I don't have any ODBC CDR stuff. I unloaded the ODBC Asterisk modules
and the problem occurred again about an hour later.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodney G.
McDuff
Sent: 01 June 2006 01:32
To: Asterisk Users Mailing List - Non
yes use sox. that's what am using
From: "Mimmus" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
Subject: [Asterisk-Users] Converting .wav to .WAV
Date: Wed, 31 May 2006 19:33:11 +0200
Hi
i am experiencing some problem with asterisk and misdn
i've patched and recompiled the 2.6.15.5 kernel on the server
i use a BN8S0 card with alla channels in TE mode.
i can load hfcmulti and mISDN_dsp
i load this with:
/sbin/modprobe hfcmulti layermask=0xf protocol=0x22 type=0x08
and then
Using svn trunk, I was trying to see what the astdb entry in the
sip.conf file does.
Nothing :)
I presume that it's meant to create an entry in the astdb.
so, I have
astdb=chan2ext/SIP/grandstream1=1234
in sip.conf
But database show only gives
*CLI> database show
/SIP/Registry/706
On Wed, 31 May 2006, Steven wrote:
What were the kernel parameters that you changed? (what OS, by the way?)
I am running CentOS 4.3, but have not changed any kernel settings yet.
Nothing exciting, just adding noapic did improve a lot on the hits:
title CentOS (2.6.9-34.ELsmp)
root (hd0,0)
Bruce,the sys stats shown above is at no call and I run the ps -auwxx, i couldn't see any process taking up the resources.for example, the maximum of cpu usage was asterisk -g -c and mysql and they are together
0.5% and 0.4% respectively.I have other servers running on Dell PowerEdge 2850 and
I think WAV is the file format and .wav is the file extension of wave file.
RecordPad sounds generate an extension of .WAV this creates some kind of
conflict. When files from RecordPad or WavePad dont play on asterisk simply
resample it with sox in same WAV format and you'd be fine.
From:
thanks for your response
Make sure that the channel where the calls come in route the call to the
context where you defined the scripts.
How can I do this?
big thanks
issam
- Original Message -
From: "Henk" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discuss
> > Which DSP based boards does Asterisk support for G729 and are any of these
> > more cost effective than piling on Pentiums?
>
> There are none at this time.
>
> > BTW: Can AMD CPUs handle a higher G729 load in 64 bit mode?
>
> Yes. The G.729 codec we distribute is marginally (6-7%) faster on
yes, it was a typo... and the problem of working too much...
> crontab? I restart my asterisk nightly with cron but a simple typo
> could make that every minute instead of every day...
> >Probably any of you meet with the following problem:
> >asterisk is restarting in a minute (if no active c
Hi All,
I have a SIP provider that tells me that my RTP stream uses a
"20bytes payload in the g729 coded data". And they would like that we
change this to 30bytes (3 frames).
But maybe I'm wrong but isn't a certain payload just a standard for a
codec ?
And if I'm wrong, how can I chang
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> -=-=-=-=-=-
> -=-=-=-=-=-
>
> Eh, I'm thinking I don't like labels very much. They aren't all they are
> cracked up to be.
>
> Previously, using extensions of the format extension-function, like
> 2944000-open or
> 2
Hi!
Any of you played with tarification tone?
We are planning to insert and asterisk box in front of a panasonic
with PRI, but the old pbx still needs the tarification tone.
Btw, it would be nice, if we could use the tone is asterisk itself
(rather than connect the cdr with a tarification system).
In article <[EMAIL PROTECTED]>,
Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Ira wrote:
>
> > I would be happy to do this. Is there something that describes how I
> > might accomplish this. One of the things I've never quite figured out
> > is how to save the console output and a SIP debug cause
I have just finished building a prototype IVR server on a pc for
demonstration purpose.
My goal is to build a IVR server with the 4G memory, dual xeon processor and
a 4 x E1 card. The server would strictly receive incoming calls and serve
WAV files.
my question is: Is this not an over kill?.
use DBput a DBget (http://www.voip-info.org/wiki/view/Asterisk+database)
astdb=chan2ext/SIP/grandstream1=1234 is only variable
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Thursday, June 01, 2006 9:39 AM
To: asterisk-
Hello Masters
Here i going explain what Iam doing and where i need help ..
Iam
running Sip Express Router ,Asterisk, on same box (for testing) my Sip
express router is working fine and i can accept global register
requests with valid account and in front of S
Err, I'm not trying to write to the db using the dialplan. In sip.conf
there seems to be the ability to automatically create a db entry on
startup. The line in sip.conf is
astdb=chan2ext/SIP/grandstream1=1234; ensures an astDB entry exists
But it doesn't ensure an astDB entry exists :)
Ju
Hi! Im looking for a very basic example for the following simple problem. I've been searching voip-info.org and looked in the ORA book without a clue. I have a SIP account at
sip.provider.com and my own asterisk server. What I want is the following: I. Register my phone to my asterisk serve
On 6/1/06, Tony Mountifield <[EMAIL PROTECTED]> wrote:
# script /tmp/output.txt
Script started, file is /tmp/output.txt
# exec asterisk -rv
... do asterisky stuff ...
host*CLI> exit
Script done, file is /tmp/output.txt
#
Actually you need another exit in there:
# script /tmp/output.txt
Scr
I have Digium IAXy 101I.
To provision the IAXy, I am following the instrution to download the
utility (iaxyprov package):
on my linux server (asterisk) I type
#cd /usr/src
and
#export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
then
#cvs login (with password anoncvs)
but after typing the pas
look for SER and Asterisk on voip-info.
I think, you plan to got to UA->SER-(mediaproxy)->Asterisk->PSTN
if yes, ser will communicate UA (user agent) on one leg, and asterisk on
other. you can use your asterisk to billing and pstn connection.
on incoming call dial $phone/ip.address.of.ser
>
On Thu, 2006-06-01 at 11:18 +0200, Benjamin Stocker wrote:
At least you know to break this down into different parts, it still
amazes me how many people look at something as one big thing instead of
several smaller things that interrelate :)
you should have example config files that came with as
digium no longer use cvs.
You need to download using subversion.
Julian.
Andrea Bencini wrote:
I have Digium IAXy 101I.
To provision the IAXy, I am following the instrution to download the
utility (iaxyprov package):
on my linux server (asterisk) I type
#cd /usr/src
and
#export CVSROOT=:pser
Andrea Bencini wrote:
Unknown host cvs.digium.com
Quoted from the message on the 23rd.
"As announced when the Asterisk project converted to Subversion as our
version control system late last year, it is time to decommission our
CVS servers.
As of some time in the next couple of days, th
Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk.
For the second time now, I've had asterisk on a production machine
completely freeze (with no messages in any of the log files) and
eventually had to be kill -9'd.
The machine has a a TDM400 with 1xFXS and 3xFXO cards in it
Further to my previous email, I have definitely established that the
audio gets choppy only when the path includes sip and capi.
PAP2 to Asterisk to MyNetFone to PSTN is fine.
PAP2 to Asterisk MOH is fine.
PBX (via capi) to Asterisk MOH is fine
PBX (via capi) to Asterisk to PAP2 is choppy
PBX (via
In article <[EMAIL PROTECTED]>,
Andrew Furey <[EMAIL PROTECTED]> wrote:
> On 6/1/06, Tony Mountifield <[EMAIL PROTECTED]> wrote:
> > # script /tmp/output.txt
> > Script started, file is /tmp/output.txt
> > # exec asterisk -rv
> > ... do asterisky stuff ...
> > host*CLI> exit
> > Script done, fi
Hi!I've a question:
I've 2 asterisk, I want pull the ethernet wire and then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connection goes on the second asterisk?
I want this:
I call to asterisk1, then I pull the ethernet wire down, vrrp makes up the other ast
Thomas Kenyon wrote:
Is it neccesary to upgrade Zaptel at the same time as upgrading asterisk.
I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons
and Sounds.
Doug
___
--Bandwidth and Colocation provided by Easynews.com --
I there any good reason that is doesn't get posted to the ftp site?
People that only use stable may find it easier.
--
--
Steven
http://www.glimasoutheast.org
"Doug Lytle" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> Andrea Bencini wrote:
>> Unknown host cvs.digium.com
>>
>>
Forrest Beck wrote:
Does anyone have a working implementation of SIP Presence? I have a new
Grandstream GX-2000 phone with the supported hardware and I am not sure how
to setup presence with asterisk.
I've just been through this myself. It is relatively simple once you
manage to figure it
Steven wrote:
> I there any good reason that is doesn't get posted to the ftp site?
> People that only use stable may find it easier.
You mean like this?
http://ftp.digium.com/pub/telephony/
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.
Steven wrote:
I there any good reason that is doesn't get posted to the ftp site?
People that only use stable may find it easier.
I wouldn't be able to answer that. I'm just a every day user, such as
yourself, that saw the posting.
Doug
___
--B
I had exactly the same problem a couple of weeks ago, but have since
moved to fc5 on that box. If I recall correctly, it was the
/etc/cron.daily/prelink
that caused the problem.
Rich
Sean Kennedy wrote:
chan,
Run each script seperately to determine which one causes the crash.
From there, c
I've never attempted to use this feature, so I can neither confirm nor
deny whether it works/doesn't work/used to work/etc.
But what I find really odd, is that the code doesn't even appear to try
and parse "astdb" when it's loading the config, at least insofar as I
can tell. A quick grep -i astdb
Yeah, just found http://bugs.digium.com/view.php?id=3359 where it seems
to have been closed out, the code never making it into chan_sip.c
However, the option *did* make it's way into sip.conf, so I guess that
the real bug is that the option is in sip.conf.
Bummer.
Devels: Any chance of getti
Interesting
I always kind of thought is was a cool option to have, though (as I
already mentioned) never needed it in my situation(s).
That's pretty strange that the option exists in the sample, though.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Julian Lyndon-Smith wrote:
> However, the option *did* make it's way into sip.conf, so I guess that
> the real bug is that the option is in sip.conf.
Yes, and I will fix that in a few minutes.
> Devels: Any chance of getting 3359 re-opened and put into asterisk ?
No, because it's doesn't really
Wonderful explanation!
Just a note:
> So, having done all this, restart asterisk, then reboot your
> phones (an asterisk restart confuses hints/presence on
> grandstream phones sometimes)
It seems that Asterisk >= 1.2.7 solved this issue.
Bye
Domenico Viggiani
__
Hello Guru
Thanks for giving reply. so, i can use mediaproxy for both SER
and as well as ASTERISK
but you told me about voip-info but i dint find much docs regarding SER+asterisk cookbooks
and you told me you can use Asterisk as billing for pstn of
incoming calls and
yup, like that, but with an iaxyprov folder.
--
--
Steven
http://www.glimasoutheast.org
"Matt Riddell (IT)" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> Steven wrote:
>> I there any good reason that is doesn't get posted to the ftp site?
>> People that only use stable may fi
On Thu, 2006-06-01 at 12:30 +0200, Shenen Shenen wrote:
> Hi!I've a question:
> I've 2 asterisk, I want pull the ethernet wire and then reconnect it
> after 5 second, using the VRRP protocol, where must I set the IP for
> the connection goes on the second asterisk?
> I want this:
> I call to asteri
nice to see your feedback! looks promising.however, would you like to share the *, libpri and zaptel versions
you're running on these servers?cheers currently running asterisk 1.2.4, zaptel 1.2.5, and no libpri (we're running robbed-bit T1's, E&M Wink signalling. a migration to PRI is scheduled, a
I use the goto to jump across contexts with labels all the time.
goto(context,exten,label). works for me.
Jason
Michael Collins wrote:
Oh Crud. So, if I want to jump to another extension or context, I have
to
specify the full context, extension and priority? I can't specify a
That's a shame, as I was hoping to use it. Our sip.conf file is produced
automatically by a generator and ftp'd to the * server, so there is no
manual editing by the administrator .
I was wanting to link an [extension] to an email address so that I can
do some stuff in the dialplan. This would
Sipura 3000 or the Digium TDM03B
On 6/1/06, Crazy Boy <[EMAIL PROTECTED]> wrote:
Hi,
As you said, May I know the correct Digium or Sipura product model
(Sipura-3102 or Digium?), which is suitable to my requirements?
Thank you.
Regards,
Chandramouli
Martin Joseph <[EMAIL PROTECTED]> wrote:
That's an issue with your IP phone. Check your configuration. I believe most phones call that "digit timeout" or something like that... it should be set to about 3-4 seconds.
You can also try pressing # after dialing the number. On most phones, that will make it dial the number.
Good Luck,
b
Julian Lyndon-Smith wrote:
> Instead now I've got to make sure that the administrator is reminded to
> manually update the astdb everytime an email address for the extension
> changes or new phones / people are added or removed.
As the notes in that original bug told you, there's lots of other wa
I apreciate all the help. There is something about putting your conf file in an email that helps you see the problems. As I went over them I find small things and in the end it works. Thanks again for the advice on things to check.
On 5/31/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
Is th
Is there a way to use 'application mapping' from features.conf without the built in features (pickup, blind transfer, etc.) nor call parking?
I have been trying to comment out everything in features.conf, but my asterisk stills shows the defaults...
Koen
__
Hi,
I am keen to try out the SIP jitter buffer capability. I hear this was
available if HEAD.
I was wondering if a version of the latest STABLE with this additional
feature was available some place.. Or is it simply best to use HEAD?
Would some one be kind enough to point me in the right d
> From: Douglas Garstang <[EMAIL PROTECTED]>
> Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
> functionality, but lose cid in the dialplan. Hmmm.
>
> > -Original Message-
> > From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
> > Sent: Wednesday, May 31, 2006 1:21 PM
>
AFAIK, it's only available in Head.
Julian.
James Gardiner wrote:
Hi,
I am keen to try out the SIP jitter buffer capability. I hear this was
available if HEAD.
I was wondering if a version of the latest STABLE with this additional
feature was available some place.. Or is it simply best to
I set up hints and presence monitoring on some Polycom phones
connected to an asterisk server with the expectation that the phones that are “watching”
other extensions would be notified when the other extension sis ringing, in
addition to the other statuses (on the phone, statuses set by th
Hi,
I’m
running asterisk 1.2.0 on a debian rel 2.6.13 and when I start it with safe
asterisk I got instantly more then 10 processes. Until now I didn’t
detected any impact of this process proliferation in the system, but it is
strange and I’m not comfortable with this.
Is t
Hi , when i call to asterisk from a Skype or Voipbuster phone all the
extensios runs good , and i can stablish ZAP to SIP comunication, also
i can do a SIP to ZAP call , but when i call from a traditional
analogic phone i get these error:
chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on ca
Yeah, I know. Just was hoping to have things the easy way for me. I also
want not to have a custom patched box as I know *one* day I'll screw up
and lose / forget the patch and wonder why things aint working.
Thanks anyway. I'll stop bitching now.
Julian.
Kevin P. Fleming wrote:
Julian Lyndo
Yes! That's the answer I was hoping ! I'm not stupid - it's a *feature* :)
Anything you need testing, let me know !
Julian
Steve Murphy wrote:
From: Douglas Garstang <[EMAIL PROTECTED]>
Yikes! I'm glad I didn't take the plunge into AEL2. Get #include
functionality, but lose cid in the dialpla
Hi,
I would like to know if it is possible to redirect an incoming call to
an external phone number. Can this be done easily?
Thanks in advance,
Javier
___
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Asterisk-Users mailing list
To UNSUBSCRIB
I have an Asterisk system connected with a CLEC that provides SIP
termination. When placing calls from phones on the Astersik system to
the PSTN, the calling party hears ringing while the called party is
saying "hello".
The problem appears to happen when calling a POTS line. The problem
doe
In case this will be of any use, here it
is a list of the processes. We can see that the safe_asterisk script (PID
19368) starts the first asterisk process (PID 19389) that starts a second one
(PID 19401) and this second one is responsible to start all the others.
root
19368 1
I've been looking for the same answer and have posted it twice. I hope someone
will eventually have an answer for us both!
= = = Original message = = =
Hi,
I'm having a problem with the timeout option when dialing a ZAP channel.
The goal is to ring a number for 15 seconds, if no one picks up, g
I am trying to create a %100 g729 (with no transcoding) system (using a
Soekris, of course). I am running AstLinux with the native sounds, g729
is the only codec allowed, %100 SIP (g729 only allow=) - I think I am
covering all of my bases.
I have only "format=g729" in voicemail.conf. On an
Hi,
after some corrections in my settings IAX2 dialin seems to work now. I
get the incoming call, but i cannot here anything or can speak.
(If I take the call the other side see that the connection is
established if I close the call the other site is seeing it too)
If I press hold in Idefisk the
I set up hints and presence monitoring on some Polycom
phones connected to an asterisk server with the expectation that the phones
that are “watching” other extensions would be notified when the
other extension sis ringing, in addition to the other statuses (on the phone,
statuses set by th
Cambiando un timer que existe en el archivo mfcr2.c
La variable DEFAULT_T1 tiene el valor 5000, incrementalo a 2, compilas, instalas y listo…
mas o menos en la linea de codigo 102…
actual
#define DEFAULT_T1 5000
despues
#define DEFAULT_T1
sure,
exten => 1234567890,1,dial,SIP/[EMAIL PROTECTED]
Obviously change SIP for Zap, or IAX if you are using those.
When someone calls 1234567890, the pstn phone 9876543210 will ring.
bp
On 6/1/06, Javier Rodriguez <[EMAIL PROTECTED]> wrote:
Hi,I would like to know if it is possible to redire
Kristian Kielhofner wrote:
I am trying to create a %100 g729 (with no transcoding) system
(using a Soekris, of course). I am running AstLinux with the native
sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) -
I think I am covering all of my bases.
I have only "form
Using * 1.0.9
I have a cron job that runs every night and sucks Caller ID information from
our SQL server based CRM and imports it into the Asterisk database. We use
the Caller ID to give enhanced information about the caller ("this is his
customer number, he's a pain in the ass", for example). Th
Hello All,
Complete newbie to asterisk (OH NO). Is it possible to use my skype
out account for an outgoing trunk? If so, can the syntax be found
somewhere? Thanks, Peter
--
cybersource.us
115 Richfield Road
Williamsville, New York 14221
Of course... you only need to Dial to other port FXO connected to PSTN and passing the number as extension:
[redirection] ; your inconming-calls context
exten=>s,1,Dial(Zap/${OTHER_FXO}/${EXTERNAL_NUMBER})
exten=>s,2,Hangup
2006/6/1, Javier Rodriguez <[EMAIL PROTECTED]>:
Hi,I would like to know i
The codec is not just for transcoding audio.
It is required to read and write it as well.
--
--
Steven
http://www.glimasoutheast.org
"Kristian Kielhofner" <[EMAIL PROTECTED]> wrote in message news:[EMAIL
PROTECTED]
> I am trying to create a %100 g729 (with no transcoding) system (using a
On Wed, 2006-05-31 at 19:22 -0500, Moises Silva wrote:
> google "zaptel hdlc"
>
So it makes no difference if you are using R2 instead of ISDN?
--
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
signature.asc
D
Correct me if I'm wrong, but doing this CID stuff in AEL may not make as
much sense in terms of converting dialplans over as it seems. I say this,
because with the original usage of the CID checking in the old extension
language, you could base PRIORITIES on the CID, therefore changing only
pa
Peter,There is a bounty for someone to get this working, but there's no simple solution as of yet. There is some SIP-to-Skype software that exists, but it is currently only on Windows, and involves a very convoluted setup.
AlexOn 6/1/06, Cyber Source <[EMAIL PROTECTED]> wrote:
Hello All, Complete
Steven wrote:
> The codec is not just for transcoding audio.
> It is required to read and write it as well.
Not true. It's possible to do playback of compressed files without
having that codec installed. It should also be possible to record them.
___
--B
Kristian Kielhofner wrote:
> -- Recording the message
> -- x=0, open writing:
> /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729,
> 0x8140f88
> Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to
> find a codec translation path from g729 to slin
> Jun
Attilla De Groot a écrit :
Hi All,
I have a SIP provider that tells me that my RTP stream uses a
"20bytes payload in the g729 coded data". And they would like that we
change this to 30bytes (3 frames).
But maybe I'm wrong but isn't a certain payload just a standard for a
codec ?
You'
Hey guys,
i'm wondering if there is any good way to get app_queue working in real
roundrobin strategy. The idea
is to specify a call list of, lets say, 3 agants. Those agents should always be
called in the correct defined order.
So all calls have to get the following agent priority: 1st Agent -
On Jun 1, 2006, at 1:36 AM, Akpome Akpoguma wrote:
I have just finished building a prototype IVR server on a pc for
demonstration purpose.
My goal is to build a IVR server with the 4G memory, dual xeon
processor and a 4 x E1 card. The server would strictly receive
incoming calls and serve W
Digium Wildcard TDM400P with 4FXO port
--- Crazy Boy <[EMAIL PROTECTED]> wrote:
> Hi,
>
> As you said, May I know the correct Digium or Sipura
> product model (Sipura-3102 or Digium?), which is
> suitable to my requirements?
>
> Thank you.
>
> Regards,
> Chandramouli
>
> Martin Joseph <[EMAI
you can use set-var in sip.conf to accomplish this same thing.
On 6/1/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
Yeah, I know. Just was hoping to have things the easy way for me. I also
want not to have a custom patched box as I know *one* day I'll screw up
and lose / forget the patch an
Hello everyone,
I'm sure someone had an experience arranging hunt-group setup for
incoming calls on T1 PRI channels of Digium TE110P card.
For instance, I have main DID channel associated with number (555) 222 0001.
And I have whole bunch of other DID channels on same T1 card like (555)
222 0
On Jun 1, 2006, at 2:18 AM, Benjamin Stocker wrote:
Hi!
Im looking for a very basic example for the following simple problem.
I've been searching voip-info.org and looked in the ORA book without a
clue. I have a SIP account at sip.provider.com and my own asterisk
server. What I want is the
Doug Lytle wrote:
> Thomas Kenyon wrote:
>> Is it neccesary to upgrade Zaptel at the same time as upgrading
>> asterisk.
>>
>>
>
> I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons
> and Sounds.
>
> Doug
>
The problem with zaptel is that even if you can unload the modules an
Kevin, since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure soun
Sorry for the repost - forgot to put the proper subject last time.Kevin, since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't th
Kevin P. Fleming wrote:
Kristian Kielhofner wrote:
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729,
0x8140f88
Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to
find a codec translation path from g729 t
Hello,
Try both asterisk and ser for IM/presence .
--- Damon Estep <[EMAIL PROTECTED]> a écrit
:
> I set up hints and presence monitoring on some
> Polycom phones connected
> to an asterisk server with the expectation that the
> phones that are
> "watching" other extensions would be notified whe
I've been struggling with a distortion/crackling problem with the Monitor
command in asterisk. I've even brought the dialplan down to a very simple 3
lines...
exten => 263949,1,Answer
exten => 263949,2,Monitor(wav,${CALLERIDNUM})
exten => 263949,3,Wait(10)
The .wav files generated from the monit
Michael Konietzny wrote:
> i'm wondering if there is any good way to get app_queue working in real
> roundrobin strategy. The idea
> is to specify a call list of, lets say, 3 agants. Those agents should always
> be called in the correct defined order.
>
> So all calls have to get the following
On 17:41, Thu 01 Jun 06, Michael Konietzny wrote:
> Hey guys,
>
> i'm wondering if there is any good way to get app_queue working in real
> roundrobin strategy. The idea
> is to specify a call list of, lets say, 3 agants. Those agents should always
> be called in the correct defined order.
>
>
>Avoiding mismatched codecs goes a long way (ie Asterisk doesn't need to
>transcode).
I use a quad Xeon 700, supporting ~200 users w/ > 30 PSTN calls and ~ 40 SIP
& ~30 IAX calls on the box pretty much continuous, 16 hours a day, 7 days a
week. Looking at 'top' right now w/ 41 PSTN calls I'm seei
If you have a PRI to a telco, they probably only have a single trunk
group with 23 channels in it for your connection.
Any calls to any of your numbers may come to you on any channel.
Channels are not dedicated to individual numbers.
In other words the first call may come in on channel 1, t
If I understand this correctly, you want to be able to accept
simultaneous calls to the single DID (555) 222 0001. There is no reason
to "roll over" to another DID if there is already a call on that DID.
You can receive as many calls to a single DID for as many channels you
have on your T1.
-
Hi
If you have conference or 2-way calling (or whatever is that called by
telco), look for Flash application. Basically, you would need to flash
the line on incoming call, dial new external number with DTMF and
hangup. It will redirect the call:
exten => 52,1,Wait(1)
exten => 52,2,Flash
exte
>My goal is when a call comes to the main number which is (555) 222 0001,
>to have it roll over to the next available T1 channel.
>Or give busy signal to the caller if all channels are busy. To have one
>main number for all calls.
I believe the dialstring:
exten => _XXX,1,Dial(Zap/G0/55512
Looks like you need someone else to do the setup for you.
Anyhow, if you have 23 channels, then it rolls over to the next
channel without any configuration. The only thing that could prevent
your dialplan from handling this nicely is if you have a different
context for each channel in zapata.conf.
Andrei (MPI) wrote:
Hello everyone,
I'm sure someone had an experience arranging hunt-group setup for
incoming calls on T1 PRI channels of Digium TE110P card.
For instance, I have main DID channel associated with number (555) 222
0001.
And I have whole bunch of other DID channels on same T1
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