis and were committed to CVS
sometime mid last year.
My question is, would this have been included in stable 1.2.5 or do have to
patch it? There is very little info on
I'm sure the question about who uses Asterisk comes up a lot, but to me, that's something important to help the adoption of Asterisk. It's nice to see others that are using it. It always helpswhen you are presenting it to clients if you can let them know who else uses it. And, Aaron, to me, seeing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Aaron,
I have this working quite well. Are you using FTP? or TFTP...
We are using FTP for about 40 phones and it works like a champ. For
each phone I have...
0004f2030925.cfg
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone4710.cfg,
sip.cfg
Tele Cost Price Reducer wrote:
i would suggest Astribank-8 of XorCom. it is a dedicated
Asterisk compliant solution.
you can look at : www.xorcom.com.
Looks interesting, shame they don't have a FXO version.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am currently running asterisk 1.0.9 on a system with 2 TDM400P... I
have had fairly good success with it across the board... my only issue
is that I have monkeys who move stuff around and things get unplugged ;)
Jared Davison wrote:
I would
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
if you have an zaptel card installed and working... try to do a load
app_meetme.so and see what happens... if it loads successfully... you
should be able to conference also check your modules.conf and make
sure you don't have noload=app_meetme.so
On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote:
I have here de backtrace result
Using host libthread_db library /lib/libthread_db.so.1.
Core was generated by `asterisk -g'.
Program terminated with signal 11, Segmentation fault.
#0 0xb7ece142 in ?? ()
As I see it was in
Jared Davison wrote:
I would like to hear from anyone good or bad as what their experience has
been in recent times with STABILITY of current builds of Asterisk and
drivers for TDM400P.
The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards.
I am not concerned with: price
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Hash: SHA1
Yeah... we went through the same thing... our problem was working with
asterisk it was Oh cool! then trying to make the legend work was WTF!.
Our setup is on the back side... legend still connects via PRI to the
PSTN, but we have asterisk running as
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Do hints work in Realtime asterisk? not finding much on the list
archives or anywhere else for that matter... I have tried using -1
priority as mentioned once or twice but no joy
Thought?
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.2
Hrmm.. I put that in my iax.conf... reloaded.. and I still got:
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6 is ringing
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6
stopped sounds
Mar 24 08:25:30 VERBOSE[18185] logger.c: -- IAX2/calleveryone-6
I have compiled zaptel on Mandrake following everything I
have always done on Fedora.
It is 2.6 udev so
I had to modify the 01-devfs.rules
Make linux26
Make
Make install
Everything appears to compile correctly but it says module
not found when doing modprobe zaptel
Is this a rights
Hi,
VoIP in the Philippines can be done, HOWEVER, you will be buying your E1R2
from either Globe, Bayantel, PLDT, or Digitel. You have no other choice,
most of the time you will only have 1 carrier serving the area, Philippines
has been subdivided into what is called as congressional franchise
is there a number is the U.S that you can dial where a computer will
reply with the phone number your calling from.
carrier is sbc if that makes any difference.
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On Fri, 2006-03-24 at 07:37 -0600, Jordan Novak wrote:
I have compiled zaptel on Mandrake following everything I have always
done on Fedora.
It is 2.6 udev so…
I had to modify the 01-devfs.rules
Make linux26
Isn't needed anymore if it's a recent zaptel
just 'make' is all that's
Jordan Novak wrote:
I have compiled zaptel on Mandrake following everything I have always
done on Fedora.
It is 2.6 udev so…
I had to modify the 01-devfs.rules
Make linux26
Make
Make install…
Everything appears to compile correctly but it says module not found
when doing “modprobe
call waiting must be set to enabled for this extension.
I had to force the issue in AAH by doing this:
database put CW 38 ENABLED
where 38 was the extension with the issue. This forces call waiting to be
enabled respective of the AAH GUI.
-Original Message-
From: [EMAIL PROTECTED]
SURE!
800-444-
804-883-2001
800-437-7950
866-692-6447
:)
On 3/24/06, mike webb [EMAIL PROTECTED] wrote:
is there a number is the U.S that you can dial where a computer will
reply with the phone number your calling from.
carrier is sbc if that makes any difference.
On 3/24/06, mustardman29 [EMAIL PROTECTED] wrote:
So your Polycom 501's will eventually re-subscribe and BLF will eventually
start working again after a reboot using your patch? How long will that
take? Is the time to re-subscribe something you can set on the phone?
That would be quite
i have 6 pots lines coming in from the outside world (but we are
reducing to 4)
all the lines have the same phone number.
i have 40 analog telephones that need to be connected to them.
one way (i think) i could do this is to have a asterisk box with a
tdm400p with 4 fxo's connected to the pots,
I am using ParkAndAnnounce to Park the call and explicitly retrieving using ParkedCall app in the dial plan. I am trying to guess the parking lot being used in a particular call by incrementing a counter just before the ParkAndAnnounce and decrement the counter just before the ParkedCall. I am not
Hi,
I'm looking for a web GUI to offer my end-users (Hosted PBX), and I thought
I'd pick a few brains first.
I'm not looking to configure the Asterisk server itself, VI works adequately
for that. But I want to give Web access to as many of the following
features:
1) Voicemail configs: NIP,
I tried many different combination of nofaststart, noh245tunneling and no
success.
Balgaa
- Original Message -
From: yusuf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, March 24, 2006 4:27 PM
Subject: Re:
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http://bugs.digium.com/view.php?id=5884
Haven't tried it out yet.
Benoit Panizzon
--
I m p r o W a r e A G-System Services
__
Zurlindenstrasse 29 Tel
I installed as su, and tried to compile using only make. No
problems were reported during compiling but problem persists. Any other ideas?
Jordan Novak
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Hi -
I have an fairly vanilla answering machine (actually its a combo
cordless phone/answering machine) attached to an FXS port (on a TDM400
Card).
Everything is working as planned except I seem to be having a bit of
trouble with the answering machine. After is answers it plays the
outgoing
Ouch, Come on! One must know.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wai
WuSent: Thursday, March 23, 2006 4:25 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Changing
codec.
Hi,
Is there a way to
tell Asterisk to change the codec being used in
We've been running on an ICS7750 for almost 4 years. It's ridiculously
expensive. We've looked at the cost of setting up call center like
features, call recording etc and it boils down to a forklift upgrade
that will end up costing anywhere from $100-$250K. That's partially
our problem, since we
You could use contexts for this. By default put everyone into the
'internal' context. Managers would go into the 'managers' context,
which would include the 'internal' context.
The manager context specifically would have the exten's to monitor or
barge into calls. By including the internal
[EMAIL PROTECTED] is believed to have said:
Hi,
which OSX softphone do you use that supports IAX2 protocol with Asterisk?
thanks
Mike
Hi Mike,
look for LoudHush on VersionTracker...
HTH
Aldo
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thanks Dovid ,
i solved that yes it was DTMF issue .
thanks
Salaque
On 3/24/06, Dovid Bender [EMAIL PROTECTED] wrote:
probably a DTMF issue. Try changing it. Font have the
link here. Go to voip-info.org and search for DTMF
type.
--- Mohammad Salaque [EMAIL PROTECTED] wrote:
Dear List,
check your /lib/modules for a custom kernel, copy it over to your
current kernel..
On 3/24/06, Jordan Novak [EMAIL PROTECTED] wrote:
I installed as su, and tried to compile using only make. No problems were
reported during compiling but problem persists. Any other ideas?
Jordan Novak
Hello,
I use Mandrake 10.1 and I had no problem, just
had to install the kernel sources and follow the
instructions in README.udev
I do make linux26.
you have to reboot after you follow the instructions in REAME.udev
so it can take effect. Make sure zapata.conf is in /etc and check
for
You will probably have to build that yourself, or really customize
something off the shelf. Depending on what phones you are using you
might be able to do that via the phones xml interface.
Have fun with that I would be interested to see how it goes.
--
Justin
-Original Message-
From:
Yeah, that's kinda what I've got set up in mine:
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=44198/phone.cfg,
sip.cfg MISC_FILES= LOG_FILE_DIRECTORY=44198
OVERRIDES_DIRECTORY=44198 CONTACTS_DIRECTORY=44198/
It's pulling 44198/phone.cfg from the server fine, but for some reason
it's not
I am using Asterisk 1.2.5 and when ever I dial to either another extension
or to an outside number, I seem to be experiencing a really bad echo
problem. The echo is so bad, that Asterisk is almost unusable.
I am using VoIPJet as my outgoing IAX provider and do not use any Zaptel
hardware.
I do know what you're saying about the need to know who all uses Asterisk
is important. It may not be a bad idea to get a definitive list together
at some point in the future so that anyone that's trying to get approval
can sit down and show a list of successful deployments. Guess in time
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call
Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/
1.) setup your /etc/asterisk/sccp.conf with something like:
[devices]
type= 7970 ; device type (see below)
autologin = 30,31, ; lines list. You can add an empty line for an
empty button (7960,
On Thursday 23 March 2006 22:14, BJ Weschke wrote:
There's been two very recent commits (one less than an hour ago) that
may very well correct your issues.
The patch at http://bugs.digium.com/view.php?id=5884 fixes the problem!
Cheers,
Henning Holtschneider
--
LocaNet oHG -
Anyone have a Snom they're happy with? How did you manage that? :)
I have a system of:
Asterisk 1.2.3
2 Wildcard TDM400P Rev I and E/F
1 Snom 360 + sidecar
~15 Sipura/Linsys SPA-841
~15 Grandstream 101
Everything (currently) is on the same network, not a router to be seen
between any
I have an fairly vanilla answering machine (actually its a combo
cordless phone/answering machine) attached to an FXS port (on a TDM400
Card).
Everything is working as planned except I seem to be having a bit of
trouble with the answering machine. After is answers it plays the
outgoing
Perhaps a page on the wiki would work? We could set the ground rules
similar to other industries: no names, nothing more defining than a
region, the number of units, etc. Would that be useful?
For example, I can describe this organization as a security company in
Southwest Missouri using
Do queue names have a max length? I have a queue named 'oneeighty_techsupp' in
queues.conf, and a 'show queues' returns a truncated queue name. Is that just a
display bug, or do queues names have a max length of 12?
demeter*CLI show queues
oneeighty_te has 0 calls (max unlimited) in 'rrmemory'
Perhaps a page on the wiki would work? We could set the ground rules
similar to other industries: no names, nothing more defining than a
region, the number of units, etc. Would that be useful?
For example, I can describe this organization as a security company in
Southwest Missouri using
[EMAIL PROTECTED] named]# grep ANI /home/software/asterisk/asterisk-1.2/doc/*
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLERANI}
* Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLINGANI2}
Hello all, I first want to thank everyone for all your
contributions. Ive building an asterisk system for a month or so now and
without everyone in the online asterisk community I wouldnt have made it
this far yet. Thanks! ok, mushiness out of the way.. :)
I am looking for a failover
On 3/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Do queue names have a max length? I have a queue named 'oneeighty_techsupp'
in queues.conf, and a 'show queues' returns a truncated queue name. Is that
just a display bug, or do queues names have a max length of 12?
demeter*CLI show
Anyone have a Snom they're happy with? How did you manage that? :)
I have a system of:
Asterisk 1.2.3
2 Wildcard TDM400P Rev I and E/F
1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841
~15 Grandstream 101
Everything (currently) is on the same network, not a router to be seen
between any
I know the answer. The answer is NO! Asterisk does not support
changing the codec during a call. It also does not support changing the
codec on an INCOMING call. Of course, as you know by reading
README.variables, SIP_CODEC can force a specific codec on an OUTGOING call.
Wai Wu wrote:
Matt wrote:
SURE!
800-444-
804-883-2001
800-437-7950
866-692-6447
:)
On 3/24/06, mike webb [EMAIL PROTECTED] wrote:
is there a number is the U.S that you can dial where a computer will
reply with the phone number your calling from.
carrier is sbc if that makes any difference.
Pretty much
setpriority(0, 0, 20);
This is for Perl, of course.
Benny Amorsen wrote:
RS == Roger Schreiter [EMAIL PROTECTED] writes:
RS Is there any mean to let AGI scripts run in a lower priority
RS (except starting a new shell from the a short initial AGI script)?
You can start the script with renice
The only reason I recommended that was to protect the privacy of those
on that list. I personally do not want a bunch of cold calls from
asterisk 'dealers' just because I chose to implement that product. Such
a list of users would make a tempting target for marketing uses...
But either way, a
For one thing, don't use the r option to dial. It can hide major
problems. If you don't hear ringing without using r then you have
massive problems.
Asterisk wrote:
Nope,
It's not a firewall problem.
I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.
It replaces the inside IP
Go through the archives (or your own inbox) for a very, very thorough
set of conversations that just passed this way only a week or two ago.
There are a few key people working on this type of 'HA' solution and
they're pretty close to making it work. They have already identified
the key issues
Well, I should say Sporadically I
can register to the virtual ip. Other times I cant.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan Mahin
Sent: Friday, March 24, 2006 12:56
PM
To: Asterisk-Users
Subject: [Asterisk-Users] Asterisk
Failover without SER
You can do a version of failover with phones that support a
backup registrar. They will repoint themselves to a second server
then.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan
MahinSent: Friday, March 24, 2006 9:56 AMTo:
Asterisk-UsersSubject: [Asterisk-Users]
Hi Brian,
For the conf on Xfer issue, use the latest beta
http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin
Regards,
-
Usman Tahir
snom technology AG
Gradestraße 46
D-12347 Berlin.
Tel: +49 30 398330
Fax: +49 30
Hi,
I am looking at a project which requires VoIP QoS monitoring and failover
re-routing to PSTN, without dropping the call ideally. While I have hardware
solutions available such as the Quintum Tenor series, I see no reason why
Asterisk can't have this feature with some effort obviously. Plus I
You wouldn't have to post anonymously -- only if it makes you feel better.
I could have really used such a resource in January -- Digium's list
of success stories is a little thin.
On 3/24/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Perhaps a page on the wiki would work? We could set the ground
In short, does this work yet?
ie putting agents into Realtime. Can't find any info on it...
Thanks,
Doug.
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I see your reasoning on that one. Perhaps it is best to leave the
information about the asterisk-users list so they can contact us. I'm
just worried about the idea that people aren't going to want to subscribe
to a 4000 message a month list just to find out about a system.
Especially with
In my queue, I have defined a periodic announcement with a message that goes
something like if you would like to leave a voice message now, please press
1 However, when a user presses 1 *during* the message, the playback stops
and the user still remains in the queue (listening to music on
Anyone have a Snom they're happy with? How did you manage that? :)
I have a system of:
Asterisk 1.2.3
2 Wildcard TDM400P Rev I and E/F
1 Snom 360 + sidecar
~15 Sipura/Linsys SPA-841
~15 Grandstream 101
Everything (currently) is on the same network, not a router to be seen
We've actually got two servers handling all the call volume, and when one
server goes down, the other one fields all phone calls. We're using a
combination of dialplan magic and dns to make it work. As long as your
phones can handle multiple ip's for host records and you've got the
dialplan
Hi,
I want
my users to be able to get into VoiceMailMain when they press * while listening
to their own greeting. It`s standard operating procedure with most
voicemails I have ever used,and luckily it seems Asterisk can support this
behaviorwith the "a" extension.
The
only thing, is even
I tried to get a government/enterprise SIG or UG off the ground a
number of months ago, with very limited success. If there is sufficient
interest now, I could be persuaded to try again.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
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