Gustavo,
Are your Linux box sending e-mails? If that is the case, all that you have
to do is put the e-mail address to e-mail on the voicemail.conf
If you don't get e-mails in 95% of the cases is because your box has
sendmail disable, or whatever e-mail software you 're using.
As example:
[root@SIP1-MI asterisk]# cat voicemail.conf
; Voicemail Configuration
;
[general]
format=wav49|gsm|wav
serveremail=seneca_phones@senecacom.net
attach=yes
; How many miliseconds to skip forward/back when rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
;Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
;Max number of failed login attempts
maxlogins=3
; Supported values:
;'filename' filename of a soundfile (single ticks around the
filename
; required)
; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, ...)
; B or b or h Month name (January, February, ...)
; d or e numeric day of month (first, second, ..., thirty-first)
; Y Year
; I or l Hour, 12 hour clock
; H Hour, 24 hour clock (single digit hours preceded by
"oh")
; k Hour, 24 hour clock (single digit hours NOT preceded by
; "oh")
; M Minute
; P or p AM or PM
; Q "today", "yesterday" or ABdY (*note: not
standard strftime
; value)
; q "" (for today), "yesterday", weekday, or
ABdY (*note: not
; standard strftime value)
; R 24 hour time, including minute
;
;
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H
'digits/hundred' M
'hours'
;
; Each mailbox is listed in the form
;
<mailbox>=<password>,<name>,<email>,<pager_email>,<options>
; if the e-mail is specified, a message will be sent when a message is
; received, to the given mailbox. If pager is specified, a message will be
; sent there as well.
;
[default]
1234 => 4242,Example Mailbox,root@localhost
2486950002 => 1234,Carlos Alperin,carlos@senecacom.net
3311 => 1234,Carlos Alperin,carlos@senecacom.net
3312 => 1234,Oscar Mendez,carlos@senecacom.net
3321 => 1234,Marina Veron,marina_veron@hotmail.com
3322 => 1234,Mateo Siporin,carlos@senecacom.net
;
;[test]
;3311 => 1234,Carlos Alperin,carlos@senecacom.net
;3312 => 1234,Carlos Alperin,carlos@senecacom.net
;3321 => 1234,Carlos Alperin,carlos@senecacom.net
;3322 => 1234,Carlos Alperin,carlos@senecacom.net
7710726 => 0726,Test Test,calperin@senecacom.net
7710727 => 0727,Carlos Carlos,calperin@senecacom.net
This is a voicemail.conf of a group in my test asterisk server.
Saludos,
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
Calperin@senecacom.net
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gustavo A.
Gonzalez
Sent: Monday, July 18, 2005 4:16 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Mail Notification
Hi all!, i search for some information about to setup my asterisk box with
e-mail notification when a I call the voicemail application. Voicemail
application works fine in the Dial Plan but nothing happens with email
notification ...so what i need to know about this?...wiki pages did not help
me ....thanks!
G.
----- Original Message -----
From: <asterisk-users-request@lists.digium.com>
To: <asterisk-users@lists.digium.com>
Sent: Monday, July 18, 2005 2:00 PM
Subject: Asterisk-Users Digest, Vol 12, Issue 117
Send Asterisk-Users mailing list submissions to
asterisk-users@lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request@lists.digium.com
You can reach the person managing the list at
asterisk-users-owner@lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. Asterisk Comedian Web page login (Kurt Pasewaldt)
2. Asterisk @ Home incoming CID (maoleson@mchsi.com)
3. massive outbound calling... (Goolsby, Daniel S (Daniel))
4. Re: SpanDSP+astfax with multiple fax pages (Lee Howard)
5. Re: System Jsut hangs Up (sylvain garcia)
6. Re: Iaxy and Echo (Adam Goryachev)
7. Re: Asterisk/Hylafax <=> Receive/Send faxes (Lee Howard)
8. Re: Teliax to VoIPJet (Andrew Latham)
9. IP Trunking for LD? (Matthew S. Krawitz)
10. Re: swissvoice (Doug Lytle)
11. Re: Iaxy and Echo (Aaron with Morad)
12. long pause on dialing.. (Goolsby, Daniel S (Daniel))
13. Comments on Areski Calling Card Solution plz (Arnd Vehling)
14. IAX register confusion (David Cook)
15. Transcoding problems (Martin Sutherland)
16. Re: Asterisk@home not accepting IAX calls from outside
(Mark Phillips)
17. Codecs and bandwidth (Tim Pushor)
18. RE: Teliax to VoIPJet (Wiley Siler)
19. Re: long pause on dialing.. (Randy Williams)
20. RE: swissvoice (Florian Overkamp)
21. Re: long pause on dialing.. (Giorgio Incantalupo)
22. Re: Memory leak in asterisk CVS (Erik Espinoza)
23. Re: SoftPhones: Bad, or just bad QoS? (Time Bandit)
24. Re: long pause on dialing.. (Randy Williams)
----------------------------------------------------------------------
Message: 1
Date: Mon, 18 Jul 2005 11:29:23 -0400
From: Kurt Pasewaldt <kurtwp@gmail.com>
Subject: [Asterisk-Users] Asterisk Comedian Web page login
To: Asterisk <asterisk-users@lists.digium.com>
Message-ID: <723ac8b605071808296d5c212a@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
When I try to login into voicemail through the web interface It states
incorrect login.
In my voicemail.conf I have all voicemail boxes set under local. I
changed the symbolic
link to reflect the new directory under /var/spool/asterisk. Am I
missing something?
My vm link = /var/spool/asterisk/voicemail/local.
Kurt
------------------------------
Message: 2
Date: Mon, 18 Jul 2005 15:38:12 +0000
From: maoleson@mchsi.com
Subject: [Asterisk-Users] Asterisk @ Home incoming CID
To: asterisk-users@lists.digium.com
Message-ID:
<071820051538.14724.42DBCCE40008B32D00003984219792676102019C0A04010E03@mchsi
.com>
OK, here is the scenario, Asterisk @ Home 1.0 with TDM04B and TDM40B. I can
receive and place calls with no issues, however, when I receive a call, the
CID
only shows "Analog Line" on the Grandstream 2000XP phone. Does anyone
have
any
ideas even where to look to change this?? Is it a setting in the phone,
Asterisk, or both??
Thanks,
Marc
------------------------------
Message: 3
Date: Mon, 18 Jul 2005 10:40:25 -0500
From: "Goolsby, Daniel S (Daniel)" <daniel.goolsby@mci.com>
Subject: [Asterisk-Users] massive outbound calling...
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID:
<AFB16F9873513044B5D75AB44FF81A23041F866A@skyteles2.us.skytel.com>
Content-Type: text/plain; charset=us-ascii
Does anyone know what kind of limitations asterisk has when it comes to
massive outbound dialing.. i.e. how many sip/iax phones could be dialed
at the same time-- and if someone answered, play a .wav file?
Or outbound throughput on zaptel devices?
Say if I had a dual xeon with 2 quad t1 cards, hooked up to a 100mbit
lan. Anyone know how many it could actually sustain w/o the voice file
being distorted on playback?
Daniel
------------------------------
Message: 4
Date: Mon, 18 Jul 2005 08:38:19 -0700
From: Lee Howard <faxguy@howardsilvan.com>
Subject: Re: [Asterisk-Users] SpanDSP+astfax with multiple fax pages
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBCCEB.9030600@howardsilvan.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Paul van Brouwershaven wrote:
> HylaFAX can (we ar doing this now), but not with E1 or T1. So you can
> only send with a maximum 2 channels. (with two default analog modems)
HylaFAX can use E1 and T1 fax modems just fine (24 and 30 channels each).
Furthermore, HylaFAX also supports multiport modems (usually up to 8
ports each).
But, yes, if you are only going to use internal ISA or external modems
connected to the motherboard's serial ports, then yes, you're limited to
two modems. But that's completely ignoring the possibilities of using
PCI and USB modems.
Lee.
------------------------------
Message: 5
Date: Mon, 18 Jul 2005 17:39:45 +0200
From: sylvain garcia <sylvain.garcia@anyware-tech.com>
Subject: Re: [Asterisk-Users] System Jsut hangs Up
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBCD41.8080201@anyware-tech.com>
Content-Type: text/plain; charset="iso-8859-1"
Tim King a ?crit :
> I took care of my earlier problem. But now if I call in it just says
> goodbye, And on my extension no matter what I do it seems to just hang
> up on me immediately. It's a slackware 10.1 box with Digium 22b card.
> I am running AMP so its mysql driven. I'm not seeing any errors. It
> just hangs up.
>
>
>
> Tim King
>
> Network Engineer
>
> Computer & Network Solutions LLC
>
> 1331 Plainfield Ave
>
> Grand Rapids MI 49505
>
>
>
> Phone: 800-669-3290
>
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Could you describe your problem with your extensions.conf
And send me your email please
sorry for my english i'm french
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------------------------------
Message: 6
Date: Tue, 19 Jul 2005 01:46:09 +1000
From: Adam Goryachev <mailinglists@websitemanagers.com.au>
Subject: Re: [Asterisk-Users] Iaxy and Echo
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1121701569.1611.19.camel@watcher>
Content-Type: text/plain
On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad
wrote:> I have been searching for a while and can't find anything specific
> like this.
>
> Here's is my setup:
>
> IAXy -- broadband network -- Asterisk -- TE110P -- Channel
> Bank -- POTS lines (FXO)
>
> Everything works fine except for the echo at the IAXy. There is no
> echo on the POTS end, so Asterisk is doing a good job of echo
> canceling. Is there any provisioning in the IAXy to do echo
> canceling?
If you get echo on the IAXy end, then asterisk is NOT doing it's echo
cancellation function fully. The POTS user will never get echo if you
don't generate any, or their echo cancellation is functioning correctly.
So, you need to tune the echo cancellation at your asterisk box (or
perhaps you can do that in your channel bank?? I dunno how clever those
things are)...
Regards,
Adam
------------------------------
Message: 7
Date: Mon, 18 Jul 2005 08:47:37 -0700
From: Lee Howard <faxguy@howardsilvan.com>
Subject: Re: [Asterisk-Users] Asterisk/Hylafax <=> Receive/Send faxes
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBCF19.8040203@howardsilvan.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Jian Hong GUAN wrote:
> Can you tell me how to configure Hylafax + Asterisk in order to be
> able to receive/send faxes.
If you have an incoming T1/E1 line:
telco --> T1/E1 --> TE405P/TE410P --> Asterisk --> TE405P/TE410P
(another port) --> T1/E1 fax modem --> HylaFAX
or:
telco --> T1/E1 --> TE405P/TE410P --> Asterisk --> TE405P/TE410P
(another port) --> channel bank --> analog fax modem --> HylaFAX
If you don't have an incoming T1/E1 line (you've just got analog lines
coming in) then just get yourself another analog line for your analog
fax modem and bypass Asterisk altogether. Look up the archives for
"TDM" and "fax" to get a synopsis as to why you don't
want to run fax
through Asterisk on analog channels.
Lee.
------------------------------
Message: 8
Date: Mon, 18 Jul 2005 10:50:34 -0500
From: Andrew Latham <lathama@gmail.com>
Subject: Re: [Asterisk-Users] Teliax to VoIPJet
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <c39c115d05071808505e7e0279@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
HUH? Why?
If you are having Cellphones dialed for the user its one thing but
what is the goal....
On 7/18/05, code select <code.select@gmail.com>
wrote:> I'm trying to setup asterisk to accept call from Teliax, request the
> 10 digit number from user, then dial it thru the VoIPJet. If I'm not
> wrong I will be charged by both providers because both connection is
> active during conversation. So my question is can I set the things so
> that I pay only to VoIPJet? Specific configuration snippets will be
> greatly appeciated.
>
> Thank you.
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
<sig>
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: lathama@lathama.com - lathama@yahoo.com - lathama@gmail.com
If any of the above are down we have bigger problems than my email!
</sig>
------------------------------
Message: 9
Date: Mon, 18 Jul 2005 11:57:55 -0400
From: "Matthew S. Krawitz" <suntigen@gmail.com>
Subject: [Asterisk-Users] IP Trunking for LD?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <E028F6FC-7D31-4860-AF5C-F9AC933DF74E@gmail.com>
Content-Type: text/plain; charset="us-ascii"
I'm sure this topic has been discussed to death, but I haven't found
a comprehensive answer yet...
I have a very large installation of Cisco Call Managers connecting
directly local and LD T1's for service.
I would like to replace some of my LD T1's with IP trunks (or
something like that). I would need low-cost domestic and
international LD... but quality and reliability is our top priority,
so we're not simply looking for low-bid. I assume IAX2 is the
protocol I should use, but who are the major players in providing
this type of service?
Thanks!
- matthewk (MSK2)
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------------------------------
Message: 10
Date: Mon, 18 Jul 2005 12:04:49 -0400
From: Doug Lytle <support@drdos.info>
Subject: Re: [Asterisk-Users] swissvoice
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBD321.2050509@drdos.info>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
thomas DEILLON wrote:
>Hello,
>
>I have swissvoice phones and when i use one, a have in asterisk lines like:
>Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning negative timestamp
>-13691.-232125
>
>have a idea ?
>
>
>
>
Yes, Kevin said this earlier today:
2 wrote:
> i get lots of the below from friday 15.7.05 cvs as well
>
> ERROR[1171] UTILS.C:509 TVFIX: WARNING NEGATIVE TIMESTAMP -194931. ...
I will be looking into this issue later today.
Doug
------------------------------
Message: 11
Date: Mon, 18 Jul 2005 10:04:23 -0600
From: "Aaron with Morad" <aaronc@morad.ab.ca>
Subject: Re: [Asterisk-Users] Iaxy and Echo
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <00f001c58bb2$5def98b0$7702a8c0@aaron>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
Thanks Adam. My channel banks are pretty old (NEC ND4's) so they don't
do
anything for echo. I'll have to try tweaking Asterisk.
Aaron
----- Original Message -----
From: "Adam Goryachev" <mailinglists@websitemanagers.com.au>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Monday, July 18, 2005 9:46 AM
Subject: Re: [Asterisk-Users] Iaxy and Echo
> On Mon, 2005-07-18 at 09:25 -0600, Aaron with Morad wrote:
>> I have been searching for a while and can't find anything specific
>> like this.
>>
>> Here's is my setup:
>>
>> IAXy -- broadband network -- Asterisk -- TE110P -- Channel
>> Bank -- POTS lines (FXO)
>>
>> Everything works fine except for the echo at the IAXy. There is no
>> echo on the POTS end, so Asterisk is doing a good job of echo
>> canceling. Is there any provisioning in the IAXy to do echo
>> canceling?
>
> If you get echo on the IAXy end, then asterisk is NOT doing it's echo
> cancellation function fully. The POTS user will never get echo if you
> don't generate any, or their echo cancellation is functioning
correctly.
>
> So, you need to tune the echo cancellation at your asterisk box (or
> perhaps you can do that in your channel bank?? I dunno how clever those
> things are)...
>
> Regards,
> Adam
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 12
Date: Mon, 18 Jul 2005 11:08:19 -0500
From: "Goolsby, Daniel S (Daniel)" <daniel.goolsby@mci.com>
Subject: [Asterisk-Users] long pause on dialing..
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID:
<AFB16F9873513044B5D75AB44FF81A23041F866B@skyteles2.us.skytel.com>
Content-Type: text/plain; charset=us-ascii
I have an Asterisk setup with AMP installed. I have phone extensions
from 7000 to 7010.
I experience long delays when dialing a 9 digit number as opposed to a
10-digit number. How do you get around not having to press the # key to
speed up the dialing process? For any length phone number for that
matter-- like dialing another extension.
If I dial 7005, I'll have to wait a while.. but it's instant when I
press the # key.
Daniel
------------------------------
Message: 13
Date: Mon, 18 Jul 2005 18:17:37 +0200
From: Arnd Vehling <av@nethead.de>
Subject: [Asterisk-Users] Comments on Areski Calling Card Solution plz
To: Asterisk Users <asterisk-users@lists.digium.com>
Message-ID: <42DBD621.4020403@nethead.de>
Content-Type: text/plain; charset=us-ascii; format=flowed
Hi,
can anyone who has the Areski Calling Card solution on Asterisk
working comment on it? Is is stable enough for a production system?
Any pros and cons?
thx,
Arnd
------------------------------
Message: 14
Date: Mon, 18 Jul 2005 12:03:28 -0400
From: David Cook <dbc_asterisk@advan.ca>
Subject: [Asterisk-Users] IAX register confusion
To: asterisk-users@lists.digium.com
Message-ID: <1121702608.42dbd2d07532c@advan.ca>
Content-Type: text/plain; charset=ISO-8859-1
I have been unable to understand the connection between an IAX
registration for dynamic IP assignment and and the host definition.
I have signed up with an ITSP for a DID. My ip is dynamic and although I
have a dynamic DNS name, we are registering and outbound works fine.
I'm at a loss to understand the relationship between the registration
and the [section] definition in iax.conf that will allow me to specify
my context for inbound calls and deal with the inbound DID.
For example:
register => myuser:mypasswd@my.itsp.com
;OK. This part works fine. My dial statement calls
; exten =>
_NXXNXXXXXX,3,Dial,IAX2/myuser:mypasswd@my.itsp.com/${EXTEN},45,tr)
;
; VoIP Local service from myitsp
;[something] ???
[LO_TRNK_MYSWITCH]
type=peer
host=dynamic
context=from-myitsp
secret=mypasswd
qualify=3000
; How do I construct this entry? I would _like_ the entry to be labelled
; LO_TRNK_MYSWITCH so I can maintain a naming convention that makes
; sense.
; How do I associate this with the inbound itsp so the calls come into
; the "s" extension in a particular context so I can deal with the
DID?
I simply don't see how I associate the inbound stream with my section
heading?
Thanks, dbc.
--
David Cook
------------------------------
Message: 15
Date: Mon, 18 Jul 2005 17:14:18 +0100
From: "Martin Sutherland" <martin@ukgastech.co.uk>
Subject: [Asterisk-Users] Transcoding problems
To: <asterisk-users@lists.digium.com>
Message-ID: <s2dbe38f.025@aztec.ukgastech.co.uk>
Content-Type: text/plain; charset=US-ASCII
I have just purchased 20 licenses for the G729a codec from digium and set
about changing the defaults to use this codec in all cases to reduce the
bandwidth requirements (all my SIP devices support this codec). To my dismay
I then find that calls coming from SIP devices to the outside via the
chan_capi channel no longer work and give the message "no translator path
exists for channel type CAPI (native 8) to 256". I understood that Asterisk
would transcode between different codecs? In fact only two of my SIP users
are allowed access to the outside via a BRI interface, but I now have to set
them to always use g711a/u just in case they make a call via the chan_Capi.
I am using chan_capi-cm-0.5.3
------------------------------
Message: 16
Date: Mon, 18 Jul 2005 12:21:47 -0400
From: Mark Phillips <g7ltt@g7ltt.com>
Subject: Re: [Asterisk-Users] Asterisk@home not accepting IAX calls
from outside
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBD71B.8080401@g7ltt.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Problem solved. Wrong context supplied by me - doh!!
Mark Phillips wrote:> I've been banging my head with this all day.
>
> I today switched from a very old CVS build to AAH1.3 and so far
> everything has been easy. However I cannot accept calls from a
> previously working IAX trunk.
>
> I've set up an trunk with all the same credentials as before and can
> call the folks at the other pbx. However whenever they call me I tell
> them that I don't have an extension/context by the name they dialed.
>
> Any ideas?
>
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
------------------------------
Message: 17
Date: Mon, 18 Jul 2005 10:22:27 -0600
From: Tim Pushor <timp@crossthread.com>
Subject: [Asterisk-Users] Codecs and bandwidth
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBD743.80803@crossthread.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses 64K for bidirectional communication
(using ulaw, correct?), but on several occasions now have seen
references to ulaw voip conversations using 64K per side of the
conversation, plus packet overhead
(http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down
now - plus other references) for a total of over 128K per ulaw 'full
duplex' voice conversation?
Thanks
Tim
------------------------------
Message: 18
Date: Mon, 18 Jul 2005 09:34:11 -0700
From: "Wiley Siler" <wsiler@education2020.com>
Subject: RE: [Asterisk-Users] Teliax to VoIPJet
To: "Andrew Latham" <lathama@gmail.com>, "Asterisk Users
Mailing List
- Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Message-ID:
<2036A1C10A032D41BE3A2AEBAF1999573AFA4B@E2DSAZ.education2020.com>
Content-Type: text/plain; charset="US-ASCII"
This sounds like DISA which is great for saving bucks on LD if used
right...
You will still need two channels and thus it will still cost for both
legs...
Nature of the beast...
W
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew
Latham
Sent: Monday, July 18, 2005 8:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Teliax to VoIPJet
HUH? Why?
If you are having Cellphones dialed for the user its one thing but
what is the goal....
On 7/18/05, code select <code.select@gmail.com>
wrote:> I'm trying to setup asterisk to accept call from Teliax, request the
> 10 digit number from user, then dial it thru the VoIPJet. If I'm not
> wrong I will be charged by both providers because both connection is
> active during conversation. So my question is can I set the things so
> that I pay only to VoIPJet? Specific configuration snippets will be
> greatly appeciated.
>
> Thank you.
> _______________________________________________
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> Asterisk-Users@lists.digium.com
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--
<sig>
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
WWW: http://lathama.com
Email: lathama@lathama.com - lathama@yahoo.com - lathama@gmail.com
If any of the above are down we have bigger problems than my email!
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------------------------------
Message: 19
Date: Mon, 18 Jul 2005 12:34:51 -0400
From: Randy Williams <randyw@techsource.com>
Subject: Re: [Asterisk-Users] long pause on dialing..
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>, "Goolsby, Daniel S (Daniel)"
<daniel.goolsby@mci.com>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1121704491.42dbda2bac806@kinesis.swishmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Greetings,
This may be an artifact of the particular phone you are using. I know that
both
Grandstream and SNOM products allow you to set a timeout for
"auto-dial"
(which
is how long to wait after the last button press before dialing the number
present).
I have my units set to three settings:
2 seconds for the receptionist
5 seconds for most everyone else
30 seconds for some of our elder employees who need extra time while
transcribing a phone number
Check your phone settings to see if there is something you can set.
However, there may be something else at fault...
RandyW
Quoting "Goolsby, Daniel S (Daniel)" <daniel.goolsby@mci.com>:
> I have an Asterisk setup with AMP installed. I have phone extensions
> from 7000 to 7010.
>
> I experience long delays when dialing a 9 digit number as opposed to a
> 10-digit number. How do you get around not having to press the # key to
> speed up the dialing process? For any length phone number for that
> matter-- like dialing another extension.
>
> If I dial 7005, I'll have to wait a while.. but it's instant when I
> press the # key.
>
> Daniel
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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------------------------------
Message: 20
Date: Mon, 18 Jul 2005 18:32:24 +0200
From: "Florian Overkamp" <florian@obsimref.com>
Subject: RE: [Asterisk-Users] swissvoice
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <E1DuYY5-0007eH-00@clio>
Content-Type: text/plain; charset="US-ASCII"
Hi,
> -----Original Message-----
> I have swissvoice phones and when i use one, a have in
> asterisk lines like:
> Jul 18 17:16:22 ERROR[15251]: utils.c:509 tvfix: warning
> negative timestamp
> -13691.-232125
> the swissvoice firmware is IP10 SP v1.0.0 (Build 11) and
> asterisk version is
> the cvs of 18 july 2005 (today).
Swissvoice phones tend to have a few interesting side effects in their rtp
timestamping, we have filed some issues on that. However, it would be fun to
hear what the actual problem is you are experiencing :-)
Best regards,
Florian
------------------------------
Message: 21
Date: Mon, 18 Jul 2005 18:40:21 +0200
From: Giorgio Incantalupo <gincantalupo@fgasoftware.com>
Subject: Re: [Asterisk-Users] long pause on dialing..
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <42DBDB75.4060308@fgasoftware.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
it is hard to answer without the right piece of extensions.conf but
remember there is a digit timeout in Asterisk: you enter 9 digits but in
the dialplan there is a match for 10 digits so how can Asterisk know if
you want to call the 9-digits number or the 10-digits? After 9 digits it
waits for a while...if another digit is dialed then it can call the
10-digit number otherwise it calls the 9-digits number. You can lower
Asterisk digit timeout but remember that not all users are so fast to
dial...
Giorgio.
Goolsby, Daniel S (Daniel) wrote:
>I have an Asterisk setup with AMP installed. I have phone extensions
>from 7000 to 7010.
>
>I experience long delays when dialing a 9 digit number as opposed to a
>10-digit number. How do you get around not having to press the # key to
>speed up the dialing process? For any length phone number for that
>matter-- like dialing another extension.
>
>If I dial 7005, I'll have to wait a while.. but it's instant when I
>press the # key.
>
>Daniel
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
------------------------------
Message: 22
Date: Mon, 18 Jul 2005 09:29:47 -0700
From: Erik Espinoza <erik.espinoza@gmail.com>
Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <b86db13f050718092949538912@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Hi Walter,
I had high load and extreme memory usage on my machine. My machine
wasn't running on SMP. My point was that the cvs version you were
using contained some bad patches, and it was probably a good idea to
upgrade or move to stable.
Thanks,
Erik
On 7/18/05, Walter Klomp <walter@aglow.com.sg>
wrote:> Hi Erik,
>
> You put me to a page which refers to high load on CPU on SMP. Nothing to
do> with memory leak. Furthermore I am not running SMP.
>
> Any other suggestions in which direction to look? Am I the only one
> experiencing this ?
>
> Do you mean if I update to the today's CVS the memory leak issue will
be
> resolved ?
>
> Thanks
> Walter
>
> --- Original Message below ---
>
> Message: 20
> Date: Sat, 16 Jul 2005 21:42:44 -0700
> From: Erik Espinoza <erik.espinoza@gmail.com>
> Subject: Re: [Asterisk-Users] Memory leak in asterisk CVS
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <b86db13f0507162142593096f@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Known issue. This was reverted later.
>
> Check the thread on the mailing list
>
> http://lists.digium.com/pipermail/asterisk-users/2005-July/116246.html
>
> Thanks,
> Erik
>
> On 7/16/05, Walter Klomp <walter@aglow.com.sg> wrote:
> > Hi,
> >
> > My Asterisk CVS is apparently not doing much (other than keeping SIP
&
> > IAX2 registrations alive and doing some ZAP calls (without
> > echo-cancellation), but slowly the memory is filling up, so much so
that
> > 100m virtual memory is used up within 12 hours and I have to restart
the
> > asterisk application every 48 hours to make sure I have enough
memory...
> >
> > How can I help resolve this problem?
> >
> > Problem occurs on both Sangoma and Digium installed systems. Fedora
Core
> > 3 and Centos 4.1 don't make a difference either.
> >
> > My version is Asterisk CVS-HEAD built on a i686 running Linux on
> > 2005-07-11 16:29:02
> >
> > I have removed the mailbox entries in my sip.conf which greatly
reduced
> > this problem. So, I suspect it may be in the sip or iax channel
> application.
> >
> > I also run quite a bit of agi scripts but none of them were
"alive" when
> > these memory-usage increases as shown below over a 1 minute interval
> > with only 4 zap channels alive (2 calls) occured:
> >
> > ps -AF output... using this script:
> > n=1;while [ 1 ]; do i=`ps -AF|grep ast|grep sbin|grep -v grep`;
m=`echo
> > $i|cut -f 6 -d\ `;if [ `echo $m` -ne `echo $n` ]; then echo $i;
n=`echo
> > $m`;fi;done
> >
> > root 15875 26881 0 15727 46240 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> > root 15875 26881 0 15725 46248 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> > root 15875 26881 0 15725 46256 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> > root 15875 26881 0 15725 46268 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> > root 15875 26881 0 15725 46280 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> > root 15875 26881 0 15725 46288 0 10:01 ? 00:00:00 /usr/sbin/asterisk
-fp
> >
> > Hope we can fix this somehow.
> >
> > Walter Klomp
> > Singapore.
> >
> >
>
>
>
>
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------------------------------
Message: 23
Date: Mon, 18 Jul 2005 12:54:41 -0400
From: Time Bandit <timebandit001@gmail.com>
Subject: Re: [Asterisk-Users] SoftPhones: Bad, or just bad QoS?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1e2050d505071809546c963237@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
> This is software. Use manageble software. If "software" means
separate
> setup on each desktop, then don't use it. If you spend that much time
on
> setting up phones, imagine how long it takes you to update other
> software packages. This is, then, a symptom of a general problem.
I would like to implement central management in my softphone. What
would be the best way to accomplish this ?
Currently, all the settings are stored in the registry under
HKEY_CURRENT_USER. So, if you use a roaming profile, the settings
follow you.
I would appreciate people's input on what would be desirable, and I'll
try to implement it so it would be more easy to manage.
Thanks
------------------------------
Message: 24
Date: Mon, 18 Jul 2005 12:34:51 -0400
From: Randy Williams <randyw@techsource.com>
Subject: Re: [Asterisk-Users] long pause on dialing..
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>, "Goolsby, Daniel S (Daniel)"
<daniel.goolsby@mci.com>
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1121704491.42dbda2bac806@kinesis.swishmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Greetings,
This may be an artifact of the particular phone you are using. I know that
both
Grandstream and SNOM products allow you to set a timeout for
"auto-dial"
(which
is how long to wait after the last button press before dialing the number
present).
I have my units set to three settings:
2 seconds for the receptionist
5 seconds for most everyone else
30 seconds for some of our elder employees who need extra time while
transcribing a phone number
Check your phone settings to see if there is something you can set.
However, there may be something else at fault...
RandyW
Quoting "Goolsby, Daniel S (Daniel)" <daniel.goolsby@mci.com>:
> I have an Asterisk setup with AMP installed. I have phone extensions
> from 7000 to 7010.
>
> I experience long delays when dialing a 9 digit number as opposed to a
> 10-digit number. How do you get around not having to press the # key to
> speed up the dialing process? For any length phone number for that
> matter-- like dialing another extension.
>
> If I dial 7005, I'll have to wait a while.. but it's instant when I
> press the # key.
>
> Daniel
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
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End of Asterisk-Users Digest, Vol 12, Issue 117
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