Yeah, thanks, that was the way I was leaning to. Just was wanting to
know if it was a syntax I was getting wrong, or if there is no other way
of doing this.
Julian.
Mojo with Horan Company, LLC wrote:
I guess you could do this, but it would be a little cumbersome:
context incoming {
s
hello
I have 2 services with 2different numbers. the
first is 88 and the second is 99. if a user call 88 I want to
execute the script1 and if he call 99 I execute the script2.
How can I do my configs files?
big Thanks
issam
___
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,ChandramouliMartin Joseph [EMAIL PROTECTED] wrote: On May 31, 2006, at 10:32 PM, Crazy Boy wrote: Hi Friends, I have successfully implemented
Create the 2 extensions in /etc/asterisk/extension.conf
exten = 8,1,Answer()
.
Script 1
.
exten = 9,1,Answer()
.
Script 2
.
Make sure that the channel where the calls come in route the call to the
context where you defined the scripts.
Hope this helps,
Henk
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 -
yes use sox. that's what am using
From: Mimmus [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: 'Asterisk Users Mailing List - Non-Commercial
Discussion'asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
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
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 Discussion'
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 AMD64
in
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... shrug
Probably any of you meet with the following problem:
asterisk is restarting in a minute (if no active
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 change
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
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
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 causes way more
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
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:
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 Sip express router
(SER) Iam using
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 :)
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
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
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
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
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
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,
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
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
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, file is
Hi!I've a question:
I've 2 asterisk, I want pull the ethernetwireand then reconnect it after 5 second, using the VRRP protocol, where must I set the IP for the connectiongoes on the second asterisk?
I want this:
I call to asterisk1, then Ipull the ethernet wire down, vrrp makes up the other
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
Quoted
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
___
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
___
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,
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
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 billing os SER on out
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 find it
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 asterisk1,
nice to see your feedback! looks promising.however, would you like to share the *, libpri and zaptel versions
you're running on these servers?cheerscurrently running asterisk 1.2.4, zaptel 1.2.5, and no libpri (we're running robbed-bit T1's, EM Wink signalling. a migration to PRI is scheduled, and
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
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:
On
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,
bp
On
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 ways
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 the
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
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
To:
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
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 the
Hi,
Im
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 didnt
detected any impact of this process proliferation in the system, but it is
strange and Im not comfortable with this.
Is this a know
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
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
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
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
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
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
does
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 0 10:58
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,
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
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 redirect an
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
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).
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 if
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
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
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.
___
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 1
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 ?
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
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 [EMAIL PROTECTED]
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 and
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
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 and
reload them
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
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
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
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 when
the
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 monitor
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 agent
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.
So all
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 seeing
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,
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
exten
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