Re: [asterisk-users] Source Based Call Routing

[Ron Arts]

Daniel,

attach a dialplan variable to each extension using setvar
in sip.conf:

[6318]
type=friend
username=6318
secret=xx
host=dynamic
nat=no
dtmfmode=rfc2833
qualify=0
amaflags=billing
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=phone
setvar=__usetrunk=1

you can use the ${usetrunk} variable in your dialpan.

Ron


Daniel Cole wrote:

Hi List,

I have a scenario that I want to try out (we potential have a client who would 
need this), but I am as of yet unable to find much help with it.

What we want to do is have an asterisk box with a large number of extensions 
(1000+). This asterisk box will have approximately 3 SIP trunks setup back to 
providers. What we want to do is to be able to define groups of extensions that 
use specific outbound trunks.

Approximately a third of the extensions will one the first trunk, a third the 
second trunk, and the rest will use the last trunk. We also need control over 
assigning with trunks the given extensions will use.

Any suggestions on how to get this to work would be very much appreciated.


Many Thanks,

Daniel

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[asterisk-users] Problem with DTMF dialing

[Ian]

Hi all

I have a small problem here. I asked this question on another asterisk 
mailing list, but nobody seemed to be able to help me there.


We are running

   * Asterisk 1.4.17
   * Libpri 1.4.3
   * Zaptel 1.4.8

on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo 
cancelation and a quad FXO card.


We have 4 analog lines, one of which is a Cellphone line for least cost 
routing.


The  problem I am having is dialing out using DTMF signalling. At the 
moment I am making do with Pulse dialing through the 3 analog lines. I 
can recieve calls on the Cellphone line without any problems, but cant 
dial out through it, as a cellphone cant do pulse dialing. I have run 
"ztmonitor 1 -f gains", where 1 is the zap channel where the cellphone 
is located, while dialing the number 072 031 1294. I then went to 
audacity, on my own pc, and converted the raw file into mp3 format, 
which is available for download at 
http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the 
playback I concluded that the DTMF signals being sent is totally wrong.


The relevant pieces of my configs are below

Your help in this matter will be greatly apreciated.

Regards
Ian

--
www.vddi.co.za 
I Coetzee
IT Technician
Telephone   :   012 664 2300
Cellphone   :   079 522 6519
Fax :   012 644 2902
E-mail  :   [EMAIL PROTECTED]
Skype   :   vddb_igcoetzee


*/etc/asterisk/zapata.conf*
; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
;;; line="1 WCTDM/0/0"
;Cellphone
signalling=fxs_ks
callerid=asreceived
context=incoming_calls
callerid=
group=2
busydetect=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
pulsedial=no
callprogress=yes
busycount=5
toneduration=500
subscribecontext=GXP_BLF
overlapdial=no
channel => 1


;;; line="2 WCTDM/0/1"
;Landline
signalling=fxs_ks
callerid=asreceived
context=incoming_calls
callerid=
group=1,2
busydetect=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
pulsedial=yes
callprogress=yes
busycount=5
toneduration=300
subscribecontext=GXP_BLF
channel => 2

*/etc/zaptel.conf*
# Autogenerated by /usr/sbin/zapconf on Wed Jan 16 12:23:09 2008 -- do 
not hand edit

# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
fxsks=1
fxsks=2
fxsks=3
fxsks=4
# channel 5, WCTDM/0/4, no module.
# channel 6, WCTDM/0/5, no module.
# channel 7, WCTDM/0/6, no module.
# channel 8, WCTDM/0/7, no module.

# Global data

loadzone= za
defaultzone = za*
*
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Re: [asterisk-users] Queue - ${ANSWEREDTIME}

[Paul Hales]

It doesn't actually work at all - I tried, and even logged a bug with
digium with no luck. :(

Are the queue logs not quite good enough?

PaulH


On Tue, 2008-01-29 at 17:20 -0800, Johnny Tam wrote:
> How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs
> up, I can calculate how much time the each call lasts?
> 
> Thanks
> 
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[asterisk-users] sipsock_read: BAD! BAD! BAD!

[Douglas Garstang]

Does anyone know the cause of these BAD BAD BAD messages?
I think I lost all my calls when it happened too. We have nagios running 
against IAX and nagios reports that IAX is down. It would seem that the entire 
application locks up when this happens and calls are dropped.

Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846)
Verbosity is at least 3
flexo*CLI> show channels
Channel  Location State   Application(Data) 
SIP/teleglobe-09f887 (None)   Down(None)
1 active channel
12 active calls
Jan 30 03:28:13 WARNING[2671]: channel.c:781 channel_find_locked: Avoided 
deadlock for '0x9f6a6f0', 10 retries!
Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get 
the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] 
Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 
Jan 30 03:28:14 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD!
Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get 
the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] 
Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 
Jan 30 03:28:15 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD!
Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get 
the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] 
Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 
Jan 30 03:28:16 ERROR[26983]: chan_sip.c:11453 sipsock_read: BAD! BAD! BAD!
Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11451 sipsock_read: We could NOT get 
the channel lock for SIP/teleglobe-09f83250 - Call ID [EMAIL PROTECTED] 
Jan 30 03:28:19 ERROR[26983]: chan_sip.c:11452 sipsock_read: SIP MESSAGE JUST 
IGNORED: BYE 

Doug.






  

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Re: [asterisk-users] SIP DTMF Troubleshoot

[Andrew Joakimsen]

Everything seems find on my end. Here's the setup:

Linksys SPA922 <-> Asterisk 1.4 <---> Quintum T1 gateway

Between Asterisk and Quintum if I use G729 RFC2833 DTMF works with no
issues, however if I use uLaw this is where there is a problem. For
some reason the Quintum gateway does not support uLaw + RFC2833.

Also does not matter if I use Asterisk 1.2 or a grandstream or the
proverbial SIP tin can; The scenario is always the same.


On Jan 28, 2008 7:03 PM, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> I think your best bet is to do a packet capture and look for RTP packets
> with an RTP Event payload ("rtpevent" display filter).
>
>
> On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
>
> > How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
> > messages related to DTMF... or if I just do a global SIP debug for
> > that matter I am using RFC DTMF but it's not being passed to the
> > PSTN and I need to debug this further. I've tried to increase the
> > verbosity and the debug ('set debug n') and that didn't help either. I
> > assume this is because even RFC2833 sends the DTMF as RTP which
> > wouldn't show up anyways but how to troubleshoot DTMF issues?
> >
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> >
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: +1-678-954-0670
> Direct : +1-678-954-0671
>
>
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Re: [asterisk-users] asterisk gateway

[Andrew Joakimsen]

Any GSM 900/1800 gateway will work with a Nextel (US) SIM card.

However I assume you actually want to register on a local iDEN network
and not be roaming internationally (Nextel does not have any GSM
roaming partners in the US) That is not possible.

On Jan 29, 2008 9:34 PM, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> Hello everybody
>
> Anyone, to know a gateway that works with nextel simm cards?
> I'm looking for them, in internet, but I did'n look.
>
> Best regards
>
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[asterisk-users] Best practice security for internet access to Asterisk

[Duncan Turnbull]

Hi All

For the scenario of a single asterisk server that needs to serve clients 
on the net, as well as local office clients, I would be very interested 
in people's views of the best method to handle security to prevent net 
based attacks while still allowing the client access.

Some of the challenges I see are:
- preventing brute force and bot type attacks
- monitoring for unusual events and notifying and acting appropriately
- limiting damage if someone does get in
- avoiding a Denial or degradation of service on your asterisk platform
- making it easy for staff to use

Some of this can be done with
- firewall control - but its hard to limit where your clients will come 
from, besides restricting ports
- scripts monitoring logs, I saw a recipe for checking password failures 
then blocking that ip after x failures, I imagine this could get quite 
sophisticated
- using separate restrictions for offnet users but this kind of makes it 
harder for the staff members.
- using a proxy in front of asterisk for SIP, to limit the available 
extensions and minimise the scanning impact on the asterisk box. I am 
hoping this could detect and prevent illegitimate or poorly formed 
requests or unknown user agents. Staff should be using a standard set.
- using iax softclients to shift the attack requirements - I don't know 
much about how well these work
- running all clients over a vpn e.g open vpn, but this is not so good 
for wireless handsets or other devices that can't do a vpn

I am interested in all views and recommendations

Thanks very much

Cheers Duncan

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[asterisk-users] asterisk gateway

[Carlos Rojas]

Hello everybody

Anyone, to know a gateway that works with nextel simm cards?
I'm looking for them, in internet, but I did'n look.

Best regards
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Re: [asterisk-users] When can I AIG?

[Steve Totaro]

I just save a bunch of money on my car insurance by switching to AIG!!!

Sorry, couldn't resist,
Steve Totaro

On Jan 29, 2008 8:11 PM, Guilherme Loch Waltrick Góes <[EMAIL PROTECTED]> wrote:
> Basically you initiate the call via a Originate method inn your Java App and
> bridge it to your AGI. Follow the documentation on the site and you should
> be in a good path.
>
>  Best Regards,
>
>
>
> On Jan 29, 2008 7:01 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > Hey there,
> >
> >
> >
> > I've actually looked through the site a bunch and found some great
> information. The thing I'm missing, and I think it's because of my lack of
> experience with Asterisk and setting up a dial plan, is the multitude of
> ways/places where I can instantiate the AIG command. Do I have to control
> the entire call from beginning to end, or can I just call the Asterisk-Java
> class at certain points with certain parameters?
> >
> >
> >
> > Thanks!
> >
> >
> > Evan
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Guilherme Loch
> Waltrick Góes
> > Sent: Tuesday, January 29, 2008 1:50 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] When can I AIG?
> >
> >
> >
> >
> >
> >
> >
> > Have a look at asterisk-java.org. I has everything you need.
> >
> >
> >
> >
> > On Jan 29, 2008 4:35 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > Hey Guys,
> >
> >
> >
> > I've been doing some research into the AGI-Java connector and was
> wondering if somebody could help me with my architecture.
> >
> >
> >
> > What I'd like to do, is kick off an external java class when a user:
> >
> > 1. Initiates an outgoing call
> >
> > 2. Hangs up the outgoing call
> >
> > 3. Has an incoming call
> >
> > 4. Hangs up the incoming call
> >
> > 5. Misses a call
> >
> > 6. Has a voicemail
> >
> >
> >
> > I'd also like to be able to access all the call details (user number, ext
> number, start time, end time) I see lot of references to the dial plan
> instantiating the AGI in the tutorials, but I was wondering if it is
> possible to use it in these scenarios?
> >
> >
> >
> > How would I need to connect these things up for each user?
> >
> >
> >
> > Sorry if this question is a big basic/elementary, but I'm just getting
> started with Asterisk and would really appreciate some help. Thanks a ton!
> >
> >
> >
> > Evan Ruff
> >
> >
> >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > --
> > Guilherme Loch Góes
> >
> > Visite nossa loja virtual: http://www.shopvoip.com.br
> >
> > Notícias e Fórum sobre VoIP com software livre:
> http://www.asteriskexperts.com.br
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Guilherme Loch Góes
>
> Visite nossa loja virtual: http://www.shopvoip.com.br
>
> Notícias e Fórum sobre VoIP com software livre:
> http://www.asteriskexperts.com.br
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Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite

[Steve Totaro]

On Jan 29, 2008 5:55 PM, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
> Franklin,
>
> Because ChanSpy() is a "passive" monitor, there is nothing about the
> implementation that would cause Asterisk to shunt the speech back to
> itself.  Asterisk only does this in situations where it is out of the
> media path and needs to insinuate itself back into it for the purpose
> of generating media, such as on-hold music, IVR, etc.
>
> What you're wanting should, in my opinion, basically be submitted as a
> feature request.  Perhaps the developers can add a flag to the ChanSpy()
> invocation repertoire to make this work.
>
> Cheers,
>
> -- Alex
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: +1-678-954-0670
> Direct : +1-678-954-0671


Alex, he was not asking why, it is obvious he knows why.

He was asking for a solution or idea on how to work around this issue.

Are you using Sangoma cards?  If so, I might have a very good answer
for you, as well as another very possible different solution.  Both
would be outside of Asterisk so some kind of magic would have to
happen to associate the call being spied on to the channel but that
should not be that difficult if you even need it.

Another solution is to track down the code referenced here
http://bugs.digium.com/view.php?id=9888 and modify chanspy to do a
reinvite back to asterisk before starting the spy.

Anyways, I am sure it can be done.  The question is how much time is
it worth to make it happen.

Maybe we should meet for lunch this week.  I can meet you in cow
country or Philly if you want, your choice.  I have to go to both this
week anyways and would like to catch up with things since Astricon.

Thanks,
Steve Totaro

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[asterisk-users] Queue works with across server agent?

[Johnny Tam]

I am using Queue to handle some incoming calls.  I wonder if the agent is 
across multiple servers, will this work?

Thanks in advance
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[asterisk-users] Queue - ${ANSWEREDTIME}

[Johnny Tam]

How to make ${ANSWEREDTIME} to work with Queue, so when the user hangs up, I 
can calculate how much time the each call lasts?

Thanks
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Re: [asterisk-users] When can I AIG?

[Guilherme Loch Waltrick Góes]

Basically you initiate the call via a Originate method inn your Java App and
bridge it to your AGI. Follow the documentation on the site and you should
be in a good path.
Best Regards,

On Jan 29, 2008 7:01 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:

>  Hey there,
>
>
>
> I've actually looked through the site a bunch and found some great
> information. The thing I'm missing, and I think it's because of my lack of
> experience with Asterisk and setting up a dial plan, is the multitude of
> ways/places where I can instantiate the AIG command. Do I have to control
> the entire call from beginning to end, or can I just call the Asterisk-Java
> class at certain points with certain parameters?
>
>
>
> Thanks!
>
>
> Evan
>
>
>
>
>
>
>
>
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Guilherme Loch
> Waltrick Góes
> *Sent:* Tuesday, January 29, 2008 1:50 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] When can I AIG?
>
>
>
> Have a look at asterisk-java.org. I has everything you need.
>
>
>
> On Jan 29, 2008 4:35 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:
>
> Hey Guys,
>
>
>
> I've been doing some research into the AGI-Java connector and was
> wondering if somebody could help me with my architecture.
>
>
>
> What I'd like to do, is kick off an external java class when a user:
>
> 1. Initiates an outgoing call
>
> 2. Hangs up the outgoing call
>
> 3. Has an incoming call
>
> 4. Hangs up the incoming call
>
> 5. Misses a call
>
> 6. Has a voicemail
>
>
>
> I'd also like to be able to access all the call details (user number, ext
> number, start time, end time) I see lot of references to the dial plan
> instantiating the AGI in the tutorials, but I was wondering if it is
> possible to use it in these scenarios?
>
>
>
> How would I need to connect these things up for each user?
>
>
>
> Sorry if this question is a big basic/elementary, but I'm just getting
> started with Asterisk and would really appreciate some help. Thanks a ton!
>
>
>
> Evan Ruff
>
>
>
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Guilherme Loch Góes
>
> Visite nossa loja virtual: http://www.shopvoip.com.br
>
> Notícias e Fórum sobre VoIP com software livre:
> http://www.asteriskexperts.com.br
>
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>



-- 
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
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Re: [asterisk-users] Source Based Call Routing

[Daniel Cole]

Thank you Greg and Alex for your contribution.

I will use your leads to see what I can get asterisk to do :)


Many Thanks,

Daniel


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Grey Man
Sent: Wednesday, 30 January 2008 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Source Based Call Routing

- Original Message 
> From: Daniel Cole <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Sent: Tuesday, 29 January, 2008 10:31:55 PM
> Subject: [asterisk-users] Source Based Call Routing
>
> Hi List,
>
> I have a scenario that I want to try out (we potential have a client
>
 who would need this), but I am as of yet unable to find much help
> with
>
 it.
>
> What we want to do is have an asterisk box with a large number of
>
 extensions (1000+). This asterisk box will have approximately 3 SIP
> trunks
>
 setup back to providers. What we want to do is to be able to
> define
>
 groups of extensions that use specific outbound trunks.
>
> Approximately a third of the extensions will one the first trunk, a
>
 third the second trunk, and the rest will use the last trunk. We also
> need
>
 control over assigning with trunks the given extensions will use.
>
> Any suggestions on how to get this to work would be very much
>
 appreciated.
>

Hi Daniel,

3 different contexts in your dial plan would work. Assign each block of 
accounts (rather than extensions) to the context with the routes that they 
should use. To change an account from using one trunk to another it would be as 
simple as changing its context.

Regards,

Greyman.





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Re: [asterisk-users] Source Based Call Routing

[Alex Balashov]

I would still say the easiest thing by far is to introduce a mediator
in the dial plan that is far more intelligent and extensible than the
dial plan logic itself.  Enter FastAGI.  Then you can just do it ...
however you want.

On Wed, 30 Jan 2008, Paul Hales wrote:

>
> You can also look at routing based on number ranges (if you keep the
> separate numbers in separate number ranges) but I would guess that this
> is not going to suit your needs.
>
> Maybe storing all the accounts in mysql (realtime) would also be a good
> planh.
>
> PaulH
>
>
> On Wed, 2008-01-30 at 09:31 +1100, Daniel Cole wrote:
>> Hi List,
>>
>> I have a scenario that I want to try out (we potential have a client who 
>> would need this), but I am as of yet unable to find much help with it.
>>
>> What we want to do is have an asterisk box with a large number of extensions 
>> (1000+). This asterisk box will have approximately 3 SIP trunks setup back 
>> to providers. What we want to do is to be able to define groups of 
>> extensions that use specific outbound trunks.
>>
>> Approximately a third of the extensions will one the first trunk, a third 
>> the second trunk, and the rest will use the last trunk. We also need control 
>> over assigning with trunks the given extensions will use.
>>
>> Any suggestions on how to get this to work would be very much appreciated.
>>
>>
>> Many Thanks,
>>
>> Daniel
>>
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Source Based Call Routing

[Paul Hales]

You can also look at routing based on number ranges (if you keep the
separate numbers in separate number ranges) but I would guess that this
is not going to suit your needs.

Maybe storing all the accounts in mysql (realtime) would also be a good
planh.

PaulH


On Wed, 2008-01-30 at 09:31 +1100, Daniel Cole wrote:
> Hi List,
> 
> I have a scenario that I want to try out (we potential have a client who 
> would need this), but I am as of yet unable to find much help with it.
> 
> What we want to do is have an asterisk box with a large number of extensions 
> (1000+). This asterisk box will have approximately 3 SIP trunks setup back to 
> providers. What we want to do is to be able to define groups of extensions 
> that use specific outbound trunks.
> 
> Approximately a third of the extensions will one the first trunk, a third the 
> second trunk, and the rest will use the last trunk. We also need control over 
> assigning with trunks the given extensions will use.
> 
> Any suggestions on how to get this to work would be very much appreciated.
> 
> 
> Many Thanks,
> 
> Daniel
> 
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Re: [asterisk-users] chanspy does not pull the call back to asterisk after a reinvite

[Alex Balashov]

Franklin,

Because ChanSpy() is a "passive" monitor, there is nothing about the
implementation that would cause Asterisk to shunt the speech back to
itself.  Asterisk only does this in situations where it is out of the
media path and needs to insinuate itself back into it for the purpose
of generating media, such as on-hold music, IVR, etc.

What you're wanting should, in my opinion, basically be submitted as a
feature request.  Perhaps the developers can add a flag to the ChanSpy()
invocation repertoire to make this work.

Cheers,

-- Alex

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] chanspy does not pull the call back to asterisk after a reinvite

[Franklin Webb]

Hello all,

I am allowing a reinvite between a snom 320 phone and a SIP gateway to take 
load off my Asterisk server.  When I put the caller on hold, for example, 
Asterisk successfully reinserts itself into the rtp stream to play music on 
hold to the caller, but when I do a chanspy Asterisk does not seem to pull the 
call back.  If I am spying on a channel when the call build up happens the 
reinvite never occurs and it works, but I cannot jump in and spy on a call in 
progress once the reinvite has happened.

Has anyone run into this issue any maybe have a solution, or does anyone know 
of a good way to get that call back onto the Asterisk switch from another 
extension prior to calling chanspy?

Thanks much,

Franklin Webb

-- 
Franklin Webb
Asst Project Manager
Inter Medi@ Marketing Solutions
610-701-9670
[EMAIL PROTECTED]


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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[C F]

On Jan 29, 2008 3:54 PM, Grey Man <[EMAIL PROTECTED]> wrote:
> - Original Message 
> > From: C F <[EMAIL PROTECTED]>
> > To: Asterisk Users Mailing List - Non-Commercial Discussion 
> > 
> > Sent: Tuesday, 29 January, 2008 8:05:00 PM
> > Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> >
> > Grey,
> > Just tested with 1.2.13
> > Asterisk always (blind or attd xfer) creates 2 records.
> > A few points, NEVER rely on source as the billable number.
> > Always use account codes.
> > Match the lastdata field against dst fields to figure out that it was
> > an xfer when doing the rating. The lastdata field will have the right
> > number.
> >
>
>
> That will work for blind transfers but not attended and even in the blind 
> transfer case the CDR's still aren't correct you're relying on an 
> informational field.
>

I tested it. It does so for both attended and blind. However, I only
tested it with SIP xfers and not with Tt.

>
>
>
>   Make the switch to the world's best email. Get the new Yahoo!7 Mail 
> now. www.yahoo7.com.au/worldsbestemail
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>
>
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Re: [asterisk-users] Source Based Call Routing

[Grey Man]

- Original Message 
> From: Daniel Cole <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Tuesday, 29 January, 2008 10:31:55 PM
> Subject: [asterisk-users] Source Based Call Routing
> 
> Hi List,
> 
> I have a scenario that I want to try out (we potential have a
> client
> 
 who would need this), but I am as of yet unable to find much help
> with
> 
 it.
> 
> What we want to do is have an asterisk box with a large number
> of
> 
 extensions (1000+). This asterisk box will have approximately 3 SIP
> trunks
> 
 setup back to providers. What we want to do is to be able to
> define
> 
 groups of extensions that use specific outbound trunks.
> 
> Approximately a third of the extensions will one the first trunk,
> a
> 
 third the second trunk, and the rest will use the last trunk. We also
> need
> 
 control over assigning with trunks the given extensions will use.
> 
> Any suggestions on how to get this to work would be very
> much
> 
 appreciated.
> 

Hi Daniel,

3 different contexts in your dial plan would work. Assign each block of 
accounts (rather than extensions) to the context with the routes that they 
should use. To change an account from using one trunk to another it would be as 
simple as changing its context.

Regards,

Greyman.
 




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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

> - Original Message 

> From: Matt <[EMAIL PROTECTED]>

> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 

> Sent: Tuesday, 29 January, 2008 9:24:14 PM

> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's


>

> The provider can disable transfers (which is what we do), but why can a PBX 
> not still allow it?  Our PBX customers all can do 
> transferring... but that's because billing isn't needed THERE.  The billing, 
> if any, is done on our end, or their providers end.   
> This really seems like a very small and moot point that is being blown up.

 

Depends how much it could cost you I guess :). If you're not supporting 
transfers it's a moot point if you are it's a bit more interesting.

> If the receptionist needs to transfer the call, then she should be able to do 
> that within the confines of her PBX... the transfer of
> her call should NEVER go back out her PBX back to the supplier, for if it 
> does, her PBX now loses control of that call.

 

Our customer base is residential and small business. They don't want to either 
pay for or support another a PBX thats what they've come to us for in the first 
place a lot of the time.

Regards,

Greyman.








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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

- Original Message 
> From: Kevin P. Fleming <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Tuesday, 29 January, 2008 9:34:23 PM
> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> 
> Grey Man wrote:
> 
> > That will work for blind transfers but not attended and even in
> the
> 
 blind transfer case the CDR's still aren't correct you're relying on
> an
> 
 informational field.
> 
> I think there is an important point being missed here; Asterisk did not
> originate the concept of CDRs, nor did it specify what they contain or
> how they are to be collected and generated.
> 
> CDRs have existed for decades before Asterisk was created, and they are
> a fairly well understood concept throughout the telephony switching
> industry. They were designed for billing, and in many
> telephony
> 
 networks
> are still used for billing.

Hi Kevin,

Thanks for responding. I'd actually prefer to use some form of real-time call 
control for billing within Asterisk but that's another story.

For the half a dozen or so integrations we have done with PSTN carriers the 
CDRs are integral to the whole process. Arguably the biggest step in the whole 
interconnect process is matching up the CDRs for agreement.

> However, CDRs were created before the users of those services had the
> ability to transfer calls, make three-way calls, make conference calls,
> and do other magical things. As such, there is no way in a CDR to
> represent this activity in any *complete* manner. 

I understand there are likely to always remain certain things that CDR's cannot 
cope with but I don't think transfers fall into that category. Would there be 
anything wrong with recording a CDR for each end of a bridge instead of one CDR 
per bridge? If one end of the bridge changes, as in the case of a transfer, you 
get one CDR. When the bridge hangs up you get two CDR's which in fact does make 
sense as a bridge is two calls/channels.

I'd be more than happy to produce call flows for: transfers, 3 way call, 
whatever else; with the exact CDRs if that would help to clarify things.

> Doing so will require a redesign of the CDR system, which Steve Murphy has 
> already begun for
> Asterisk 1.6.

Yes and thanks must go to Steve for delving into this very unglamorous area 
it's certainly not up there with video conferencing. The worrying thing though 
is the CDR's for attended transfers in 1.4 are now worse than they were in 1.2. 
I've read through Steve's blog posting on the new design and I think there are 
still some problems with the CDR scenarios. Using overlapping CDRs to determine 
if a transfer was in progress is fragile (what happens if simultaneous calls 
are supported) and apart from that the new CDRs will still don't provide enough 
information to bill all the call legs involved in a transfer.

> As far as I am aware, everyone who builds a complete billing system for
> Asterisk and expects it to be accurate and reliable uses other means in
> addition to CDRs for collecting the information, or they restrict their
> users to not performing actions that will break the billing process.


That's fair enough I guess but there are quite a few people using Asterisk that 
have been relying exclusively on its CDRs that weren't aware of the 
inaccuracies. Certainly the 3 providers I found in the last two days weren't 
(I've emailed them now). 

I don't think it would be insurmountable to improve the CDR design in Asterisk. 
Maybe it won't get to a stage where it's perfect but if a new design was 
produced it would pave the way for those of us that this is a big deal for to 
assist in the implementation.

Regards,

Greyman.



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Re: [asterisk-users] Source Based Call Routing

[Alex Balashov]

I would broker the dial-out requests through FastAGI and put the logic
that examines extensions and implements the load balancing / distribution
in there.

On Wed, 30 Jan 2008, Daniel Cole wrote:

> Hi List,
>
> I have a scenario that I want to try out (we potential have a client who 
> would need this), but I am as of yet unable to find much help with it.
>
> What we want to do is have an asterisk box with a large number of extensions 
> (1000+). This asterisk box will have approximately 3 SIP trunks setup back to 
> providers. What we want to do is to be able to define groups of extensions 
> that use specific outbound trunks.
>
> Approximately a third of the extensions will one the first trunk, a third the 
> second trunk, and the rest will use the last trunk. We also need control over 
> assigning with trunks the given extensions will use.
>
> Any suggestions on how to get this to work would be very much appreciated.
>
>
> Many Thanks,
>
> Daniel
>
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] Source Based Call Routing

[Daniel Cole]

Hi List,

I have a scenario that I want to try out (we potential have a client who would 
need this), but I am as of yet unable to find much help with it.

What we want to do is have an asterisk box with a large number of extensions 
(1000+). This asterisk box will have approximately 3 SIP trunks setup back to 
providers. What we want to do is to be able to define groups of extensions that 
use specific outbound trunks.

Approximately a third of the extensions will one the first trunk, a third the 
second trunk, and the rest will use the last trunk. We also need control over 
assigning with trunks the given extensions will use.

Any suggestions on how to get this to work would be very much appreciated.


Many Thanks,

Daniel

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Re: [asterisk-users] softmodems bank for ast.

[Andrew Joakimsen]

I think its totally possible. Most of these "winmodems" are just sound
cards with a hybrid.

On Jan 29, 2008 11:51 AM, Pepe Aracil <[EMAIL PROTECTED]> wrote:
> Hi.
>
> We need a full featured modem bank 20+ to attend data calls.
>
> IAXmodem only supports fax protocols because spandsp only support fax 
> protocols.
>
> The idea is to do a IAX wrapper like IAXmodem but with a full featured
> (but propietary) softmodem library like PCTEL or linuxant.
>
>
> I hate astribank+hardmodems solution.
>
> Some body think that is possible?
>
> Sorry for my horrible English. ;)
>
> Bye.
>
>
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Re: [asterisk-users] Do Asterisk requires audio codec to be installed?

[Andrew Joakimsen]

No, there is no need for any audio codec to be installed. In that case
it would just be the words worst B2B SIP UA.

By default Asterisk comes with quite a few codecs.

On Jan 29, 2008 5:03 AM,  <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Hi,
>
>  Can you please tell me whether Asterisk requires any audio or video codec
> to be installed separately or it supports itself?
>
>
>  Thanking you,
>
>  Preeta
>
> P Please do not print this email unless it is absolutely necessary. Spread
> environmental awareness.
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> WARNING: Computer viruses can be transmitted via email. The recipient should
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Re: [asterisk-users] Queue member add

[Mark Michelson]

Rob Schall wrote:
> Hopefully a fairly easy question for the group...
> 
> I have a queue which should contain about 10 agents (it will be all the
> phones in the office). This office is remote, so I would like to add
> their sip phones into the queue remotely. Also, if the system ever gets
> reloaded or rebooted, I need those agents to remain in the queue.
> 
> Question:
> 1) How do you remotely add agents to their respective queues and attach
> that to a specific sip phone?

You can add queue members remotely from the manager by using the QueueAdd 
manager action. If you can open an Asterisk CLI remotely, then you can use the 
"queue add member" command ("add queue member" if you're using 1.2) from there 
as well.

> 2) How do you keep those phones in that queue even after the system reboots?

set persistentmembers=yes in queues.conf

> Rob

Mark Michelson


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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Kevin P. Fleming]

Grey Man wrote:

> That will work for blind transfers but not attended and even in the blind 
> transfer case the CDR's still aren't correct you're relying on an 
> informational field.

I think there is an important point being missed here; Asterisk did not
originate the concept of CDRs, nor did it specify what they contain or
how they are to be collected and generated.

CDRs have existed for decades before Asterisk was created, and they are
a fairly well understood concept throughout the telephony switching
industry. They were designed for billing, and in many telephony networks
are still used for billing.

However, CDRs were created before the users of those services had the
ability to transfer calls, make three-way calls, make conference calls,
and do other magical things. As such, there is no way in a CDR to
represent this activity in any *complete* manner. Doing so will require
a redesign of the CDR system, which Steve Murphy has already begun for
Asterisk 1.6.

As far as I am aware, everyone who builds a complete billing system for
Asterisk and expects it to be accurate and reliable uses other means in
addition to CDRs for collecting the information, or they restrict their
users to not performing actions that will break the billing process.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Benny Amorsen]

Matt <[EMAIL PROTECTED]> writes:

> Asterisk is doing exactly as it should.. when it steps out of the media
> path, the CDR is also dropped, as asterisk is no longer responsible for
> that call.

Even if Asterisk stays in the media path, the CDR's are dropped.

It is an annoying problem. Hopefully the new CDR system provides a way
to avoid it. It doesn't affect us so much because customers don't get
to send transfer requests to our billing PBX's, but it's still silly.


/Benny



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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Matt]

>
>
> Disbaling transfers is an attractive option from my point of view but not
> from my customer's. Being able to transfer an incoming call from the
> receptionist to the required person is something businesses will consider
> changing provider for in my experience.
>

The provider can disable transfers (which is what we do), but why can a PBX
not still allow it?  Our PBX customers all can do transferring... but that's
because billing isn't needed THERE.  The billing, if any, is done on our
end, or their providers end.This really seems like a very small and moot
point that is being blown up.

If the receptionist needs to transfer the call, then she should be able to
do that within the confines of her PBX... the transfer of her call should
NEVER go back out her PBX back to the supplier, for if it does, her PBX now
loses control of that call.
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[asterisk-users] ShoreTel <-> Asterisk Integration

[Joe Evans]

Does anyone have experience using ShoreTel SIP trunks to integrate an 
Asterisk system?

I am having trouble when the ShoreTel system transfers an incoming call 
from a SIP trunk to the voicemail system. From the SIP traffic, it looks 
like it negotiates a codec correctly, but once the RTP stream starts the 
call drops or there is no audio. I see errors in Asterisk such as:

chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 
[EMAIL PROTECTED] for seqno 104 (Critical Request)

Has anyone run into this before or have any ideas?

Thanks,
Joe

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Re: [asterisk-users] External Incomming Call Directed PickUP

[Fernando Berretta]

Someone could please help me ?

Regards,
Fernando

Fernando Berretta wrote:

Dear Lacy,

We are using Standard FreePbx installation and we are trying to direct 
pickup all the calls with **EXT NUMBER.


[app-pickup]
include => app-pickup-custom
exten => _**.,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten => _**.,n,Pickup(${EXTEN:2})

This is the FreePbx configuration for call pickup but doesn't work for 
calls which comes from users which are in other context neither 
incoming calls from-trunk


Any help will be appreciated

Regards,
Fernando

Lacy Moore wrote:

My magic orb is on the fritz.  Can you give some more info?  What
extension is ringing?  What are you dialing to pick up?  What does
your conf files look like?

I think I might know what the problem is, but I need a little more
info.  Read "core show application Pickup" carefully, and then re-read
it 3 or 4 more times.  It seems odd at first, but then you catch on.
You are picking up the calling channel, not the called extension.

On Jan 25, 2008 5:28 PM, Fernando Berretta <[EMAIL PROTECTED]> wrote:
  

Hi,

I'm having problems with Directed PickUn and Asterisk 1.4.

Directed call pickup **EXT works ok with internal calls which are in the
same CONTEXT but,, with calls in which are from other context  or
incoming calls from IVR this function doesn't work as is pointed in
http://bugs.digium.com/view.php?id=11639

I'm using FreePbx 2.3,, and dont know how to solve or workaround this
problem

Could some one please help me.

Best Regards,
Fernando

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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

- Original Message 

> From: Matt <[EMAIL PROTECTED]>

> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 

> Sent: Tuesday, 29 January, 2008 8:39:25 PM

> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's


> 
> Grey... I'm not debating that this is how it works.   We provide wholesale 
> VoIP and retail VoIP.   Transfers are disabled on both
> of those.  That was one of the first things we did... all media and calls 
> stay in our system.  If the company doesn't have
> transfers disabled, that is their own fault, and their loss.  I know exactly 
> what you are referring to, and technically I'd say

Disbaling transfers is an attractive option from my point of view but not from 
my customer's. Being able to transfer an incoming call from the receptionist to 
the required person is something businesses will consider changing provider for 
in my experience.

There is no way Asterisk is correct with regards the CDR's produced by a 
transfer. That's what I'm hoping to get people to agree on and think about a 
change for.

> Asterisk is still correct, because the leg of the call that billing was 
> happening on (the sip client) is no longer there.

 

Correct? There are still two other calls that were initiated by the user why 
should they be dismissed because the first call hungup? In any case they are 
not dismissed entirely they are just combined and recorded inaccurately.

Regards,

Greyman








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Re: [asterisk-users] When can I AIG?

[Evan Ruff]

Hey there,

I've actually looked through the site a bunch and found some great information. 
The thing I'm missing, and I think it's because of my lack of experience with 
Asterisk and setting up a dial plan, is the multitude of ways/places where I 
can instantiate the AIG command. Do I have to control the entire call from 
beginning to end, or can I just call the Asterisk-Java class at certain points 
with certain parameters?

Thanks!

Evan




From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guilherme Loch 
Waltrick Góes
Sent: Tuesday, January 29, 2008 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] When can I AIG?

Have a look at asterisk-java.org. I has everything 
you need.

On Jan 29, 2008 4:35 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:

Hey Guys,



I've been doing some research into the AGI-Java connector and was wondering if 
somebody could help me with my architecture.



What I'd like to do, is kick off an external java class when a user:

1. Initiates an outgoing call

2. Hangs up the outgoing call

3. Has an incoming call

4. Hangs up the incoming call

5. Misses a call

6. Has a voicemail



I'd also like to be able to access all the call details (user number, ext 
number, start time, end time) I see lot of references to the dial plan 
instantiating the AGI in the tutorials, but I was wondering if it is possible 
to use it in these scenarios?



How would I need to connect these things up for each user?



Sorry if this question is a big basic/elementary, but I'm just getting started 
with Asterisk and would really appreciate some help. Thanks a ton!



Evan Ruff



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--
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre: 
http://www.asteriskexperts.com.br
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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

- Original Message 
> From: C F <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Tuesday, 29 January, 2008 8:05:00 PM
> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> 
> Grey,
> Just tested with 1.2.13
> Asterisk always (blind or attd xfer) creates 2 records.
> A few points, NEVER rely on source as the billable number.
> Always use account codes.
> Match the lastdata field against dst fields to figure out that it was
> an xfer when doing the rating. The lastdata field will have the right
> number.
> 


That will work for blind transfers but not attended and even in the blind 
transfer case the CDR's still aren't correct you're relying on an informational 
field.



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Re: [asterisk-users] When does Asterisk "REFER"?

[Grey Man]

- Original Message 
> From: Philip Prindeville <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Tuesday, 29 January, 2008 7:11:01 PM
> Subject: [asterisk-users] When does Asterisk "REFER"?
> 
> I was wondering under what conditions Asterisk will hand off a call to 
> another switch.
> 
> I'm trying to verify that my local PSTN's Coppercom switch operates 
> correctly...  and wanted to know how to get a call REFER'd to another 
> end-point.
> 
> Thanks,
> 
> -Philip
> 

 
Hi Philip,

You can use the transfer command in your dial plan apart from that I don't know 
of any case when Asterisk would initiate a REFER. Generally Asterisk is the one 
handling the REFER's in reponse to user's transfer requests.

Regards,

Aaron



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Re: [asterisk-users] When does Asterisk "REFER"?

[Kevin P. Fleming]

Philip Prindeville wrote:

> I'm trying to verify that my local PSTN's Coppercom switch operates 
> correctly...  and wanted to know how to get a call REFER'd to another 
> end-point.

I don't think Asterisk will ever generate a REFER, but the only possible
way it could would be using the Transfer() application in the dialplan.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Matt]

>
>
>
> Hi Matt,
>
> Sadly I understand all to well how transfers work. I've had to go over and
> over this for the last 12 months trying to find different ways of handling
> it. I'm talking about blind and attended call transfers here not IAX or any
> other kind. We are not taking Asterisk out of the media path and even if we
> were you wouldn't want to be losing CDR's from a provider's point of view,
> whoever set the call up is still paying for it regardless of where the media
> has been re-invited to.
>
> Out of the 8 Asterisk based providers I have tested 3 have this issue and
> the other 5 don't support transfers.
>
> It's dead simple for anyone to test. Find an Asterisk provider that
> supports transfers, connect with the xten, do a blind or attended transfer
> and check the CDR's. Call a free or cheap destination as the first leg of
> your transfer and the expensive destination second. You'll be pleasantly
> suprised at the bill!
>
> Regards,
>
> Greyman.
>
>
>
Grey... I'm not debating that this is how it works.   We provide wholesale
VoIP and retail VoIP.   Transfers are disabled on both of those.  That was
one of the first things we did... all media and calls stay in our system.
If the company doesn't have transfers disabled, that is their own fault, and
their loss.  I know exactly what you are referring to, and technically I'd
say Asterisk is still correct, because the leg of the call that billing was
happening on (the sip client) is no longer there.
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Re: [asterisk-users] Queue member add

[Alex Balashov]

Queue members can be SIP channel names, including ones that are reachable 
at remote destination URIs, if that is what you re asking.  e.g.

member => SIP/[EMAIL PROTECTED]

Queue members are made persistent in AstDB with the 'persistentmembers = 
yes' option and survive reboots.

On Tue, 29 Jan 2008, Rob Schall wrote:

> Hopefully a fairly easy question for the group...
>
> I have a queue which should contain about 10 agents (it will be all the
> phones in the office). This office is remote, so I would like to add
> their sip phones into the queue remotely. Also, if the system ever gets
> reloaded or rebooted, I need those agents to remain in the queue.
>
> Question:
> 1) How do you remotely add agents to their respective queues and attach
> that to a specific sip phone?
> 2) How do you keep those phones in that queue even after the system reboots?
>
> Rob
>
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

> - Original Message 

> From: Matt <[EMAIL PROTECTED]>

> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 

> Sent: Tuesday, 29 January, 2008 7:28:32 PM

> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's


>
> Grey,

>I don't think you understand how transfers work.   Let's take for example:


>
>USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B).


>
>1 Dials A and transfers the call to B.

>The call data is now NO LONGER in the asterisk path, therefore asterisk has 
>nothing to do with the CDR.  However, the call >legs are still going out of 
>the providers trunking.   This is not a problem with asterisk, but a logic 
>problem with you/providers >dial-plan.  

 
>
>Asterisk is doing exactly as it should.. when it steps out of the media path, 
>the CDR is also dropped, as asterisk is no longer >responsible for that call.



Hi Matt,

Sadly I understand all to well how transfers work. I've had to go over and over 
this for the last 12 months trying to find different ways of handling it. I'm 
talking about blind and attended call transfers here not IAX or any other kind. 
We are not taking Asterisk out of the media path and even if we were you 
wouldn't want to be losing CDR's from a provider's point of view, whoever set 
the call up is still paying for it regardless of where the media has been 
re-invited to.

Out of the 8 Asterisk based providers I have tested 3 have this issue and the 
other 5 don't support transfers.

It's dead simple for anyone to test. Find an Asterisk provider that supports 
transfers, connect with the xten, do a blind or attended transfer and check the 
CDR's. Call a free or cheap destination as the first leg of your transfer and 
the expensive destination second. You'll be pleasantly suprised at the bill!

Regards,

Greyman.










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[asterisk-users] Queue member add

[Rob Schall]

Hopefully a fairly easy question for the group...

I have a queue which should contain about 10 agents (it will be all the
phones in the office). This office is remote, so I would like to add
their sip phones into the queue remotely. Also, if the system ever gets
reloaded or rebooted, I need those agents to remain in the queue.

Question:
1) How do you remotely add agents to their respective queues and attach
that to a specific sip phone?
2) How do you keep those phones in that queue even after the system reboots?

Rob

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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[C F]

Grey,
Just tested with 1.2.13
Asterisk always (blind or attd xfer) creates 2 records.
A few points, NEVER rely on source as the billable number.
Always use account codes.
Match the lastdata field against dst fields to figure out that it was
an xfer when doing the rating. The lastdata field will have the right
number.


On Jan 29, 2008 2:28 PM, Matt <[EMAIL PROTECTED]> wrote:
> Grey,
> I don't think you understand how transfers work.   Let's take for example:
>
> USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B).
>
> 1 Dials A and transfers the call to B.
> The call data is now NO LONGER in the asterisk path, therefore asterisk has
> nothing to do with the CDR.  However, the call legs are still going out of
> the providers trunking.   This is not a problem with asterisk, but a logic
> problem with you/providers dial-plan.
>
> Asterisk is doing exactly as it should.. when it steps out of the media
> path, the CDR is also dropped, as asterisk is no longer responsible for that
> call.
>
>
>
> On Jan 29, 2008 12:48 PM, Grey Man <[EMAIL PROTECTED]> wrote:
>
> >
> >
> >
> > - Original Message 
> > > From: Richard Revels <[EMAIL PROTECTED]>
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> > > Sent: Tuesday, 29 January, 2008 12:21:16 PM
> > > Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> > >
> > > It's not Asterisk, it's SIP.  Transfer takes the signaling off the
> > > Asterisk box.
> > >
> > > In features.conf, replace blind transfer with a call to a macro.
> > > Then
> > >
> >
> > > redo your dialplan with the 'g' option on inward dial commands. When
> > > the called party uses the transfer command, your macro should read
> > > the
> > >
> >
> > > digits to call and then store them in the db, a unique global, or
> > > GROUP
> > >
> >
> > > () variable. Then it should hang up. This will cause the calling leg
> > > to exit the dial command to the next priority which should be a check
> > > of the variable. If digits are present, use the dial command to call
> > > them at your provider. No fuss, no muss.
> > >
> > > You should make sure the peer entry for the outbound side includes
> > > canreinvite=yes so only the signaling remains on your box and the
> > > media is invited off.
> > >
> > > You should also ignore calls to your macro that hit from the inbound
> > > call leg. Just return immediately and neither side will ever know the
> > > inbound call leg left for a moment.
> > >
> > > Sent from my iPhone
> >
> >
> > Hi Richard,
> >
> > I'm not actually sure we're talking about the same thing here. It's not
> transfers I have a problem with it's the CDRs the transferred calls end up
> generating. In this case I am the provider and transfers through our
> Asterisk servers work fine it's just that we can't properly bill for them.
> >
> >
> > Regards,
> >
> > Greyman.
> >
> >
> >
> >  Make the switch to the world's best email. Get the new Yahoo!7 Mail
> now. www.yahoo7.com.au/worldsbestemail
> >
> >
> >
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> >
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> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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Re: [asterisk-users] speex, ilbc and g729 codecs

[SIP]

bilal ghayyad wrote:
> Hi List;
>
> Anyone tried to use speex, ilbc and g729 and come back
> with a preferred one in the quality?
>
> Regards
> Bilal
>
>
>   
> 
> Never miss a thing.  Make Yahoo your home page. 
> http://www.yahoo.com/r/hs
>
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>   
Depends on what you're using it for.  We've had excellent luck with iLBC 
for voice quality and g729 as well. Speex is better than GSM, but not 
spectacularly.

The problem with all of them of course, is transcoding. They have less 
of a bandwidth footprint for acceptable quality, but more of a processor 
footprint.

N.

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Re: [asterisk-users] Change Default Voicemail Message

[Matt]

The issue is simple.  You make the voicemail box be the same as the room
number, then you get:

Playback(Welcome-2-nursing-home)
Voicemail(xxx,s)

Voicemail plays "you've reached room, xxx, please leave a message" then
beeps and records message.

On Jan 7, 2008 5:29 PM, Daniel Cole <[EMAIL PROTECTED]> wrote:

>  Thank you for your reply Trevor.
>
> Is there an easy way to achieve this with a computer generated voice? We
> do not wish to manually record the messages if possible, in the interests of
> a consistent message across all voicemail boxes. What would be the easiest
> way to do this?
>
> Also, can you please give me some pointers on how to get the voicemail to
> play the separate message before the normal voicemail message? I'm guessing
> it would be done with a custom voicemail content, but im not sure how to
> write it correctly.
>
> Many Thanks,
>
> Daniel Cole
>
>  --
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Trevor G. Hammonds
> *Sent:* Monday, 7 January 2008 3:52 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* Re: [asterisk-users] Change Default Voicemail Message
>
>  Daniel,
>
> You could have Alison record a prompt "Welcome to (nursing home)" and
> re-record the prompt "The person at extension..." to be "The person in
> room...".  Then have your dialplan play the "Welcome To..." message before
> sending the call to voice mail.  Then callers will hear "Welcome to (Nursing
> Home).  The person in room 5 is unavailable.  Please leave your message..."
> and if the resident has a recorded personal greeting or name, it would
> replace the "The person..." portion with either the resident's recorded name
> or greeting.
>
>
>
> Sincerely,
>
> Trevor Hammonds
>
>
>
>
>
> *From:* Daniel Cole
> *Sent:* Sunday, January 06, 2008 6:15 PM
>
>
>
> Hello List,
>
>
>
> I have a client (a nursing home)  that we are looking at installing a
> trixbox for. One of the features that they would really like is a
> customized, standard voicemail recording for each of the residents rooms.
>
>
>
> We are looking for something along the lines of a voicemail recording like
> this:  "Welcome to (nursing home). You have reached room 5. Please leave a
> message after the tone".
>
>
>
> What would be the easiest way to get this to work. I have had a look at a
> few options, but I cant seem to find what I am after.
>
>
>
> Any help would be much appreciated.
>
>
>
>
>
> Thank You,
>
>
>
> Daniel Cole
>
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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Matt]

Grey,
I don't think you understand how transfers work.   Let's take for example:

USER-1 dials LOCATION A and then LOCATION B (referred to as 1,A,B).

1 Dials A and transfers the call to B.
The call data is now NO LONGER in the asterisk path, therefore asterisk has
nothing to do with the CDR.  However, the call legs are still going out of
the providers trunking.   This is not a problem with asterisk, but a logic
problem with you/providers dial-plan.

Asterisk is doing exactly as it should.. when it steps out of the media
path, the CDR is also dropped, as asterisk is no longer responsible for that
call.

On Jan 29, 2008 12:48 PM, Grey Man <[EMAIL PROTECTED]> wrote:

> - Original Message 
> > From: Richard Revels <[EMAIL PROTECTED]>
> > To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> > Sent: Tuesday, 29 January, 2008 12:21:16 PM
> > Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> >
> > It's not Asterisk, it's SIP.  Transfer takes the signaling off the
> > Asterisk box.
> >
> > In features.conf, replace blind transfer with a call to a macro.
> > Then
> >
>
> > redo your dialplan with the 'g' option on inward dial commands. When
> > the called party uses the transfer command, your macro should read
> > the
> >
>
> > digits to call and then store them in the db, a unique global, or
> > GROUP
> >
>
> > () variable. Then it should hang up. This will cause the calling leg
> > to exit the dial command to the next priority which should be a check
> > of the variable. If digits are present, use the dial command to call
> > them at your provider. No fuss, no muss.
> >
> > You should make sure the peer entry for the outbound side includes
> > canreinvite=yes so only the signaling remains on your box and the
> > media is invited off.
> >
> > You should also ignore calls to your macro that hit from the inbound
> > call leg. Just return immediately and neither side will ever know the
> > inbound call leg left for a moment.
> >
> > Sent from my iPhone
>
>
> Hi Richard,
>
> I'm not actually sure we're talking about the same thing here. It's not
> transfers I have a problem with it's the CDRs the transferred calls end up
> generating. In this case I am the provider and transfers through our
> Asterisk servers work fine it's just that we can't properly bill for them.
>
> Regards,
>
> Greyman.
>
>
>
>  Make the switch to the world's best email. Get the new Yahoo!7 Mail
> now. www.yahoo7.com.au/worldsbestemail
>
>
>
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[asterisk-users] speex, ilbc and g729 codecs

[bilal ghayyad]

Hi List;

Anyone tried to use speex, ilbc and g729 and come back
with a preferred one in the quality?

Regards
Bilal


  

Never miss a thing.  Make Yahoo your home page. 
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[asterisk-users] When does Asterisk "REFER"?

[Philip Prindeville]

I was wondering under what conditions Asterisk will hand off a call to 
another switch.

I'm trying to verify that my local PSTN's Coppercom switch operates 
correctly...  and wanted to know how to get a call REFER'd to another 
end-point.

Thanks,

-Philip


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Re: [asterisk-users] Quad core is not a good idea! (was: Asterisk on Dell PowerEdge 2950)

[broadband Voice]

Thanks for the response. I have not bought it yet and here are the specs I
am considering. Any comments before I make the purchase.

   *PowerEdge 2950 III*
Date   1/29/2008 12:40:22 PM Central Standard Time   Catalog
Number   4 Retail 04Catalog Number / Description   Product Code   SKU
Id *PowerEdge 2950 III*:
Dual Core Intel(R) Xeon(R) 5160, 4MB Cache, 3.0GHz, 1333MHz FSB   29530W
[223-4926]   1*Additional Processor*:
Dual Core Intel(R) Xeon(R) 5160, 4MB Cache, 3.00GHz, 1333MHZ FSB   2PW30
[311-6222]   2*Memory*:
4GB 667MHz (4x1GB), Dual Ranked DIMMs   4G4D6D   [311-6154]   3*Operating
System*:
No Operating System   NOOS   [420-6320]   11*Backplane*:
1x6 Backplane for 3.5-inch Hard Drives   1X6353   [311-7936]   18*Primary
Controller*:
SAS 6/iR Integrated, x6 Backplane   S6IX6   [341-5941]   9*Hard Drive
Configuration*:
Integrated SAS/SATA, SAS 6/iR Integrated, No RAID   6SS   [341-5720]   27
*Primary Hard Drive*:
250GB 7.2K RPM Serial ATA 3Gbps 3.5-in HotPlug Hard Drive   250S2
[341-3037]   8*2nd Hard Drive*:
250GB 7.2K RPM Serial ATA 3Gbps 3.5-in HotPlug Hard Drive   250S2
[341-3037]   23*Chassis Configuration*:
No Rack Rails Included   NORAIL   [310-7411]   28*Riser Card*:
Riser with 3 PCIe Slots   PCIE   [320-4607]   7*Power Supply*:
Non-Redundant Power Supply   NRPS3   [310-9895]   36*Bezel*:
Rack Bezel   BEZEL   [313-3920]   17*Network Adapter*:
Dual Embedded Broadcom(R) NetXtreme II 5708 Gigabit Ethernet NIC   OBNIC
[430-1764]   13*TCP/IP Offload Engine Enablement*:
Broadcom TCP/IP Offload Engine Not Enabled   NTOEKEY   [430-1765]   6*
Documentation*:
Electronic Documentation and OpenManage CD Kit   EDOCS   [310-7415]   21
*CD/DVD Drive*:
24X IDE CD-ROM   24XCD   [313-3932]   16*Floppy Drive*:
No Floppy Drive for x6 Backplane   NFDX6   [341-3685]   10*Mouse*:
Mechanical Two-Button Mouse, USB   USBMW   [310-8171]   12*Server
Accessories*:
USB to PS2 Adapter for KVM Connectivity   USBPS2C   [310-6690]   57
*Hardware
Support Services*:
3Yr BASIC SUPPORT: 5x10 HW-Only, 5x10 NBD Onsite   U3OS
[960-8162][960-8192][970-4070][984-1399][984-1417]   29*Installation
Services*:
No Installation Assessment   NOINSTL   [900-9997]   32

Print 


On 1/29/08, Massimo Nuvoli <[EMAIL PROTECTED]> wrote:
>
> broadband Voice ha scritto:
> > Does anyone have any compatibilty issues with Dell *PowerEdge^TM   2950
> III
> > 2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards.
> > Thanks.
>
> Consider 2 socket dual core CPU with "more" mhz, Asterisk is more IO
> than computation. The quad core CPU is lower in MHZ and all the CPUs
> share the same IO.
>
> The architecture design of the XEON board is wrong for really IO
> intensive apps, consider stepping up to a AMD arch.
>
> Also the 2950 is a 2U server that use IRQ sharing (WHY?), wrong with
> boards like the T1 Digium..
>
> Bye.
>
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Re: [asterisk-users] POE draw on Aastra 480i

[Octavio Ruiz]

> > Allen Casteran wrote:
> > Anyone know what the POE draw is for the Aastra 480i phones?
> > We have switches that will do 15 watts on 12 ports but only do 7.7 watts on 
> > all 24 ports.
> > A Cisco 3560 switch will do 15.6 watts on all 24 ports.
> > Just trying to find out if we need that much power.

> Drew wrote:
> According to Aastra tech support, 5 watts (peak) per 480i.
> We are testing five phones running on a Linksys SRW208P that will only 
> support full 15W on up to
> 4 of 8 ports. I can power up the switch while all phones are connected 
> without any issues.
> I would expect your lower power switch will provide ample power.

But, PoE class does not matter? Did you plug five Aastra phones?

I'm suspicious about how that scenario worked, I mean, as far as i
know Aastra phones should register as a zero PoE class, that means it
would reserve up to 12.94 watts no matter how many watts uses.  So, my
guess here is even if the phone use only 5 watts, the switch already
reserved 12.94 watts for it. I would love to see what happens if you
plug a sixth phone or figure out if you used an Aastra phone. Can you
tell us what model/brand you used?

Dimensioning PoE devices over capable switches has been a new issue
which involves many factors like those described before.

Regards,


PD. Sorry about  the original thread break off, I've been unable to
find the original one.
-- 
Octavio H. Ruiz Cervera
Neocenter, SA. de CV.
http://www.neocenter.com/
Soluciones para Centros de Contacto y Telefonía IP
Tel.: (+52 55) 8590-9000 Ext. 9016
Mobile: (+55 155) 5514-087790
Mobile: (+55 155) 5541-351242

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Re: [asterisk-users] When can I AIG?

[Guilherme Loch Waltrick Góes]

Have a look at asterisk-java.org. I has everything you need.

On Jan 29, 2008 4:35 PM, Evan Ruff <[EMAIL PROTECTED]> wrote:

>   Hey Guys,
>
>
>
> I've been doing some research into the AGI-Java connector and was
> wondering if somebody could help me with my architecture.
>
>
>
> What I'd like to do, is kick off an external java class when a user:
>
> 1. Initiates an outgoing call
>
> 2. Hangs up the outgoing call
>
> 3. Has an incoming call
>
> 4. Hangs up the incoming call
>
> 5. Misses a call
>
> 6. Has a voicemail
>
>
>
> I'd also like to be able to access all the call details (user number, ext
> number, start time, end time) I see lot of references to the dial plan
> instantiating the AGI in the tutorials, but I was wondering if it is
> possible to use it in these scenarios?
>
>
>
> How would I need to connect these things up for each user?
>
>
>
> Sorry if this question is a big basic/elementary, but I'm just getting
> started with Asterisk and would really appreciate some help. Thanks a ton!
>
>
>
> Evan Ruff
>
>
>
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-- 
Guilherme Loch Góes

Visite nossa loja virtual: http://www.shopvoip.com.br

Notícias e Fórum sobre VoIP com software livre:
http://www.asteriskexperts.com.br
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[asterisk-users] When can I AIG?

[Evan Ruff]

Hey Guys,



I've been doing some research into the AGI-Java connector and was wondering if 
somebody could help me with my architecture.



What I'd like to do, is kick off an external java class when a user:

1. Initiates an outgoing call

2. Hangs up the outgoing call

3. Has an incoming call

4. Hangs up the incoming call

5. Misses a call

6. Has a voicemail



I'd also like to be able to access all the call details (user number, ext 
number, start time, end time) I see lot of references to the dial plan 
instantiating the AGI in the tutorials, but I was wondering if it is possible 
to use it in these scenarios?



How would I need to connect these things up for each user?



Sorry if this question is a big basic/elementary, but I'm just getting started 
with Asterisk and would really appreciate some help. Thanks a ton!



Evan Ruff

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Re: [asterisk-users] Asterisk mem leak behavior?

[Kevin Kiely]

 
 
 
Kevin, 
 
After upgrading to the latest build of everything have you seen the problem
anymore?
 
Don't know yet, waiting for it to break ( not a good feeling as you know)
 
What's your hardware and software configs? Maybe we can find a similarity in
our systems. 
 
It's a dell poweredge with a basic config, 23b + 1 d,  Sangoma a101,
hw-d-channel.
 
What HW are you using and are you using any SPANDSP?
 

- Mark
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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Grey Man]

- Original Message 
> From: Richard Revels <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Tuesday, 29 January, 2008 12:21:16 PM
> Subject: Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's
> 
> It's not Asterisk, it's SIP.  Transfer takes the signaling off the  
> Asterisk box.
> 
> In features.conf, replace blind transfer with a call to a macro. 
> Then
> 
  
> redo your dialplan with the 'g' option on inward dial commands. When  
> the called party uses the transfer command, your macro should read
> the
> 
  
> digits to call and then store them in the db, a unique global, or
> GROUP
> 
 
> () variable. Then it should hang up. This will cause the calling leg  
> to exit the dial command to the next priority which should be a check  
> of the variable. If digits are present, use the dial command to call  
> them at your provider. No fuss, no muss.
> 
> You should make sure the peer entry for the outbound side includes  
> canreinvite=yes so only the signaling remains on your box and the  
> media is invited off.
> 
> You should also ignore calls to your macro that hit from the inbound  
> call leg. Just return immediately and neither side will ever know the  
> inbound call leg left for a moment.
> 
> Sent from my iPhone
 

Hi Richard,

I'm not actually sure we're talking about the same thing here. It's not 
transfers I have a problem with it's the CDRs the transferred calls end up 
generating. In this case I am the provider and transfers through our Asterisk 
servers work fine it's just that we can't properly bill for them.

Regards,

Greyman.



  Make the switch to the world's best email. Get the new Yahoo!7 Mail now. 
www.yahoo7.com.au/worldsbestemail



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Re: [asterisk-users] Do Asterisk requires audio codec to be installed?

[Rajeev Natarajan]

Asterisk supports a whole bunch of codecs in the regular install -
ulaw, alaw, gsm,ilbc being the more popular ones. A common paid codec
is g729 - avbl at digium.com

-rajeev



On 1/29/08, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> Can you please tell me whether Asterisk requires any audio or video codec to
> be installed separately or it supports itself?
>
>
> Thanking you,
>
> Preeta
>
> Please do not print this email unless it is absolutely necessary. Spread
> environmental awareness.
>
> The information contained in this electronic message and any attachments to
> this message are intended for the exclusive use of the addressee(s) and may
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> the intended recipient, you should not disseminate, distribute or copy this
> e-mail. Please notify the sender immediately and destroy all copies of this
> message and any attachments.
>
> WARNING: Computer viruses can be transmitted via email. The recipient should
> check this email and any attachments for the presence of viruses. The
> company accepts no liability for any damage caused by any virus transmitted
> by this email.
>
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>

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Re: [asterisk-users] Asterisk mem leak behavior?

[Mark Greene]

Kevin,

After upgrading to the latest build of everything have you seen the problem
anymore?

What's your hardware and software configs? Maybe we can find a similarity in
our systems.

- Mark
On Jan 29, 2008 9:53 AM, Kevin Kiely <[EMAIL PROTECTED]> wrote:

>  Mark,
>
>
>
> I thought I would chime in here on your problem.  Oddly, I have having the
> same issue with a PRI with similar symptoms.  The odd part is that I have
> never had an issue like this with a asterisk PRI setup. My setup is a PRI
> with a Sangoma card with the exact same issue with 1.4.14.  After a few
> days we are unable to communicate with the PRI,  The D-channel goes
> offline as well but the physical circuit stays up with no alarms.  It
> doesn't give one a comfort level with uptime.  I had also re-compiled the
> asterisk 1.4.14 along with zaptel and libpri sources and it still failed.
>  I have since updated to the latest asterisk, zaptel and libpri .17 with
> the hopes that it will be fixed.  I thought perhaps the card may have had
> an issue but now I am beginning to wonder.
>
>
>
> Kevin
>
>
>
>
>  --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Mark Greene
> *Sent:* Tuesday, January 29, 2008 9:49 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Asterisk mem leak behavior?
>
>
>
>
> I've tried exiting the CLI in hopes that my being in there, though it
> wouldn't make any sense, was keeping it from restarting. No luck.
>
>
>
> I've already setup a cron script to restart asterisk at night when there
> is no traffic going over it. But I hate to just treat the symptoms. I want
> to solve the problem. It's hard to sleep knowing there is a "ghost" in one
> of my machines.
>
>
>
> It only takes restarting asterisk, nothing else, including zaptel. Once
> asterisk restarts it's ready to go.
>
>
>
> I can't make heads or tails of it. There are no PRI errors when all this
> going on either. Debug shows nothing by usual comm chatter between the
> system and C/O.
>
>
>
> - Mark
>
>
>
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date:
> 1/28/2008 9:32 PM
>
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[asterisk-users] softmodems bank for ast.

[Pepe Aracil]

Hi.

We need a full featured modem bank 20+ to attend data calls.

IAXmodem only supports fax protocols because spandsp only support fax protocols.

The idea is to do a IAX wrapper like IAXmodem but with a full featured
(but propietary) softmodem library like PCTEL or linuxant.


I hate astribank+hardmodems solution.

Some body think that is possible?

Sorry for my horrible English. ;)

Bye.


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Re: [asterisk-users] Asterisk 1.4.18-rc2 Now Available

[Dave Fullerton]

Alex Balashov wrote:
> What is the command to obtain official release notes?
> 
> On Tue, 29 Jan 2008, The Asterisk Development Team wrote:
> 
>> Asterisk 1.4.18-rc2 is now available.  One of the developers made a change to
>> chan_sip that they wanted to get in to this release.  A few other bug fixes 
>> were
>> added, as well.
>>
>> This release candidate is published for anyone that is interested in helping 
>> to
>> test it for a couple of days before it is officially released.  To download 
>> the
>> release candidate, use the following svn command:
>>
>> $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2
>>
>> If you would like it in tarball format, use the following commands:
>>
>> $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 
>> asterisk-1.4.18-rc2
>> $ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/
>>
>> Thanks!


The ChangeLog file will be in the svn checkout. If you only want to 
download the ChangeLog you can use this:

svn export http://svn.digium.com/svn/asterisk/tags/1.4.18/ChangeLog

-Dave

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Re: [asterisk-users] Asterisk 1.4.18-rc2 Now Available

[Alex Balashov]

What is the command to obtain official release notes?

On Tue, 29 Jan 2008, The Asterisk Development Team wrote:

> Asterisk 1.4.18-rc2 is now available.  One of the developers made a change to
> chan_sip that they wanted to get in to this release.  A few other bug fixes 
> were
> added, as well.
>
> This release candidate is published for anyone that is interested in helping 
> to
> test it for a couple of days before it is officially released.  To download 
> the
> release candidate, use the following svn command:
>
> $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2
>
> If you would like it in tarball format, use the following commands:
>
> $ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 
> asterisk-1.4.18-rc2
> $ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/
>
> Thanks!
>
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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[asterisk-users] Asterisk 1.4.18-rc2 Now Available

[The Asterisk Development Team]

Asterisk 1.4.18-rc2 is now available.  One of the developers made a change to 
chan_sip that they wanted to get in to this release.  A few other bug fixes 
were 
added, as well.

This release candidate is published for anyone that is interested in helping to 
test it for a couple of days before it is officially released.  To download the 
release candidate, use the following svn command:

$ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2

If you would like it in tarball format, use the following commands:

$ svn export http://svn.digium.com/svn/asterisk/tags/1.4.18 asterisk-1.4.18-rc2
$ tar -czvf asterisk-1.4.18-rc2.tar.gz asterisk-1.4.18-rc2/

Thanks!

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Re: [asterisk-users] Asterisk mem leak behavior?

[Kevin Kiely]

Mark,
 
I thought I would chime in here on your problem.  Oddly, I have having the
same issue with a PRI with similar symptoms.  The odd part is that I have
never had an issue like this with a asterisk PRI setup. My setup is a PRI
with a Sangoma card with the exact same issue with 1.4.14.  After a few days
we are unable to communicate with the PRI,  The D-channel goes offline as
well but the physical circuit stays up with no alarms.  It doesn't give one
a comfort level with uptime.  I had also re-compiled the asterisk 1.4.14
along with zaptel and libpri sources and it still failed.  I have since
updated to the latest asterisk, zaptel and libpri .17 with the hopes that it
will be fixed.  I thought perhaps the card may have had an issue but now I
am beginning to wonder.
 
Kevin
 
 
  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene
Sent: Tuesday, January 29, 2008 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk mem leak behavior?
 

I've tried exiting the CLI in hopes that my being in there, though it
wouldn't make any sense, was keeping it from restarting. No luck. 
 
I've already setup a cron script to restart asterisk at night when there is
no traffic going over it. But I hate to just treat the symptoms. I want to
solve the problem. It's hard to sleep knowing there is a "ghost" in one of
my machines. 
 
It only takes restarting asterisk, nothing else, including zaptel. Once
asterisk restarts it's ready to go. 
 
I can't make heads or tails of it. There are no PRI errors when all this
going on either. Debug shows nothing by usual comm chatter between the
system and C/O. 
 
- Mark
 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date: 1/28/2008
9:32 PM
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Re: [asterisk-users] SET with pipe symbol

[Tilghman Lesher]

On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
> I want to place a pipe symbol in a variable by using the command Set
> I tried the following code:
> Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
>
> When I call to my applicatie I see the following output in my CLI :
> Ignoring entry '612345678' with no = (and not last 'options'
> entry)
> (in my test call ${CALLERID(number) = 061234578)
>
> I tried to escape the pipe symbol by using \ (backslash)
> With the same result
> Also I tried to place the variable between single or double quotes, but
> with the same result.
>
> Does anybody now how place a pipe symbol in variable.

You can't, in 1.4.  This is by design.  We have removed this restriction in
1.6.  As a workaround, in 1.4, use the NoOp instruction with the SET dialplan
function, i.e.
NoOp(${SET(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))})

-- 
Tilghman

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[asterisk-users] codec_g729a.so problem...

[arkda]

Recently with Asterisk 1.4.17 I've been running into some stability issues.
I started looking through my logs, and I found this:

[Jan 29 09:41:45] WARNING[13132]: loader.c:620 inspect_module: Module
'codec_g729a.so' was not compiled against a recent version of Asterisk and
may cause instability.

I'm using the newest version of codec_g729a.so from the Digium website
(v33). I've tried using the 686, 586, and 386 versions (my platform is 32
bit).

Is there a version that has been properly compiled for 1.4.17? I never had
this problem with previous versions of Asterisk (1.4.15 and below), and I
can't seem to find any information on this.
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[asterisk-users] test please ignore

[Ian]

Just testing to see if my emails to this mailing list gets through. 
Tried posting a question, but it failed

Thanks

Ian

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Re: [asterisk-users] test please ignore

[SIP]

Ian wrote:
> Just testing to see if my emails to this mailing list gets through. 
> Tried posting a question, but it failed
>
> Thanks
>
> Ian
>
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>   
I as well have been having rotten luck lately with the mailing list. 
Replies to questions get axed. New questions get axed.  I mailed the 
list admin, and never got a reply... for all I know, the mail I sent got 
axed.

N.

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Re: [asterisk-users] Asterisk mem leak behavior?

[Mark Greene]

I've tried exiting the CLI in hopes that my being in there, though it
wouldn't make any sense, was keeping it from restarting. No luck.

I've already setup a cron script to restart asterisk at night when there is
no traffic going over it. But I hate to just treat the symptoms. I want to
solve the problem. It's hard to sleep knowing there is a "ghost" in one of
my machines.

It only takes restarting asterisk, nothing else, including zaptel. Once
asterisk restarts it's ready to go.

I can't make heads or tails of it. There are no PRI errors when all this
going on either. Debug shows nothing by usual comm chatter between the
system and C/O.

- Mark
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Re: [asterisk-users] Dial agent channel - busy

[Thomas Kenner]

show agents:

6001 (First Agent) available at '6001' (musiconhold is 'default')
6002 (Test Agent) available at '6002' (musiconhold is 'default')
2 agents configured [2 online , 0 offline]
__


show queues:

testQueue2   has 0 calls (max unlimited) in 'ringall' strategy (10s holdtime), 
W:0, C:1, A:0, SL:100.0% within 60s
   Members:
  Agent/6002 (Not in use) has taken 1 calls (last was 88511 secs ago)
   No Callers

testQueuehas 0 calls (max unlimited) in 'ringall' strategy (11s holdtime), 
W:0, C:1, A:0, SL:100.0% within 60s
   Members:
  Agent/6001 (Not in use) has taken 1 calls (last was 101522 secs ago)
   No Callers
__


Thomas

On Tuesday 29 January 2008 02:13:10 Paul Hales wrote:
> What does 'show agents' give you? 'show queues' would be useful too.
>
> PaulH


-- 
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Re: [asterisk-users] OT: Call for beta testers (well... perhaps late Alpha).

[SIP]

SIP wrote:
> We've just launched the beta of a free service which is, really, still
> only JUST out of the alpha stages.
>
> http://www.voipmagnet.com
>
> The basic idea is this: it's an opt-in directory focused on VoIP contact
> info (with elements of social networking and privacy control).
>
> Again, the service is very rough, but we'd like input from the VoIP
> community.  There are a good many things that are likely buggy, broken,
> or not yet implemented, but we feel that's what a beta is for. If you
> have any questions/concerns/issues/feedback/abuse, feel free to send it
> to me directly via email or post it to the list.
>
> Some things we know will be changed soon:
> -ability to add multiple VoIP accounts with each login (this is in the
> DB, but the interface elements are there yet -- we didn't like the way
> they looked first pass 'round)
> -ability to invite friends to join up so you can share contact info with
> friend groups without making everything (or anything) public
> -search optimisations (searching is functional, but a bit rough. We're
> looking for any and all input there, of course)
>
> We've a whole plate load of ideas for what to cram into the service, but
> the intent is to keep it free and provider-agnostic, but still maintain
> a centralised location where anyone can go and look up friends or
> coworkers to find out what VoIP service they're using and how to get in
> touch.
>
> We're open to any and all suggestions to what to add/change/fix to make
> this a service out of which the community will get some use.
>
>
>   


We're getting some good feedback on the service, and we've made some
optimisations to the search functionality, but we're still looking for
some aggressive testing. If anyone has a second or two to check things
out, we'd greatly appreciate it.


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[asterisk-users] SET with pipe symbol

[Arjan Kroon | Mobillion]

Hi,

 

I want to place a pipe symbol in a variable by using the command Set

I tried the following code:

Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))

 

When I call to my applicatie I see the following output in my CLI :

Ignoring entry '612345678' with no = (and not last 'options'
entry)

(in my test call ${CALLERID(number) = 061234578)

 

I tried to escape the pipe symbol by using \ (backslash)

With the same result

Also I tried to place the variable between single or double quotes, but
with the same result.

 

Does anybody now how place a pipe symbol in variable.

 

Kind Regards,

 

 

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Re: [asterisk-users] transcoder

[Greg Oliver]

Cisco routers with DSPs as ip2ip gw will do it if you want to spend a  
few bucks

On Jan 29, 2008, at 2:36 PM, "Khaled Chehab" <[EMAIL PROTECTED]>  
wrote:

> Dears
>
> Any one knows a standalone voip transcoder software name,not an ip  
> pbx.
> What I want is to  transcode the incoming sip calls from g711 to  
> g723 or
> ilbc or g729 . and forward it to a media gateway ..
>
>
> Regards
>
> Khaled chehab
>
>
>
>
> *
> No employee or agent is authorized to conclude any binding agreement  
> on behalf of Xplorium with another party by e-mail without express  
> written confirmation by an officer of Xplorium. Any views expressed  
> by an individual in this electronic message do not necessarily  
> reflect views of Xplorium or its subsidiaries and associates.
>
> This electronic message and its attachments are solely addressed to  
> the addressee(s), and contain confidential information protected  
> from disclosure belonging to Xplorium.
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> If you are not the intended addressee of this electronic message and  
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Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Atis Lezdins]

On 1/29/08, Richard Revels <[EMAIL PROTECTED]> wrote:
> It's not Asterisk, it's SIP.  Transfer takes the signaling off the
> Asterisk box.
>
> In features.conf, replace blind transfer with a call to a macro.  Then
> redo your dialplan with the 'g' option on inward dial commands. When
> the called party uses the transfer command, your macro should read the
> digits to call and then store them in the db, a unique global, or GROUP
> () variable. Then it should hang up. This will cause the calling leg
> to exit the dial command to the next priority which should be a check
> of the variable. If digits are present, use the dial command to call
> them at your provider. No fuss, no muss.
>
> You should make sure the peer entry for the outbound side includes
> canreinvite=yes so only the signaling remains on your box and the
> media is invited off.
>
> You should also ignore calls to your macro that hit from the inbound
> call leg. Just return immediately and neither side will ever know the
> inbound call leg left for a moment.
>
> Sent from my iPhone
>
> On Jan 28, 2008, at 11:56 PM, Grey Man <[EMAIL PROTECTED]> wrote:
>
> > Hi All,
> >
> > PLEASE READ if you depend on Asterisk CDR's and support transfers.
> >
> > Apologies for the shout but I'm desperate to get others to agree
> > Asterisk has a
> > big problem with the CDR's that are generated for transfers. I can
> > understand
> > why not too many people are interested as transfers are complicated
> > and
> > messy. However for those of us having to support transfers and
> > depending on
> > Asterisk CDR's for our billing we are in a sticky predicament! For
> > anyone
> > using Asterisk in a provider environment unaware of any problem I
> > urge you to
> > do a simple blind transfer on your system and check your CDR's. Most
> > Asterisk
> > based providers I tested are blocking transfers but I did find some
> > other
> > providers out there missing billable call legs!
> >
> > My goal is to try and get acknowledgement that there is a serious
> > problem
> > here that warrants a re-think about how Asterisk CDR's are generated.
> >
> > In an effort to succinctly encapsulate the problem I've produced the
> > call and CDR
> > flows below. Hopefully they make sense but if not I'm more than
> > happy to elaborate
> > and share my test results (the flows below won't be legibile without
> > a mono spaced
> > font, copy and pasting into notepad will make them readable).
> >
> > Blind Transfer (1.2 and 1.4):
> >
> > Time   CallsCDRs
> > | Dest  | Dur(s) |
> > |---||
> > T0 -| Alice --> * --> Bob   |   ||
> > |   |   ||
> > Tt -| Carol <-- * --> Bob  -|  Bob  |   Tt   |
> > |   |   ||
> > Te -| End  -| Carol |   Te   |
> >
> >
> > Attended Transfer (1.2):
> >
> > Time   CallsCDRs
> > | Dest  | Dur(s) |
> > |---||
> > T0 -| Alice --> * --> Bob   |   ||
> > |   |   ||
> > T1 -| Alice --> * --> Carol |   ||
> > |   |   ||
> > Tt -| Carol <-- * --> Bob  -| Bob   |   Tt   |
> > |   | Carol | Tt - T1|
> > |   |   s   |   Tt   |
> > |   |   ||
> > Te -| End  -|   s   |   Te   |
> >
> >
> > Attended Transfer (1.4):
> >
> > Time   CallsCDRs
> > | Dest  | Dur(s) |
> > |---||
> > T0 -| Alice --> * --> Bob   |   ||
> > |   |   ||
> > T1 -| Alice --> * --> Carol |   ||
> > |   |   ||
> > Tt -| Carol <-- * --> Bob  -|   ||
> > |   |   ||
> > Te -| End  -|  Bob  |   Te   |
> > |  Bob  | Te - T1|
> >
> > To put it another way here are some examples of how Asterisk systems
> > and
> > transfers can be exploited.
> >
> > 1. Place a call to a mobile you plan on having a lengthy call to. As
> > soon as the
> > call is establised blind transfer it to a low or free cost
> > destination. You will
> > only be billed for the mobile call up to the time it takes you to do
> > the transfer
> > the remainder of the call will be billed at the low cost or free
> > destination.
> >
> > 2. With Asterisk 1.4 place a call to two billable destinations and
> > then transfer
> > them together. You'll only be billed for each destination up until
> > the time it takes
> > you to transfer.
> >
> > 3. With Asterisk 1.2 place a call to a low cost or free destination.
> > Then place a
> > call to an expensive destination and do an attended transfer. You'll
> > only be
>

Re: [asterisk-users] Asterisk's DANGEROUS Transfer CDR's

[Richard Revels]

It's not Asterisk, it's SIP.  Transfer takes the signaling off the  
Asterisk box.

In features.conf, replace blind transfer with a call to a macro.  Then  
redo your dialplan with the 'g' option on inward dial commands. When  
the called party uses the transfer command, your macro should read the  
digits to call and then store them in the db, a unique global, or GROUP 
() variable. Then it should hang up. This will cause the calling leg  
to exit the dial command to the next priority which should be a check  
of the variable. If digits are present, use the dial command to call  
them at your provider. No fuss, no muss.

You should make sure the peer entry for the outbound side includes  
canreinvite=yes so only the signaling remains on your box and the  
media is invited off.

You should also ignore calls to your macro that hit from the inbound  
call leg. Just return immediately and neither side will ever know the  
inbound call leg left for a moment.

Sent from my iPhone

On Jan 28, 2008, at 11:56 PM, Grey Man <[EMAIL PROTECTED]> wrote:

> Hi All,
>
> PLEASE READ if you depend on Asterisk CDR's and support transfers.
>
> Apologies for the shout but I'm desperate to get others to agree  
> Asterisk has a
> big problem with the CDR's that are generated for transfers. I can  
> understand
> why not too many people are interested as transfers are complicated  
> and
> messy. However for those of us having to support transfers and  
> depending on
> Asterisk CDR's for our billing we are in a sticky predicament! For  
> anyone
> using Asterisk in a provider environment unaware of any problem I  
> urge you to
> do a simple blind transfer on your system and check your CDR's. Most  
> Asterisk
> based providers I tested are blocking transfers but I did find some  
> other
> providers out there missing billable call legs!
>
> My goal is to try and get acknowledgement that there is a serious  
> problem
> here that warrants a re-think about how Asterisk CDR's are generated.
>
> In an effort to succinctly encapsulate the problem I've produced the  
> call and CDR
> flows below. Hopefully they make sense but if not I'm more than  
> happy to elaborate
> and share my test results (the flows below won't be legibile without  
> a mono spaced
> font, copy and pasting into notepad will make them readable).
>
> Blind Transfer (1.2 and 1.4):
>
> Time   CallsCDRs
> | Dest  | Dur(s) |
> |---||
> T0 -| Alice --> * --> Bob   |   ||
> |   |   ||
> Tt -| Carol <-- * --> Bob  -|  Bob  |   Tt   |
> |   |   ||
> Te -| End  -| Carol |   Te   |
>
>
> Attended Transfer (1.2):
>
> Time   CallsCDRs
> | Dest  | Dur(s) |
> |---||
> T0 -| Alice --> * --> Bob   |   ||
> |   |   ||
> T1 -| Alice --> * --> Carol |   ||
> |   |   ||
> Tt -| Carol <-- * --> Bob  -| Bob   |   Tt   |
> |   | Carol | Tt - T1|
> |   |   s   |   Tt   |
> |   |   ||
> Te -| End  -|   s   |   Te   |
>
>
> Attended Transfer (1.4):
>
> Time   CallsCDRs
> | Dest  | Dur(s) |
> |---||
> T0 -| Alice --> * --> Bob   |   ||
> |   |   ||
> T1 -| Alice --> * --> Carol |   ||
> |   |   ||
> Tt -| Carol <-- * --> Bob  -|   ||
> |   |   ||
> Te -| End  -|  Bob  |   Te   |
> |  Bob  | Te - T1|
>
> To put it another way here are some examples of how Asterisk systems  
> and
> transfers can be exploited.
>
> 1. Place a call to a mobile you plan on having a lengthy call to. As  
> soon as the
> call is establised blind transfer it to a low or free cost  
> destination. You will
> only be billed for the mobile call up to the time it takes you to do  
> the transfer
> the remainder of the call will be billed at the low cost or free  
> destination.
>
> 2. With Asterisk 1.4 place a call to two billable destinations and  
> then transfer
> them together. You'll only be billed for each destination up until  
> the time it takes
> you to transfer.
>
> 3. With Asterisk 1.2 place a call to a low cost or free destination.  
> Then place a
> call to an expensive destination and do an attended transfer. You'll  
> only be
> billed for the expensive destination up unitl the time it takes to  
> do the transfer.
>
> I have opened a bug on the issue but I suspect without input from  
> others having
> the same problem it will fade away.
> http://bugs.

Re: [asterisk-users] PRI Alarms, Comes Back, But Asterisk Won't Touch It!

[Tim Panton]

On 29 Jan 2008, at 11:08, George Pajari wrote:

> Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P
> (same problem with various previous versions; same problem with
> different TE120P cards).
>
> The customer has a partial (10 B-Channel) PRI that when it is busy
> (eight or more B channels in use), tends to fail as shown below...
>
> [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown
> error 500
> [Jan 26 23:00:31] ERROR[31893] chan_zap.c: Short write: 0/15 (Unknown
> error 500)
> [Jan 26 23:00:31] WARNING[31893] chan_zap.c: Detected alarm on channel
> 1: Red Alarm
>
> we then see every channel fail with a write error followed by a Red
> Alarm. Then
>
> [Jan 26 23:00:34] VERBOSE[7646] logger.c:   == Primary D-Channel on  
> span
> 1 down
> [Jan 26 23:00:34] WARNING[7646] chan_zap.c: No D-channels available!
> Using Primary channel 24 as D-channel anyway!
> [Jan 26 23:00:36] NOTICE[7647] chan_zap.c: Alarm cleared on channel 1
> (alarm cleared messages for all channels deleted)
> [Jan 26 23:00:36] NOTICE[7646] chan_zap.c: PRI got event: No more  
> alarm
> (5) on Primary D-channel of span 1
>
> Yet even with all alarms cleared, a "pri show span 1" command shows
> "Status: Provisioned, Down, Active". It appears that Asterisk is not
> recovering from the errors.
>
> Restarting Asterisk will not bring the PRI back up -- that requires  
> the
> zaptel drivers to be unloaded and reloaded.
>
> Why is this happening and what can be done about it?
>
> -

What are the Telco techs seeing? I my experience (in the UK at least)  
it is always
worth having a chat with them to see how a PRI problem looks from  
their end.

At a guess (and an un-informed one at that) I'd say that they are  
waiting to
see the line cycle completely before attempting to recover from the  
red alarm.

Reloading the zaptel drivers does something pretty low level (drops
then brings back the carrier?) which they see and respond to.

We had one (failover) circuit which was configured to only recover when
physically unplugged!


Tim.



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[asterisk-users] PRI Alarms, Comes Back, But Asterisk Won't Touch It!

[George Pajari]

Here is the scenario: Asterisk 1.14.13; zaptel 1.4.6; Digium TE120P 
(same problem with various previous versions; same problem with 
different TE120P cards).

The customer has a partial (10 B-Channel) PRI that when it is busy 
(eight or more B channels in use), tends to fail as shown below...

[Jan 26 23:00:31] ERROR[31893] chan_zap.c: Write to 28 failed: Unknown 
error 500
[Jan 26 23:00:31] ERROR[31893] chan_zap.c: Short write: 0/15 (Unknown 
error 500)
[Jan 26 23:00:31] WARNING[31893] chan_zap.c: Detected alarm on channel 
1: Red Alarm

we then see every channel fail with a write error followed by a Red 
Alarm. Then

[Jan 26 23:00:34] VERBOSE[7646] logger.c:   == Primary D-Channel on span 
1 down
[Jan 26 23:00:34] WARNING[7646] chan_zap.c: No D-channels available!  
Using Primary channel 24 as D-channel anyway!
[Jan 26 23:00:36] NOTICE[7647] chan_zap.c: Alarm cleared on channel 1
(alarm cleared messages for all channels deleted)
[Jan 26 23:00:36] NOTICE[7646] chan_zap.c: PRI got event: No more alarm 
(5) on Primary D-channel of span 1

Yet even with all alarms cleared, a "pri show span 1" command shows 
"Status: Provisioned, Down, Active". It appears that Asterisk is not 
recovering from the errors.

Restarting Asterisk will not bring the PRI back up -- that requires the 
zaptel drivers to be unloaded and reloaded.

Why is this happening and what can be done about it?

-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
   www.netvoice.ca  www.ip-centrex.ca  www.ip-pbx.ca  www.vpas.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102) 


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[asterisk-users] transcoder

[Khaled Chehab]

Dears

Any one knows a standalone voip transcoder software name,not an ip pbx.
What I want is to  transcode the incoming sip calls from g711 to g723 or
ilbc or g729 . and forward it to a media gateway ..


Regards

Khaled chehab 




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electronic message do not necessarily reflect views of Xplorium or its 
subsidiaries and associates.

This electronic message and its attachments are solely addressed to the 
addressee(s), and contain confidential information protected from disclosure 
belonging to Xplorium.

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Xplorium does not guarantee the integrity of this electronic message and any of 
its attachments, or that they are free from computer viruses or other defects.
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Re: [asterisk-users] Dialogic card

[Steve Totaro]

On Jan 28, 2008 11:12 PM, Edgar Guadamuz <[EMAIL PROTECTED]> wrote:
> Hi list,
>
> Anyone knows where I can get information about configuring a Dialogic
> card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody
> told me that I had to buy the driver, but I don't know if this is true
> and if so, who, how and how much...
>

I was under the impression that only ABE supports Dialogic boards.  I
thought I saw that in passing so I could be totally wrong.

Thanks,
Steve Totaro

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[asterisk-users] Do Asterisk requires audio codec to be installed?

[preeta.pandey]

Hi,

Can you please tell me whether Asterisk requires any audio or video codec to be 
installed separately or it supports itself?


Thanking you,

Preeta

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Re: [asterisk-users] Change Default Voicemail Message

[chris]

Hi Dan,

Sorry to bring a thread back from the dead, but you might find the
following interesting.

> Is there an easy way to achieve this with a computer generated
> voice? We do not wish to manually record the messages if possible,
> in the interests of a consistent message across all voicemail boxes.
> What would be the easiest way to do this?

It was almost as if the guys at Nerd Vittles knew exactly what you
needed and wrote an article about it... :)

http://nerdvittles.com/index.php?p=202

Headline is:
  Allison’s Text-to-Speech Trifecta: Cepstral, Asterisk 1.4, and
  FreePBX 2.3

Opening line:
  If you've longed for a text-to-speech Asterisk toolkit that sounds
  just like the default Allison prompts that ship with Asterisk 1.4,
  then today is your lucky day.

Havn't tried it myself but hopefully it comes close to what you need
it for.

Regards,

Chris Bennett

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Re: [asterisk-users] Asterisk mem leak behavior?

[Lenz]

Really, what I would do is to set up a daily restart point when there is  
no or very little activity, something like running nightly:

asterisk -rx "stop when convenient"

and then having the monitoring script restart it immediately. Do you need  
to unload the zaptel modules as well or is restarting Asterisk enough?
l.

PS.
one of the things I like less about asterisk is that every time you have a  
different piece of iron connected to a PBX, you end up patching zaptel and  
recompiling. so all ot of time you find yourself wondering if the patch  
will be successful with the latest zaptel version or not :-(



On Tue, 29 Jan 2008 07:24:49 +0100, Mark Greene <[EMAIL PROTECTED]>  
wrote:

> So here is my setup.
>
> Hardware:
> Intel P3 1.2 Ghz
> 1 GB RAM
> 36 GB Drives Mirrored
>
> Software:
>  CentOS 5
> 2.6.18 Kernel
>
> Asterisk 1.4.14
> Zaptel 1.4.7 (redfone)
> LIbpri 1.4.2
>


-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.com

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Re: [asterisk-users] Maybe a little OT---USB Handset

[Tim Panton]

>>
> Thanks for the replies.
> I wonder if I could use the Yealink phone and write a connector to
> Asterisk with the IAX client on Sourceforge and make the handset look
> like an iaxphone?  Or maybe there is some other easier solution?   
> All I
> need is to have the ability to go
> off hook/on hook, pass DTMF, and voice obviously :-)
> JohnM


John,  I've done something similar. I had a simple commandline
IAX phone running on a low powered computer (actually an NSLU2 -
but it could be any linux machine), driving a USB audio card. I didn't  
need
a keyboard, as it  auto answered incoming calls.

If you are interested, contact me off list, and perhaps I can help.

Tim.

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