Thank you knox. Finally, I have chosen this solution : find
/var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm f, executed
every night by the CRON. However, I would have preferred this feature was
implemented in Astrisk.
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL
Hi Friends,
I bought a new Black Berry 8800 mobile. This mobile is coming with Wi-Fi
feature. After turning on this Wi-Fi feature in my mobile, It is not detecting
my wireless router in our office. How can I do this?
How can I configure my VoIP (SIP) server in this Blackberry 8800 mobile?
I
Hi all,
i never understood that why is there 2 branches of asterisk going on
parallel. asterisk 1.2.* and asterisk 1.4.*, i also heard about beginning of
another branch which will be 1.6.*. so whats the difference between these 2
or 3 versions, can anybody plz tel me?
--
Rizwan Hisham
Software
So... I guess it's something in the 3102 that must be changed so that
it will finally TX/RX voice packets to remote phones (works fine when
picking up an IP phone in the same LAN as the 3102 and Asterisk).
Here's something from an old post:
Upon replacement of the Linksys, everything worked
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett [EMAIL PROTECTED] wrote:
What's going on
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hello,
i just want to activate SMS service between my asterisk local sip accounts
and between asterisk and local sip accounts. How can i do this thin? Also i
tried smsq to an account but all i obtained is a error message:
---Cut Here---
May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open
That looks like exactly what I want, we are currently on 1.2, ill see if
i can hack similar functionality into it, if not ill have to upgrade to
1.4 (probably best anyway)
Thanks for the pointers.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hello,
I need to connect asterisk 1.2.16, with a Contect Center software that works
with TAPI.
As I know, asterisk doesn't support TAPI directly, if needs a tirth party
software.
I just reading about asttapi and Activa TAPI.
does anyone test this software? have you using asterisk againts a TAPI
Am Montag, den 21.05.2007, 23:16 -0500 schrieb Mike Hammett:
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
In the context where your internal calls usually are handled, like this
(my internal phones have SIP
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my asterisk local sip
accounts and between asterisk and local sip accounts. How can i do
this thin? Also i tried smsq to an account but all i obtained is a
error message:
Hi,
I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for
outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed
version ) to this version and in my opinion a lot more troubles arose
For outgoing calls I use a Digium B410P with chan_misdn
Hello all,
One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?
We're using Polycom telephones, do you think they could be causing it?
Thanks,
Alex
Dear all.
I have what appears to be a configuration error but I cannot for the life of me
see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
Unable to make outgoing calls via E1 euroISDN Digium
On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:
Hello all,
One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?
We're using Polycom telephones, do you think they could be causing it?
By
Hi,
Could be bandwith or/and latency ... Many causes...
Alex
Asterisk a écrit :
Hello all,
One of our clients reported that they are experiencing echo in SIP calls
(even on internal ones). What do you think could be causing echo in
internal SIP calls?
We're using Polycom telephones, do you
On 5/22/07, Matt Scott [EMAIL PROTECTED] wrote:
Dear all.
I have what appears to be a configuration error but I cannot for the life of
me see what it is. (I am a newbie)
I have searched the wikki and google etc but still none the wiser. Any help
would be very gratefully received.
Problem:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On 5/22/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
In your dial lines you have an extrac comma (,)
exten = _9xxx,1,Dial,(${OUTBOUND}/${EXTEN:1})
should be
exten = _9xxx,1,Dial(${OUTBOUND}/${EXTEN:1})
or
exten = _9xxx,1,Dial,${OUTBOUND}/${EXTEN:1}
Good catch Morgan!
Thank you.
How do you think would be the best way to approach this problem? Do you
think anything else could also produce echo as well?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Moore
Sent: Tuesday, May 22, 2007 2:16 PM
To: Asterisk Users
How could I check if bandwith or/and latency is causing it?
If I do SIP show peers it says OK (13 ms) for all peers. I guess there is a way
to gather more detailed info on SIP calls and latency?
* box is connected to the 1Gb switch with 1Gb connection, and clients have
100Gb/s speed. CPU is
We experience echo too from time to time. It's usually headset-related, but
not always. I ran a persistent ping on one of the phones, and we diagnosed a
wiring problem with it. Other phones needed a new handset. The problem is
that these problems need to be fixed on the phone NOT hearing echo.
Hi all,
how can I catch the event generated when a parked call is hung up?
In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user hangs
up the parked call, I don't know how to catch the event, from
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Thanks. I will try to ping my phones, to see what's the situation.
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gomillion
Sent: Tuesday, May 22, 2007 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Echo
ok so now I changed ext-local to dundi-ext and I created this
context at the bottom of the extensions file. This is now the case.
[dundi-priv-canonical]
; Direct numbers
exten = 5010,1,NooP(DUNDI LOOKUP 5010)
exten = 5011,1,NooP(DUNDI LOOKUP 5011)
exten = _60XX,1,Goto(dundi-ext,${EXTEN},1)
I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
1.4 this weekend. Intially everything looked like it was working properly,
but some time in the day following the upgrade, the system died to a kernel
panic. I wasn't able to catch the entire kernel dump on the console
I tried with the ping ... all of the phones respond in ca. 0.3ms, so network
seems to be OK. More than 90% of CPU on * box is idle even in peak times, so
this shouldn't cause echoes either, right? Hmmm, so handset could be an issue,
but did anyone ever experience any handset problems with
On Tue, 22 May 2007 11:45:34 +0200, randulo [EMAIL PROTECTED]
wrote:
Upon replacement of the Linksys, everything worked fine except audio
on the Sipura. Turns out you need Symmetric RTP turned on in the phone
as Chris Mason says below.
Thanks for the tip. The IP phone doesn't have a setting that
Are your phones reinviting? Do you have any strange routing weirdness, or
are they all on a single subnet?
On 5/22/07, Asterisk [EMAIL PROTECTED] wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
The only way I have ever seen any SIP and/or Network configurations is
from the Enterprise server management screen. If you purchased the 8800
thru a participating carrier, RIM offers a single user express license
for free (with purchase) Google for Free BlackBerry Express and that
should give
Thank you for reply. Can you send me some working configs? I'm still
confusing about this sms option.
On 5/22/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my asterisk
In Sip.conf I have the following: canreinvite=no
No, all telephones are on the same subnet, handled by the same switch. I cannot
verify if anything has been changed since I installed configured the network,
but as far as I know the whole network configuration is pretty straightforward,
Because there is a huge install base on 1.2 which is fairly stable but still needs bug fix/security patches. There are no new features being developed by the code group on this version, but there are outside people who are still modifying it.
1.4 has many new features and some areas re-written.
Asterisk wrote:
In Sip.conf I have the following: canreinvite=no
No, all telephones are on the same subnet, handled by the same switch.
I cannot verify if anything has been changed since I installed
configured the network, but as far as I know the whole network
configuration is pretty
On Tue, May 22, 2007 at 10:07:19AM -0400, James FitzGibbon wrote:
I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
1.4 this weekend. Intially everything looked like it was working properly,
but some time in the day following the upgrade, the system died to a kernel
[EMAIL PROTECTED] wrote:
Hi all,
how can I catch the event generated when a parked call is hung up?
In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user hangs
up the parked call, I don't know how
Try canreinvite=yes in order to confirm that CPU is not the problem.
Jorge Mendoza
Asterisk wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
peak times, so this shouldn't cause echoes either,
Alex wrote:
I tried with the ping ... all of the phones respond
in ca. 0.3ms, so network seems to be OK. More than
90% of CPU on * box is idle even in peak times, so
this shouldn't cause echoes either, right? Hmmm, so
handset could be an issue, but did anyone ever
experience any handset
Thanks guys for the tips. I will try that.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Tuesday, May 22, 2007 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Echo
Lee Jenkins wrote:
[EMAIL PROTECTED] wrote:
Hi all,
how can I catch the event generated when a parked call is hung up?
In my dialplan, when arrives a call to a specific number, Asterisk parks
the call and announces the parking slot to a number. But if the user
hangs
up the parked call,
This method does not seem to work. The action (NoOp in my case) in the h
extension is execute after have parked the call, while when I hang up the
call parked the action in h extension is not execute.
___
--Bandwidth and Colocation provided by
Hi there,
I am having some problems while trying to place phone calls through Asterisk
to Net2phone, this is my setup:
I have a SIP phone connected directly to my Asterisk box from where I want
the call to origin; in sip.conf:
[mySIP]
type=friend
username=mySIP
secret=mySecret
host=dynamic
Asterisk wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
peak times, so this shouldn't cause echoes either, right? Hmmm, so
handset could be an issue, but did anyone ever experience any handset
[EMAIL PROTECTED] wrote:
This method does not seem to work. The action (NoOp in my case) in the h
extension is execute after have parked the call, while when I hang up the
call parked the action in h extension is not execute.
So Asterisk sees the parking of the call as the hanging up of that
Hi,
Did you implement QoS (Quality of Service) in your network?
Thanks.
Regards,
Chandra
Stephen Bosch [EMAIL PROTECTED] wrote: Asterisk wrote:
I tried with the ping ... all of the phones respond in ca. 0.3ms, so
network seems to be OK. More than 90% of CPU on * box is idle even in
peak
On Monday 21 May 2007 3:38 pm, Doug Lytle wrote:
Doing a 'man sox' does wonders:
The question, however, is is Asterisk playing them louder than normal, or are
they recorded too loudly to begin with?
Adjusting volume gains on these files is the LAST thing you should do.
Determine what the
I am somewhat confused. I have the incoming (s) context playing a greeting
and callers choose one of two extensions (100, or 101)
To the caller it ALWAYS sounds as though the phone is ringing. However,
sometimes it is not actually ringing the phones
The listext.wav file suggests extensions 100
The latest and greatest Astsee is now available at
http://www.astsee.com/ I'm up to v0.5 today -- the light at the end of
the tunnel edition.
In progress is a way to audit this sort of traffic _without_ manager
credentials ;) Just by sniffing it off the wire or out of the air...
You can
On Sunday 20 May 2007 11:36 am, Jon Pounder wrote:
how many cable feet were you ever able to actually get various speeds at ?
Depended on the hardware and wire gauge. I was able to do 1250kbps
symmetrical on a 4kmish loop very reliably.
around here it might just be the geography but I think
Hi,
Anyone has details or information on how to use the SMS command to send SMS
to Fido, Bell Mobility and Rogers Wireless in Canada?
Thanks,
Andre Courchesne
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
From: Anselm Martin Hoffmeister [EMAIL PROTECTED]
Date: Tue, 22 May 2007 13:41:43 +0200
Am Dienstag, den 22.05.2007, 13:21 +0300 schrieb Jonson Player:
Hello,
i just want to activate SMS service between my asterisk local sip
accounts and between asterisk and local sip accounts. How can i do
Tzafrir Cohen escribió:
On Sat, May 19, 2007 at 07:26:33PM +0200, Angel Luis Martinez wrote:
Hi all.
When i do a service zaptel stop on my machine,sometimes it crash and i
must unplug and plug the power cord to restart the machine. Also sometimes
load zttranscode and wct4xxp, and oter times
Buddies,
I am new guy here,I installed Asterisk 1.44 and setup AsteriskNow
manually.Iwant to disable the global digest authentication for
registration so that I
can easily to test my Asterisk system with another call generation tool,how
can I do that?Will appreciate for any replies.Thanks in
Hi Alex,
This is a nice summary. Thanks a lot for your response.
My mere interest was to find out
(1) if a number is a mobile number
(2) If #1 is true, then if I had the carrier name, I could generate an SMS
to the US phone number without asking for the carrier info.
Ritesh
On 5/19/07, Alex
Googling arround I found a number of pocket pc softphones. Of those I was only
able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
Thanks,
Cosmin Prund___
--Bandwidth and Colocation provided by Easynews.com --
Am Dienstag, den 22.05.2007, 17:35 +0300 schrieb Jonson Player:
Thank you for reply. Can you send me some working configs? I'm still
confusing about this sms option.
Just to get you started, try this:
Find out which user asterisk runs as. Get a shell for that user.
Run (all in one line)
smsq
ppciax works too. And it is IAX2 softphone. Anyway, SJPhone is much better.
On 5/22/07, Cosmin Prund [EMAIL PROTECTED] wrote:
Googling arround I found a number of pocket pc softphones. Of those I was
only able to install SJ-something (removed it).
Is there one (pocket pc softphone) that
Am Dienstag, den 22.05.2007, 21:49 +0300 schrieb Cosmin Prund:
Googling arround I found a number of pocket pc softphones. Of those I was
only able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
When I searched for one, about half a year ago, there were
On Tue, 22 May 2007, Ritesh Agrawal said something to this effect:
Hi Alex,
This is a nice summary. Thanks a lot for your response.
My mere interest was to find out
(1) if a number is a mobile number
(2) If #1 is true, then if I had the carrier name, I could generate an SMS
to the US phone
Cosmin Prund wrote:
Googling arround I found a number of pocket pc softphones. Of those I
was only able to install SJ-something (removed it).
SJphone, and why did you remove it?
Is there one (pocket pc softphone) that works?
SJphone ;-) At least I've made some successful calls using
Is the autologoff function supported in Asterisk BE B.1-3? I have
configured my agents.conf with a 5 second timeout, but the agents extension
continues ringing until the call eventually goes to voicemail.
Agents.conf
[general]
persistentagents=yes
[agents]
autologoff = 5
multiplelogin = no
Hi,
I have installed TE212P. Loaded the zaptel modules wc2xxp module for
TE212P.
The span are up I can make a call, but the echo issue exists, so its same
like my old TE110P card.
So I called Digium support. They said that the card may be bad or the modules
are not loaded for
Hi,
Is there any open source pocket pc softphones available. I could find only
one MiniSip that too, it was releasing soon.
Regards
Arpit
On 5/22/07, Remco Post [EMAIL PROTECTED] wrote:
Cosmin Prund wrote:
Googling arround I found a number of pocket pc softphones. Of those I
was only able
Hello all,
Normally I just use pri's with our asterisk systems, but a request came
in to add some normal pots lines to the setup. We have 3 lines, and they
run into the fxs ports. They hit the dialplan just fine, and they always
dial the s extension. However, my question would be... Is there a
I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people on one little
DSL
Hello,
Did someone have a solution for a line fax detection for outgoing call
For exemple
I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B
whats ia want its somthing like AMD application that i use for the
answering machine .
Hello,
Did someone have a solution for a line fax detection for outgoing call
For exemple
I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B
whats ia want its somthing like AMD application that i use for the
answering machine .
Jim Suber wrote:
I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the fact that I got 5 people
David Florella wrote:
Thank you knox. Finally, I have chosen this solution : find
/var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed
every night by the CRON. However, I would have preferred this feature was
implemented in Astrisk.
You should expect this to massively
Does this even work?
exten = 5010,1,Dial(SIP/[EMAIL PROTECTED])
It keeps saying CHANUNAVAIL...
greetz
2007/5/22, Tim Verscheure [EMAIL PROTECTED]:
ok so now I changed ext-local to dundi-ext and I created this
context at the bottom of the extensions file. This is now the case.
You should expect this to massively break voice mailboxes.
Well, it won't massively break them, just a bit. We do this on some
mailboxes and it works OK. The problem is that is you delete message 1
and leave 2, a new message will become 1, thus breaking the sequence.
They will be played back as
I have an application that requires I be able to dial into an asterisk box,
then from there dial out to another user through a PSTN. I'd like to be able
to both 1) record this call and 2) let another user dial in using something
like ChanSpy to listen to the conversation.
I can get this
Sorry to say I have to disagree with you but I just had a heap of old
Voicemails which I couldn't be bothered deleting through my phone, So I
went in to /Old/ and ran rm -f on the first 20, I then had to listen to
another that wasn't deleted and it was still accessible from the phone,
upon further
Anthony Francis wrote:
Jim Suber wrote:
I'm fairly certain that zaptel is not a service.
You might try service asterisk stop
I 'm running CentOS 4.4 as well. The only problem I'm having is sometimes
the extensions don't ring even though the caller hears a ring tone.
I'm thinking maybe it's the
Jim Suber wrote:
I am somewhat confused. I have the incoming (s) context playing a
greeting and callers choose one of two extensions (100, or 101)
To the caller it ALWAYS sounds as though the phone is ringing. However,
sometimes it is not actually ringing the phones
The listext.wav file
Hi!
Googling arround I found a number of pocket pc softphones. Of those I
was only able to install SJ-something (removed it).
Is there one (pocket pc softphone) that works?
Windows Mobile 6 comes with a SIP client, however on my HTC device I
still need to use the speaker phone or a headset,
On Tue, 22 May 2007, Sean M. Pappalardo said something to this effect:
Sure, it's called a DID trunk. It's basically just a regular analog phone
line but the CO switch sends down the digits dialed in one of a few ways:
Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency (DTMF). They
Rob Schall wrote:
Normally I just use pri's with our asterisk systems, but a request came
in to add some normal pots lines to the setup. We have 3 lines, and they
run into the fxs ports. They hit the dialplan just fine, and they always
dial the s extension. However, my question would be... Is
Can anyone guide me to a how to on automating a call?
I know a little piece of code (normally python) has to be place some where
and then a file has to be mv into the spooler.
Where do I get the run down?
I have a button on another application that sends an email and I want it to
also send a
Tim Verscheure wrote:
Does this even work?
exten = 5010,1,Dial(SIP/[EMAIL PROTECTED])
if priv is a sip account it does Yes, I guess you are on the right
track.
It keeps saying CHANUNAVAIL...
greetz
2007/5/22, Tim Verscheure [EMAIL PROTECTED]:
ok so now I changed ext-local to
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