Cory Andrews wrote:
To get an accurate portrayal of defect rate, a very large sample size
will obviously result in a more accurate calculation. I calculated a
defect rate of between 1-2% for Digium products, based on an arbitrary
sample size of 5000 units. These included “ALL” Digium products
I upgraded from 1.6.2 to 1.6.6. After which, the problems started to
happen. While it isn't a good thing, at least I'm not crazy and someone
else is having the problem as well. ;).
I also turned on the logger on asterisk with full debug information.
Sure it's crazy, but maybe if one of the pho
Hi,I have AudioCodes MP-124 device. It works fine, but I want to fix one problem, but I don't know how! In AudioCodes I've created two lines for tests:52040668 and 52040669. Here is my sip.conf fragment sniffed traffic:
sip.conf[52040668]type=friendaccountcode=1videosupport=yescontext=bandymassecre
anyone have information on how the call back features work with asterisk? I means the dial plan or what so ever. thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
I want to set some variables for each phone. For that I use setvar in
Real-time. At the beginning of each context should be this include
statement before all other include statements.
How can I rewrite the dial plan, so that after the include var-key other
include statements are still used?
No log entries yet that might show whats happening and you are correct, I
cant run under strace as it would hit performance quite bad.
:( I will continue to look into the logs and hope something will show up so
I can post further.. If anybody else experiencing this can come up with some
log entry,
M.Hockings wrote:
One weakness is the incoming PSTN line, what is the best way to
protect that beyond the device at the premises entry ?
The device at the entry, assuming it was even installed correctly, is
there to protect the PSTN Central Office, NOT your equipment.
There are many sites o
yeah this post is old and there have been dozens of replies, but here's
some feedback for the list, now that i have some.
we're using a sangoma a102 card (no hw ec) with 2 pris from sbc.
asterisk 1.2.7.1, zaptel 1.2.6 (much testing previously with 1.2.5). we
first used:
KB1 (not aggressiv
I have quite the same strange behaviour in Dial(...M(x^y) )
I use it to play various annoucements and operates differents operations
to agents that answers and when the dial timeout reachs to its end the
macro is hangup although the DIALSTATUS is set to ANSWER
I'll post some logs and part
In that case, it is likely your reseller is not a "Polycom certified VoIP
reseller". Contact me off-list and I'll help you.
Nabeel Jafferali
www.voipdepot.ca
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Lucas Alvarez
> Sent: June 30, 2006 6:
Ohoh... Kevin, what version of SIP software are you running?
One of my Polycom phones just rebooted itself for no apparent reason.
> -Original Message-
> From: Kevin Smith [mailto:[EMAIL PROTECTED]
> Sent: Friday, June 30, 2006 1:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Dis
Does someone know where I can get the last sip version? My Polycom
reseller doesn't have it and I need to enable the buddy for 14 contacts.
Thanks in advance.
Lucas Alvarez
Douglas Garstang wrote:
I've never seen that problem, and I've only ever used 1.2+ with
Polycom and buddies.
---
Does anyone know how
to successfully register a Motorola VT1005? What firmware should be
used?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/
I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this
specifically
to get the Dial M(x^y) feature so that I could implement call completion
confirmation over IAX2 channels (not available in 1.0.7). The problem is that
the
call is always completed--even without the required user
Hi All.
I am testing asterisk-1.2.9.1 linked with one pabx Siemens HiPath 4000 in protocol QSIG.
I noticed that when the occupation is linearly orderly increasing, or either initiating for the first circuit, the call is completed successfully. But when the occupation is disordered and initiating it
I found my mistake jiterbuffer=yes vs. jitterbuffer=yes ;-)
currently I have this settings, and seems this working quite well,
only sometimes gaps appears, when jitter changes too much eg. >500ms -
jitterbuffer probably can't adapt so quick,
maybe good idea to set some minimum jitterbuffer value
For the typical home user who has a SIP ATA behind (usually) a Linksys home router/firewall, what's the best practice "qualify=" time we should be running on the server, to keep the home user's NAT happy?
The default, 2 seconds, is way too short (generates too much net traffic).
I am wonderin
Ah, the problem is that you are connecting FXO to FXO. The IAXy provides
dialtone and o does your Intercom system. You can try to use an FXO to
FXS converter or simply replace it with an FXO adapter.
I would also check the documentation on your intercom device. There may
be a way to switch the po
To get an accurate portrayal of defect
rate, a very large sample size will obviously result in a more accurate
calculation. I calculated a defect rate of between 1-2% for Digium
products, based on an arbitrary sample size of 5000 units. These included
“ALL” Digium products, not just TDM p
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels
that an option 'a' is available meaning 'request autoanswer'. Never
tested this before, so please do.>
Another possibility might be setting immediate=yes in iax.conf for the
iaxy? just a guess.
Moj
I tried both
I get the chan_zap.so if I recompile under
asterisk-1.2.7.1, but not under subversion TRUNK
Anyone able to do this?
- Original Message -
From:
Aaron Paxson
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, June 30, 2006 1:44 PM
Subject: [
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It would not be the iaxy... it would be the phone that is attached to
it... there are plenty of phones/answering machines /other FXS
signalling devices that can do auto answer... the iaxy is not capable
of doing that...
Sean
Jerry Geis wrote:
> Can
I see on http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels
that an option 'a' is available meaning 'request autoanswer'. Never
tested this before, so please do.
Another possibility might be setting immediate=yes in iax.conf for the
iaxy? just a guess.
Moj
Jerry Geis wrote
I’m working on quantifying an
overall defect rate for both Digium and Sangoma products, based upon overall
number of units deployed over a 12 month period versus overall number of units
RMA replaced. I believe both products to have very low DOA rates, well below
acceptable industry standar
Jon Scottorn wrote:
What kind of line is being used?
in zapata.conf:
group = 1
channel => 1,3,5,6
I create a zap group will all your lines and dial out using the zap
group ie...
Dial(Zap/g1/${EXTEN})
By using the group it dials on the first available line.
If you want a more com
thanks Dough, seems, that you mix options for old and new jitterbuffer
implementation (according to iax.conf.sample),
I think, that now is by default in compile time selected "new
jitterbuffer", so only these four options are in efect and rest are
ignored
PJ
new jitterbuffer options:
jitte
Khaled Chehab wrote:
I am using trixbox,I want ot disable and enable voicemail from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully
But at trixbox its not working
Any ideas pleas
Did you try checking with the people who _wrote_ trixbox? P
Hey Doug,
That's what I figured, but correct me if I am wrong. Isn't 1 will always
set the phones to reboot on a NOTIFY command regardless of any changes
in the configuration file? I thought 0 would means it requires both a
notify request and a change in the configuration file.
But you are r
Can an IAXY be setup to auto answer? If so how?
I mean any call coming into it automatically connect it to the phone and
send voice traffic.
Thanks,
Jerry
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSU
Andrew,you seem to be assuming a lot. These were spread out across different parts of the country (US), on projects I was involved with but deployed by more than compentent telco and engineering colleagues of mine. And ... in the majority of the cases, they were DOA (not a transient issue, noisy li
I would work that out with your vendor, as the
settings must be the same on both sides.
If national won't work for you, ask them if they
can change to something else.
What kinds of connectivity issues? Could be
line problems too.
- Original Message -
From:
James Hawks
The following command on the Asterisk console will reboot a polycom phone:
sip notify polycom-check-cfg
but in sip.conf, voIpProt.SIP.specialEvent.checkSync.alwaysReboot needs to
be set to 1.
otherwise... beats the heck out of me!
> -Original Message-
> From: Kevin Smith [mailto:[
M.Hockings wrote:
Mike (totally UNimpressed with Digium)
Point taken. I was not so much point fingers but asking what my
expectation should be and maybe shedding some frustration. I don't
really have a lot of experience with this kind of communications gear
All the more reason for
Michael Sampson wrote:
Basically I will have a call come in a PRI trunk and be routed out the
same PRI trunk. The point of this is so I can use asterisk to record the
call. Has anyone set up a system like this? I know how to get asterisk
to record a call from and extension, but not a call that
Hey everyone,
I wrote in last week about our Polycom phones rebooting. I had a nice
theory with it being the PoE switch but that was thrown out the window
today when phones even with a power supply rebooted.
So my question now points back to Asterisk. Is there any feature on
Asterisk that se
Our PRI vendor is using a Nortel DMS500 switch. Which switch
type should I use. I have been using national but we are having issues with our
connectivity.
national
dms100
4ess
5ess
euroisdn
ni1
qsig
Thank You
James Hawks
___
Andrew Kohlsmith wrote:
On Thursday 29 June 2006 21:38, M.Hockings wrote:
How reliable is Digium hardware in general.? My new TDM400P just died.
I have a number of Digium T1 products (T100P, TE410P, TE405P and TE406P) as
well as a few TDM400 based boards. No failures in the last 2 years or
Hi All. Somebody already caught the messages below?
-- Executing Dial("SIP/3347-9360", "zap/g1/3384|60") in new stack -- Requested transfer capability: 0x00 - SPEECH
-- Called g1/3384 -- Zap/1-1 is proceeding passing it to SIP/3347-9360 -- Zap/1-1 is ringing!! Not yet handling pre
Hey list!
I keep getting the error:
"Unable to create channel of type 'Zap' (cause 66 -
Channel not implemented)" error.
In looking on my filesystem, I seemed to have
"lost" the chan_zap.so module from /usr/lib/asterisk/modules. I've
re-compiled Zaptel and Asterisk, but it doesn't sh
> what brand of gsm gateway do you think works well with asterisk?
voismart.it - quadgsm
--
WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
___
--Bandwidth and Colocation provided by Easynews.com --
A
Email me off list with the phone part numbers, and I'll see what I can do.. It
probably depends on the level of cisco certification the company has. I dont
know if we can do better, but I'll see!
Steve
[EMAIL PROTECTED]
From: Louis-David Mitterrand [mailto:[
This really looks like a bug. It seems as though the '-x' option is broken as
of 1.2.9.1
Sometimes the output of the -x command will be only a single line:
hestia:(pbx1)~ # asterisk -rx 'database show'
//Agents/80014054 : [EMAIL PROTECTED];80014054
and sometimes
On Fri, 30 Jun 2006, francesco giuliani wrote:
> Armin Schindler wrote:
>
> > On Fri, 30 Jun 2006, Marco Mouta wrote:
> >
> >
> > > Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will
> > > be busy if you have already 2 calls running, so the caller party
> > > should get busy
I'm intensely curious why it doesn't currently work.
You have multiple Asterisk systems, all referring to a common table for SIP
peer information.
The fact that there is multiple Asterisk systems accessing the same MySQL data
should be completely transparent to each of them, and I don't understa
When will Digium include the octasic on the TDM2400P? And maybe the
TDM400P?
Also how does the TE415P and TE420P differ from the TE412P card?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or upd
On Fri, 2006-06-30 at 10:39 +0200, Roger Schreiter wrote:
> Hoa Thai Duy schrieb:
> > Roger
> >
> > If transcoding happened (from G.729 to GSM, any to any) then Asterisk will
> > issue no re-INVITE, for sure.
> >
> > Pls. change
> >
> > Disallow=all
> > Allow=gsm (only one codec)
>
>
> Hi,
>
Basically I will have a call come in a PRI trunk and be routed out the
same PRI trunk. The point of this is so I can use asterisk to record the
call. Has anyone set up a system like this? I know how to get asterisk
to record a call from and extension, but not a call that is just
"passing throug
Armin Schindler wrote:
On Fri, 30 Jun 2006, Marco Mouta wrote:
Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will
be busy if you have already 2 calls running, so the caller party
should get busy indication from your Telco...
No, the third call is signaled as call-w
Doug,
If you'd be willing to share the patch and AGI, I would be happy to
help test your solution. I know that myself and several others have
been looking for a way to make Asterisk do this for quite some time.
regards,
David
On 6/29/06, Doug G <[EMAIL PROTECTED]> wrote:
Well, to dial a peer d
Rich Adamson wrote:
I've tested a large number of other external adapters and have not
found a single one that had a reasonable echo canceller built in. Many
of them work fine on short pstn lines (where echo is much less of a
problem), but provided even reasonable service on longer pstn lines o
On Fri, 30 Jun 2006, Marco Mouta wrote:
> Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will
> be busy if you have already 2 calls running, so the caller party
> should get busy indication from your Telco...
No, the third call is signaled as call-waiting without attached to
a
I assume that it would be 30 licenses, so you could fully use the card as E1.
Is this correct?
Can asterisk use these licenses for other calls as well? (sip G.729 to
voicemail)
--
--
Steven
http://www.glimasoutheast.org
"Matthew Fredrickson" <[EMAIL PROTECTED]> wrote in message news:[EMAIL
Incoming you mean arrivin from a SIP trunk or from ISDN? BRI card will
be busy if you have already 2 calls running, so the caller party
should get busy indication from your Telco...
On 6/30/06, francesco giuliani <[EMAIL PROTECTED]> wrote:
Marco Mouta wrote:
> You should handle correctly Dial(.
Hi All,
Also check that TDM400 not share interrups (yes, it sounds silly, but in
some cases it were the answer for me).
Fabio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Matthew
Fredrickson
Enviado el: Jueves, 29 de Junio de 2006 09:41 p.m.
Para: Asterisk
On Jun 27, 2006, at 4:25 AM, Rob Lith wrote:
On 25/06/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED]> wrote:
Neither. It's a separate device, entirely unrelated to any TDM cards
(which means it can be used for any type of channel, not just TDM).
The final specs for the n
Marco Mouta wrote:
You should handle correctly Dial(...) return value in your dial plan,
then playback(your busy channel msg) and then dial through IAX or SIP
or whatever you want.
If you use Freepbx would be easy to learn how to write your Dialplan
Script...
On 6/30/06, francesco giuliani
Tommaso Calosi wrote:
Who knows something interesting about the new BRI digium card b410p ?
For example, will it use the misdn driver or the native zaptel? Any
interesting links will be appreciated too.
___
--Bandwidth and Colocation provided by Easyn
You should handle correctly Dial(...) return value in your dial plan,
then playback(your busy channel msg) and then dial through IAX or SIP
or whatever you want.
If you use Freepbx would be easy to learn how to write your Dialplan Script...
On 6/30/06, francesco giuliani <[EMAIL PROTECTED]> wr
I've contact Digium, and they told me they were finalizing the driver
and so on. And all the info would soon be posted at digium's website.
In fact it was supposed to be ready one week ago... At least they told me that.
On 6/30/06, Tommaso Calosi <[EMAIL PROTECTED]> wrote:
Who knows something i
== Spawn extension (intqueue, 1004, 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
Jun 30 15:18:34 WARNING[13523]: channel.c:787 channel_find_locked:
Avoided initial deadlock for '0x81fe3f8', 10 retries!
-- Stopped music on hold on Zap/2-1
___
On a zap channel, Asterisk can't tell when a call has been answered, so
starts the playback immediately. Setup a loop asking the caller to press
a key. I have the following setup:
[..]
I'm still wondering how to do it and I thought about BackgroundDetect(). Is
there any way to use it to dete
Because probably the rows/table/database name
changed.
Connect to you mysql database and find what records you have
to modify.
m.
- Original Message -
From:
Khaled
Chehab
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Se
How can I isolate directory address book search *411 depending
on context since context A user don't search at context B users
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with
Dear
I am using trixbox,I want ot disable and enable voicemail
from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was
working successfully
Database put AMPUSER/9990999 voicemail default
And
Database put AMPUSER.9990999 voicemail disables
Bu
Dear
I am using trixbox,I want ot disable and enable voicemail
from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was
working successfully
Database put AMPUSER/9990999 voicemail default
And
Database put AMPUSER.9990999 voicemail disables
But a
Who knows something interesting about the new BRI digium card b410p ?
For example, will it use the misdn driver or the native zaptel? Any
interesting links will be appreciated too.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Us
Julian J. M. wrote:
BRI ISDN is 2 channels, what would you want to do with a 3rd call?
Julian
On 6/30/06, francesco giuliani <[EMAIL PROTECTED]> wrote:
Does any boby knows how to manage a 3° incoming call in a BRI ISDN line
by chan_modem?
___
--Ba
: Hi All,
:
: I am plainging to give a solutions for a sports club. Follwing
: is the process that i need to achieve.
: If any body achieve this kind of setup pls give me a feedback, so
: that i can go through.
Have you even _tried_ to create your "dialplan"?
m.
__
I have used these in the past, with only one issue. The T1 line was at
the end of its tolerances as far as length from the repeater. The surge
suppressor ntroduced enough resistance to make the T1 bounce, like
Tigger.
Having the Telco put in a repeater closer to our facility made the
problem go aw
Snip
> On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote:
> > > trixter aka Bret McDanel wrote:
> >
> > > Lastly, and probably the least effective, is you can watch channel
> usage
> > > and when someone exceeds 5 run over to their desk and smack them
with
> a
> > > rotten fish.
> > >
> > >
If you'll use newer distribution of linux
you'll probably jump into problems with libsqlite3 (libsqlite2 is needed for
kannel).. it is well documented on kannel website.. you can contact me off-list
about kannel since this isnt't kannel mailing list...
I got kannel and asterisk running
Hi All,
I am plainging to give a solutions for a sports club. Follwing is the process that i need to achieve.
If any body achieve this kind of setup pls give me a feedback, so that i can go through.
Call flow start
[for database operations please use an access databa
The APC units work well, they have a rackmount module system also.
Protect yourself from grounding mismatch with security and paging
systems on channel banks also. Talk with your clients about emergancy
repair/replacement.
On 6/30/06, Dustin Wildes <[EMAIL PROTECTED]> wrote:
Just recently a c
Is there a fix for the problems with using the jitterbuffer on a trunked
IAX2 in asterisk 1.4 ?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mai
On Friday 30 June 2006 02:24, Philippe Lindheimer wrote:
> I would love to see some feedback on this as well. I've lost exact count
> now, but think I've seen about 5-6 failures on their cards TDM400P and
> TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that
Then put proper
On Thursday 29 June 2006 21:38, M.Hockings wrote:
> How reliable is Digium hardware in general.? My new TDM400P just died.
I have a number of Digium T1 products (T100P, TE410P, TE405P and TE406P) as
well as a few TDM400 based boards. No failures in the last 2 years or so.
> So, at over 2x the
A config file in text would be nice. Oh wait this is windows based,
config files don't exist anymore!!!
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian Stredicke
> Sent: Thursday, June 29, 2006 10:37 AM
> To: Asterisk Users
Having your users as admins on the local machine is generally a bad
thing to do, that means that any virus and/or spyware can install itself
into the machine without a problem.
It would be nice to know in what key SNOM stores the reg info, so that
one can simple grant full access to 'authenticated
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
(Sorry if you get more than one copy of this message, but I felt
that it was urgent to get this important info out.)
The values of freedom and openness are crucial to understanding
itself, so that civilization and public welfare now depend on
them,
I've got a few Cisco phones to maintain and need access to firmware
files. Dealers here in .fr want unreasonable prices for a Smartnet
subscription.
Where can I get a better deal on the Net?
You probably can't legally. Cisco controls who is allowed to resell
their contracts very very closely
Chris Mason wrote:
I have a client's installation that requires 4 lines PSTN interface only
so I am looking at 4 port FXO units. What works well with Asterisk and
is not exorbitant to purchase? Would a Sangoma remora be better?
The Sangoma A200D card has better echo canceller (if needed) compa
Just recently a client of mine took a lightning hit, which in turn blew
out their Digium TE411P board. This just so happened to be their main
office where their call center was located. We had a backup card on
hand, but this still meant downtime for the client until we got out
there to replac
M.Hockings wrote:
How reliable is Digium hardware in general.? My new TDM400P just died.
I am trying to determine if I have a lemon. This a new PC with a Digium
TDM400P in it with a single FXO and single FXS card just stopped working
today. It has been running less than three weeks with the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Philippe Lindheimer wrote:
> I would love to see some feedback on this as well. I've lost exact
> count now, but think I've seen about 5-6 failures on their cards
> TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I
> don't deal with t
I am having the same problem with my IAX clients. I posted
some issues that are causing my remote IAX agents to be disconnected due to
errors in setting up the IAX stream. I have found that calls will abandon when
a dynamic agent is logged into a down phone, the agent obviously can’t
logout
Hi Guys
With the profusion of different GUI's and Web interfaces out there could
someone possibly save me a load of time and let me know which is the best
one and why?
Also is there an independent site reviewing asterisk GUI's anywhere.
I'm looking at Cisco phones and TDM400 and X101P cards.
On
BRI ISDN is 2 channels, what would you want to do with a 3rd call?
Julian
On 6/30/06, francesco giuliani <[EMAIL PROTECTED]> wrote:
Does any boby knows how to manage a 3° incoming call in a BRI ISDN line
by chan_modem?
___
--Bandwidth and Colocation
Hi List,
i'm fiddling around with a blindtransfers. (and 3PTY)
a calls b
a transfers b to c (blindtransfer)
(c is not a party but a makro which puts b into a MeetMe conference)
the conference should be dynamically created. and "named" after the
callerid of a
therefor b has to know who which
This has been covered on the list many times, search the archives, the
Wiki and Google are your friend.
On a zap channel, Asterisk can't tell when a call has been answered, so
starts the playback immediately. Setup a loop asking the caller to press
a key. I have the following setup:
Doug,
Marcin Lukasik wrote:
But the problem is asterisk executes Playback() before the call is
actually
connected.
(On the console it says that Zap/2-1 answered while it's actually
trying to
ring on my mobile).
This has been covered on the list many times, search the archives, the
Wiki and Googl
Pavel Jezek wrote:
hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1,
but without success,
Here is my entries:
jitterbuffer=yes
dropcount=3
maxjitterbuffer=1000
maxjitterinterps=10
maxexcessbuffer=80
resyncthreshold=1000
minexcessbuffer=10
jittershrinkrate=1
--
Ben Frankl
Asterisk logs very detailed information in /var/log/asterisk/queue_log
file including abandoned calls. You can import this log to mysql with a
simple perl script running periodically.
-Original Message-
From: Michael Konietzny [mailto:[EMAIL PROTECTED]
Sent: Friday, June 30, 2006 11:44 AM
Hello for the first time :-)
I have a huge problem trying to create some sort of call back system.
What am I trying to do?
I call Asterisk, press 1 to call someone back and play announcement. Hanging
up.
Then I'm creating a file:-
Channel: Zap/2-1/07966011122
Context: call-them-back
Extension:
Does any boby knows how to manage a 3° incoming call in a BRI ISDN line
by chan_modem?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/lis
hello, I'm still trying to tune iax jitterbuffer in asterisk 1.2.9.1,
but without success,
I'm using idefisk->asterisk over cdma network, where rtt is about
100-500ms, so jitter about 400ms
but sound is very jerky, in diection idefisk->asterisk, in reverse
direction is sound relatively smoth,
s
I am using latest firmware, exactly 1.52
I am used to use PA168S phones in SIP mode (in the past I had problems
using them as IAX., i.e. passing calls and so on)
This is only for a test purpose, to test OH323 channel. It is not a
crritical issue, i never will use H323 on PA168S phones in a pro
On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote:
> > trixter aka Bret McDanel wrote:
>
> > Lastly, and probably the least effective, is you can watch channel usage
> > and when someone exceeds 5 run over to their desk and smack them with a
> > rotten fish.
> >
> > http://www.voip-info.
> trixter aka Bret McDanel wrote:
> Lastly, and probably the least effective, is you can watch channel usage
> and when someone exceeds 5 run over to their desk and smack them with a
> rotten fish.
>
> http://www.voip-info.org/wiki-Asterisk+sip+incominglimit
I can't find the 'rotten fish' stuf
[EMAIL PROTECTED] wrote:
> Hi all, I installed asterisk 1.2 branch, with oh323 channel support.
>
> Everything is fine, with netmeeting I can call and receive incoming calls,
> internal and external
>
> Then I tried to setup an AT320 phone , which is based on PA168S chip.
>
>
Which version of th
I don't use asterisk in combination with kannel.
Actually we use nowsms as SMSC gateway to connect to our provider but we deside
to
replace it by kannel.
so we store incoming messages in an sqlserver
2005 database in windows 2003 server .
please let me what you need to combine kannel and
1 - 100 of 113 matches
Mail list logo
