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 many systems, which makes this really concerning. I've started a thread on the
Well, I know to be compatible with porta-billing you need password to do ip
based auth. It's a bit goody but they basically seem to expect
if trusted ip and no Digest support then radius auth has username=src_ip and
password=x.
To put it another way it would be help full to porta-billing
On Thu, Jun 29, 2006 at 10:54:58PM -0500, Anton Krall wrote:
So, no answers? Nobody knowd why this might be happening? Nobody else
experiencing this?
Is this a reproducable issue? Have you turned on verbosity and debug and
log them (e.g. the full log)?
If still no messages and this is
ok, We are building the perfect voip company..we are trying...we need input on end-users:reply to my email with --ENDUSER in subject.with anything you would like to see your current voip provider offer online/offline ( don't say.. support, an answer on phone etc) be constructive..
reply to myt
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)
Then test, you'll see it happen.
Cheers
Hoa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello,
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?
Thanks,
___
--Bandwidth and Colocation provided by
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,
yes, to avoid transcoding problems I only have one
codec, just alaw. Anything else is disallowed.
Hello List,
i'm wondering if there is any way to get a AGI executed if a caller
disconnects while he is INSIDE the queue application. If so, i would
like to log the call as missed.
Hope someone can help.
Greetings,
Michael
___
--Bandwidth and
T. Shaw wrote:
Hello all,
I have a problem with call quality with my Asterisk setup. I'm doing
VOIP only so far, but have a zaptel TDM400P in the box not being used.
The problem i'm having is that when calls are placed, connected, and
the far-end is reporting that they are experiencing
Hi
I tried that too, but the only useful thing I can change
(besides the IP settings of the phone itself) is the "CallSv" parameter; I set
it to the IP of the SIP registrar/proxy but it still doesn't
work...
Silviu
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Martin Joseph wrote:
On Jun 29, 2006, at 2:43 PM, T. Shaw wrote:
thanks for all the responses. I feared that it might be a bandwidth
issue. We have a (supposedly) business DSL line that is 1.5M - 3M
down/ 512k up. might have to bump that up to a higher grade.
If you are actually getting
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.
I can receive call from internal or external phones, and
Hi Jeremy,
Thanks for your suggestion - but our project requires certain features
that have to be additionally implemented, which means we cannot work
with what is out there already.
Best regards,
Mike
Jeremy McNamara wrote:
Mike Puchol wrote:
Hi all,
I am looking for a developer or
Hi List
I have 10 separate SIP phones, and I wish to limit the simultaneous
available channels to 5 maximum for these. How would you go about it
without setting up a separate * box?
Cheers,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP
On Fri, 2006-06-30 at 13:31 +0400, Jean-Michel Hiver wrote:
Hi List
I have 10 separate SIP phones, and I wish to limit the simultaneous
available channels to 5 maximum for these. How would you go about it
without setting up a separate * box?
Cheers,
Jean-Michel.
you can limit it to
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
[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 the
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' stuff
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.
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
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,
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:
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:
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
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
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
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,
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
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 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.
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 cant
logout
-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 that
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
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
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)
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
-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,
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
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
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
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
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:
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
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 database
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
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.
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
: 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.
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?
___
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 --
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 at
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 at
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
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]
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
== 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
___
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
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]
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
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
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
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:
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
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
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
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
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
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
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
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,
yes, to avoid
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
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
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 indication from
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
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
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 --
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 show
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-handle message
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
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
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
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
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
The following command on the Asterisk console will reboot a polycom phone:
sip notify polycom-check-cfg ip-addr
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
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
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
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
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
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
snip
But at trixbox its not working
Any ideas pleas
Did you try checking with the people who _wrote_
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:
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
Im 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
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
-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 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:
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
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
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
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 wondering
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
1 - 100 of 114 matches
Mail list logo
