date:20060501

My experience is that a stun server does not do anything that nat=yes in
asterisk does not do. Asterisk is capable of determining the source port
and ip address of a registration, so there is no need for the UA (ATA)
to learn this information form a stun server. Keep it simple if
possible, the stun server just adds another device to manage and/or
worry about being unreachable.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, May 01, 2006 9:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

You will probably want to set a stun server in the 2100 if behind a nat.
You
can use stun.fwdnet.net for testing.  With that, you probably wont need
to
port forward & it should work.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Monday, May 01, 2006 8:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

Set nat=yes as you have
Enable qualify=yes

Important - Do a sip reload or asterisk reload (the nat and qualify
settings have to be refreshed, at least with realtime and
rtcahcefriends).

Turn off all NAT traversal features on the SPA2100

If it still does not work - your NAT router may be the issue, make sure
that security policy allows ALL outbound traffic from the SPA2100 (no
filters).

With Linksys, Belkin, and some 3com/USR NAT routers (among others I am
sure) you will need to make sure you have recent firmware on them, older
firmware (1 year or older in many cases) does not behave well with SIP
and NAT.

The NAT=yes tells asterisk to use the IP address and port of the
connection socket (a form of NAT discovery similar to a STUN server),
not what is in the registration message, and the qualify=yes tells
asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout
from expiring on the NAT router.



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Lyons
> Sent: Monday, May 01, 2006 5:01 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
> 
> I've got a Sipura SPA-1001 that I'm trying to get working with an
Asterisk
> server that's on the public Internet, while the SPA-1001
> is behind NAT.  I did the first obvious thing and mapped ports 5060
and
> 1 - 3 to the local IP address of the SPA-1001.
> Tried numerous proxy settings, have all the NAT settings == yes.
> Registration seems to be happening; with sip debug on, I see it
> get an OK and sip show peers shows it on the list.  But I can't get a
dial
> tone.
> 
> It works fine connecting to a local Asterisk box (not traversing NAT).
> 
> Anyone know the magic trick?  My sip.conf looks like:
> [homesip]
> type=friend
> username=homesip
> secret=
> context=fagi
> ;qualify=yes
> host=dynamic
> nat=yes
> 
> tried qualify both ways.  My sip show peers says:
> 
> telebox*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> homesip/homesip67.188.35.109D   N  5060
> Unmonitored
> 
> Can't seem to find enough info to get this to work, any help
appreciated
> greatly,
> 
> Eric.
> 
> 
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RE: [Asterisk-Users] I am looking for a webphone on MY SITE

2006-05-01 Thread Anton Krall

Anyway to check out Corraleta? :) 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Dean Collins
|Sent: Wednesday, April 26, 2006 1:12 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] I am looking for a webphone on MY SITE
|
|Ahh it's about time you released to market anyway.
|
|
|
|> -Original Message-
|> From: [EMAIL PROTECTED] 
|[mailto:asterisk-users- 
|> [EMAIL PROTECTED] On Behalf Of Tim Panton
|> Sent: Wednesday, 26 April 2006 1:05 PM
|> To: Asterisk Users Mailing List - Non-Commercial Discussion
|> Subject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE
|> 
|> 
|> On 26 Apr 2006, at 16:24, Dean Collins wrote:
|> 
|> > Do a google on Mexuar.
|> 
|> That's my cover blown :-)
|> 
|> Tim Panton
|> [EMAIL PROTECTED]
|> 
|> 
|> 
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RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

2006-05-01 Thread billy

You will probably want to set a stun server in the 2100 if behind a nat. You
can use stun.fwdnet.net for testing.  With that, you probably wont need to
port forward & it should work.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Monday, May 01, 2006 8:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

Set nat=yes as you have
Enable qualify=yes

Important - Do a sip reload or asterisk reload (the nat and qualify
settings have to be refreshed, at least with realtime and
rtcahcefriends).

Turn off all NAT traversal features on the SPA2100

If it still does not work - your NAT router may be the issue, make sure
that security policy allows ALL outbound traffic from the SPA2100 (no
filters).

With Linksys, Belkin, and some 3com/USR NAT routers (among others I am
sure) you will need to make sure you have recent firmware on them, older
firmware (1 year or older in many cases) does not behave well with SIP
and NAT.

The NAT=yes tells asterisk to use the IP address and port of the
connection socket (a form of NAT discovery similar to a STUN server),
not what is in the registration message, and the qualify=yes tells
asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout
from expiring on the NAT router.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Lyons
> Sent: Monday, May 01, 2006 5:01 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
> 
> I've got a Sipura SPA-1001 that I'm trying to get working with an
Asterisk
> server that's on the public Internet, while the SPA-1001
> is behind NAT.  I did the first obvious thing and mapped ports 5060
and
> 1 - 3 to the local IP address of the SPA-1001.
> Tried numerous proxy settings, have all the NAT settings == yes.
> Registration seems to be happening; with sip debug on, I see it
> get an OK and sip show peers shows it on the list.  But I can't get a
dial
> tone.
> 
> It works fine connecting to a local Asterisk box (not traversing NAT).
> 
> Anyone know the magic trick?  My sip.conf looks like:
> [homesip]
> type=friend
> username=homesip
> secret=
> context=fagi
> ;qualify=yes
> host=dynamic
> nat=yes
> 
> tried qualify both ways.  My sip show peers says:
> 
> telebox*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> homesip/homesip67.188.35.109D   N  5060
> Unmonitored
> 
> Can't seem to find enough info to get this to work, any help
appreciated
> greatly,
> 
> Eric.
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] CallerID Name problem

Cross-posted to Dev on purpose.

I missed Astricon (not willing to give up a planned vacation!).

I looks like there is already code that detects the facility IE, as asterisk 
whines about "do not know what to do with a second ROSE component" (not 
verified in current version). That second ROSE component is the message that 
says the CNAM is coming in a facility IE according to the PM sessions I have 
been able to look at.

Why not put a flag in the config? WaitforfacilityIE=yes|no <- this would allow 
the user to decide on a per span basis if it is needed.

The Bellcore spec on CNAM states that the SETUP must contain an "information 
following" message element to indicate that the CNAM will arrive in the 
Facility IE.

Libpri could be written to look for the "information following" element in 
SETUP and wait for the FACIILITY IE if it is detected AND the user flag is set 
to wait for Facility IE.

Of course, when I say "could be written", I mean by someone like Matthew 
Frederickson, not me :). I do not have the required knowledge of the code to 
even try and attempt this.

I spent way too much time on this last year trying to use a TNT as a media 
gateway, now it (the TNT) is an outbound gateway for our 911 trunks and nothing 
else.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 6:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] CallerID Name problem

You explained this very well thank you!!, We discussed (Astricon 2005 Anaheim) 
having LibPri either wait 1 second before passing the call on to asterisk, or 
waiting until CNAME was received, both ideas were not good as it will introduce 
delays for all instead of just those that needed it.

For the time being we will put the Wait(1) in.
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Monday, May 01, 2006 8:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> The CDR is written at the end of the call, the SIP invite is generated
> when you tell it to.
> 
> You must wait for the CID name before you "dial" a SIP channel, as this is
> when the SIP invite is created.
> 
> Wait 1 seems to be long enough.
> 
> For analog TDM this is not an issue as the CNAM is sent between the first
> and second ring.
> 
> In nearly every case I have seen, the facility IE that contains the CNAM
> on most NI-2 ISDN PRIs, the Facility IE is not sent until after you send
> progress back to the telco.
> 
> Like this;
> 
> Telco sends ISDN SETUP message
> You send back ISDN PROGRESS (on a normal call)
> The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM
> 
> I have looked at several protocol monitoring session form both Nortel and
> lucent class 5 switches, and every one seems to send the facility IE in
> less than 500 milliseconds following SETUP assuming you (Zaptel) reply
> PROGRESS in a timely fashion, which does not appear to ever be an issue
> with Asterisk.
> 
> I have been lead to believe there are two reasons for sending the CNAM in
> the facility IE.
> 
> One explanation is that the SETUP message has a restriction on length, and
> in order to provide other features, the CNAM had to be moved.
> 
> The other explanation is that the Telco is saving costs by not dipping
> into LIDB (line information databases) for all calls where PROGRESS is not
> received from the endpoint (a high percentage of all calls). There is a
> cost associated with LIDB dips.
> 
> Both are believable explanations, but the result is the same, you have to
> wait 1 second!
> 
> Sure would be nice if asterisk could wait 500 milliseconds instead, since
> there is a need to put a "play 1 second of silence" in the first step of
> the IVR (if used) to avoid clipping the first phrase also. The combined 2
> seconds of "wait" gets noticed by some users/callers.
> 
> For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of
> "waiting" for CNAM before generating the SIP invite, so it is not
> compatible with PRIs where the CNAM is sent in the facility IE instead of
> the SETUP message, which is the case with most telco switches with current
> software on them.
> 
> Damon
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Josh McAllister
> > Sent: Monday, May 01, 2006 4:25 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] CallerID Name problem
> >
> >
> > >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
> > >Sent: Monday, May 01, 2006 3:06 PM
> > >To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >Subject: Re: [Asterisk-Users] CallerID Name problem
> > >
> > >Do you ge

RE: [Asterisk-Users] CallerID Name problem

You explained this very well thank you!!, We discussed (Astricon 2005 Anaheim) 
having LibPri either wait 1 second before passing the call on to asterisk, or 
waiting until CNAME was received, both ideas were not good as it will introduce 
delays for all instead of just those that needed it.

For the time being we will put the Wait(1) in.
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Monday, May 01, 2006 8:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> The CDR is written at the end of the call, the SIP invite is generated
> when you tell it to.
> 
> You must wait for the CID name before you "dial" a SIP channel, as this is
> when the SIP invite is created.
> 
> Wait 1 seems to be long enough.
> 
> For analog TDM this is not an issue as the CNAM is sent between the first
> and second ring.
> 
> In nearly every case I have seen, the facility IE that contains the CNAM
> on most NI-2 ISDN PRIs, the Facility IE is not sent until after you send
> progress back to the telco.
> 
> Like this;
> 
> Telco sends ISDN SETUP message
> You send back ISDN PROGRESS (on a normal call)
> The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM
> 
> I have looked at several protocol monitoring session form both Nortel and
> lucent class 5 switches, and every one seems to send the facility IE in
> less than 500 milliseconds following SETUP assuming you (Zaptel) reply
> PROGRESS in a timely fashion, which does not appear to ever be an issue
> with Asterisk.
> 
> I have been lead to believe there are two reasons for sending the CNAM in
> the facility IE.
> 
> One explanation is that the SETUP message has a restriction on length, and
> in order to provide other features, the CNAM had to be moved.
> 
> The other explanation is that the Telco is saving costs by not dipping
> into LIDB (line information databases) for all calls where PROGRESS is not
> received from the endpoint (a high percentage of all calls). There is a
> cost associated with LIDB dips.
> 
> Both are believable explanations, but the result is the same, you have to
> wait 1 second!
> 
> Sure would be nice if asterisk could wait 500 milliseconds instead, since
> there is a need to put a "play 1 second of silence" in the first step of
> the IVR (if used) to avoid clipping the first phrase also. The combined 2
> seconds of "wait" gets noticed by some users/callers.
> 
> For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of
> "waiting" for CNAM before generating the SIP invite, so it is not
> compatible with PRIs where the CNAM is sent in the facility IE instead of
> the SETUP message, which is the case with most telco switches with current
> software on them.
> 
> Damon
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Josh McAllister
> > Sent: Monday, May 01, 2006 4:25 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] CallerID Name problem
> >
> >
> > >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
> > >Sent: Monday, May 01, 2006 3:06 PM
> > >To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >Subject: Re: [Asterisk-Users] CallerID Name problem
> > >
> > >Do you get caller ID number?  If so, WAITing is not going to help,
> since
> > you already get the info.  If you >get caller ID number, then your telco
> > is not >sending the name.
> >
> > This is not necessarily true. I've always gotten cID number, but only
> > recently when I added a wait(1) did I start getting channel vars
> populated
> > with cID Name. Same as Eric, I was getting cID Name in the CDR records
> all
> > along as well.
> >
> > Eric -- Go ahead and give it a shot... even if you are getting the cID
> > number. This will likely fix your problem.
> >
> > Josh McAllister
> >
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Using frequent keepalives to eliminate need forNAT port forwarding?

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open,
if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will
work because port 5060 on the private address will still be port 5060 on
the public address.

With PAT the port could be anything over 1024, but usually much higher,
and the originator will send to port 5060, which your NAT router will
drop.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tom Engleward
> Sent: Monday, May 01, 2006 6:25 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Using frequent keepalives to eliminate need
> forNAT port forwarding?
> 
> I have an asterisk system behind NAT, and need to
> connect to public PSTN originators via SIP or IAX2,
> but don't have the option of forwarding any ports
> (4569, 5060, etc) to the asterisk system. However, the
> NAT system does properly establish transient UDP
> forwarding on the basis of outgoing connections, so is
> it possible to configure asterisk to send frequent
> keepalive UDP packets (say every 30 seconds) from
> ports 4569 and 5060 to the PSTN originators in order
> to keep the NAT system's transient forwarding in
> effect, so that when the PSTN originator receives
> inbound calls and attempts to contact my asterisk
> system, the NAT system won't drop the packets?
> 
> 
> __
> Do You Yahoo!?
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> http://mail.yahoo.com
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RE: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

Set nat=yes as you have
Enable qualify=yes

Important - Do a sip reload or asterisk reload (the nat and qualify
settings have to be refreshed, at least with realtime and
rtcahcefriends).

Turn off all NAT traversal features on the SPA2100

If it still does not work - your NAT router may be the issue, make sure
that security policy allows ALL outbound traffic from the SPA2100 (no
filters).

With Linksys, Belkin, and some 3com/USR NAT routers (among others I am
sure) you will need to make sure you have recent firmware on them, older
firmware (1 year or older in many cases) does not behave well with SIP
and NAT.

The NAT=yes tells asterisk to use the IP address and port of the
connection socket (a form of NAT discovery similar to a STUN server),
not what is in the registration message, and the qualify=yes tells
asterisk to send periodic SIP OPTIONS queries to keep the NAT timeout
from expiring on the NAT router.



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Lyons
> Sent: Monday, May 01, 2006 5:01 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling
> 
> I've got a Sipura SPA-1001 that I'm trying to get working with an
Asterisk
> server that's on the public Internet, while the SPA-1001
> is behind NAT.  I did the first obvious thing and mapped ports 5060
and
> 1 - 3 to the local IP address of the SPA-1001.
> Tried numerous proxy settings, have all the NAT settings == yes.
> Registration seems to be happening; with sip debug on, I see it
> get an OK and sip show peers shows it on the list.  But I can't get a
dial
> tone.
> 
> It works fine connecting to a local Asterisk box (not traversing NAT).
> 
> Anyone know the magic trick?  My sip.conf looks like:
> [homesip]
> type=friend
> username=homesip
> secret=
> context=fagi
> ;qualify=yes
> host=dynamic
> nat=yes
> 
> tried qualify both ways.  My sip show peers says:
> 
> telebox*CLI> sip show peers
> Name/username  HostDyn Nat ACL Port Status
> homesip/homesip67.188.35.109D   N  5060
> Unmonitored
> 
> Can't seem to find enough info to get this to work, any help
appreciated
> greatly,
> 
> Eric.
> 
> 
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RE: [Asterisk-Users] CallerID Name problem

Title: RE: [Asterisk-Users] CallerID Name problem








Glad to help. It is NOT a BUG, but it
works that way due to the way CallerID name is transmited to the customer with
some Telcos

 

 











From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 6:29 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem



 

That worked GREAT 

Thank you so so MUCH for your help!!

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 5:06 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem

You don't need the answer, But you need
the wait. CallerID Name comes over the FACILITY messge many times and it takes
a slpit second for it to come in.

 



 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Monday, May 01, 2006 4:34 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
CallerID Name problem

Do
you wait before or after the answer? Do you even need the answer?



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 14:26:49 2006
To: Asterisk Users Mailing
 List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users]
CallerID Name problem

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial
 Discussion
> Subject: [Asterisk-Users] CallerID Name problem
>
>
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
>
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
>
>
> Any help would be great !
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[Asterisk-Users] Using frequent keepalives to eliminate need for NAT port forwarding?

2006-05-01 Thread Tom Engleward

I have an asterisk system behind NAT, and need to
connect to public PSTN originators via SIP or IAX2,
but don't have the option of forwarding any ports
(4569, 5060, etc) to the asterisk system. However, the
NAT system does properly establish transient UDP
forwarding on the basis of outgoing connections, so is
it possible to configure asterisk to send frequent
keepalive UDP packets (say every 30 seconds) from
ports 4569 and 5060 to the PSTN originators in order
to keep the NAT system's transient forwarding in
effect, so that when the PSTN originator receives
inbound calls and attempts to contact my asterisk
system, the NAT system won't drop the packets?


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RE: [Asterisk-Users] CallerID Name problem

The CDR is written at the end of the call, the SIP invite is generated when you 
tell it to.

You must wait for the CID name before you "dial" a SIP channel, as this is when 
the SIP invite is created.

Wait 1 seems to be long enough.

For analog TDM this is not an issue as the CNAM is sent between the first and 
second ring.

In nearly every case I have seen, the facility IE that contains the CNAM on 
most NI-2 ISDN PRIs, the Facility IE is not sent until after you send progress 
back to the telco.

Like this;

Telco sends ISDN SETUP message
You send back ISDN PROGRESS (on a normal call)
The Telco sends ISDN FACILITY INFORMATION ELEMENT (IE) containing CNAM

I have looked at several protocol monitoring session form both Nortel and 
lucent class 5 switches, and every one seems to send the facility IE in less 
than 500 milliseconds following SETUP assuming you (Zaptel) reply PROGRESS in a 
timely fashion, which does not appear to ever be an issue with Asterisk.

I have been lead to believe there are two reasons for sending the CNAM in the 
facility IE.

One explanation is that the SETUP message has a restriction on length, and in 
order to provide other features, the CNAM had to be moved.

The other explanation is that the Telco is saving costs by not dipping into 
LIDB (line information databases) for all calls where PROGRESS is not received 
from the endpoint (a high percentage of all calls). There is a cost associated 
with LIDB dips.

Both are believable explanations, but the result is the same, you have to wait 
1 second!

Sure would be nice if asterisk could wait 500 milliseconds instead, since there 
is a need to put a "play 1 second of silence" in the first step of the IVR (if 
used) to avoid clipping the first phrase also. The combined 2 seconds of "wait" 
gets noticed by some users/callers.

For the archives - The Lucent TNT (as of 11.0 firmware) is not capable of 
"waiting" for CNAM before generating the SIP invite, so it is not compatible 
with PRIs where the CNAM is sent in the facility IE instead of the SETUP 
message, which is the case with most telco switches with current software on 
them.

Damon












> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Josh McAllister
> Sent: Monday, May 01, 2006 4:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] CallerID Name problem
> 
> 
> >From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
> >Sent: Monday, May 01, 2006 3:06 PM
> >To: Asterisk Users Mailing List - Non-Commercial Discussion
> >Subject: Re: [Asterisk-Users] CallerID Name problem
> >
> >Do you get caller ID number?  If so, WAITing is not going to help, since
> you already get the info.  If you >get caller ID number, then your telco
> is not >sending the name.
> 
> This is not necessarily true. I've always gotten cID number, but only
> recently when I added a wait(1) did I start getting channel vars populated
> with cID Name. Same as Eric, I was getting cID Name in the CDR records all
> along as well.
> 
> Eric -- Go ahead and give it a shot... even if you are getting the cID
> number. This will likely fix your problem.
> 
> Josh McAllister
> 
> 
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Re: [Asterisk-Users] CallerID Name problem

2006-05-01 Thread Lacy Moore - Aspendora

Good to know.  I always thought that it was all sent along the D channel in the call setup.  But, until Wednesday, I'm just talking.  After Wednesday, I'll have my PRI.  At least now I know to include the wait.
 
Thanks! 
On 5/1/06, Josh McAllister <[EMAIL PROTECTED]> wrote:
>From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]] On Behalf Of Lacy Moore - Aspendora>Sent: Monday, May 01, 2006 3:06 PM>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] CallerID Name problem>>Do you get caller ID number? If so, WAITing is not going to help, since you already get the info. If you >get caller ID number, then your telco is not >sending the name.
This is not necessarily true. I've always gotten cID number, but only recently when I added a wait(1) did I start getting channel vars populated with cID Name. Same as Eric, I was getting cID Name in the CDR records all along as well.
Eric -- Go ahead and give it a shot... even if you are getting the cID number. This will likely fix your problem.Josh McAllister___--Bandwidth and Colocation provided by 
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-- Lacy MooreAspendora, Inc. 
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[Asterisk-Users] Cisco 2621 router for voice and data?

2006-05-01 Thread T. Shaw

Hey Guys/Gals, 
I was thinking of getting a cisco 2621 router with 2 FastE modules and 1
T1-Wic to use for a small business setup along with asterisk. I want to
provide redundancy between my VOIP provider and telco, or between two VOIP
providers. Also my plans are to setup VLANS to separate internal voice
traffic and regular data traffic using a cisco 2900 switch. Is the 2621
overkill for 5-7 person office? I'm thinking about scalability as well. It
could grow into +5 people within the next 6mos. I was also looking at a
cisco 1760-v modular router.

Thanks for your thoughts!

Terrelle
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[Asterisk-Users] wellgate 38XX with VAD and call files

2006-05-01 Thread Jerry Geis


Has anyone configured the wellgate 38XX sip units with VAD?
I have the unit all working but when I do a call file from
/var/spool/asterisk/outgoing/callfile (below)
--
Channel: SIP/1001/506
Context: smvoice-dialout
Extension: 9
Application: Playback
Data: demo-congrats
--
Where 506 is my cell phone number.
The demo-congrats message is already playing before I answer the call.
What is the trick to have VAD correctly work so the message does not
start to play before I answer the phone.

Thanks,

Jerry

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[Asterisk-Users] SPA-1001 behind NAT -- mucho hair pulling

2006-05-01 Thread Eric Lyons

I've got a Sipura SPA-1001 that I'm trying to get working with an Asterisk server that's on the public Internet, while the SPA-1001 
is behind NAT.  I did the first obvious thing and mapped ports 5060 and 1 - 3 to the local IP address of the SPA-1001. 
Tried numerous proxy settings, have all the NAT settings == yes.  Registration seems to be happening; with sip debug on, I see it 
get an OK and sip show peers shows it on the list.  But I can't get a dial tone.


It works fine connecting to a local Asterisk box (not traversing NAT).

Anyone know the magic trick?  My sip.conf looks like:
[homesip]
type=friend
username=homesip
secret=
context=fagi
;qualify=yes
host=dynamic
nat=yes

tried qualify both ways.  My sip show peers says:

telebox*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
homesip/homesip67.188.35.109D   N  5060 Unmonitored

Can't seem to find enough info to get this to work, any help appreciated 
greatly,

Eric. 



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RE: [Asterisk-Users] Polycom SoundPoint 501 + Asterisk

2006-05-01 Thread Douglas Garstang

I remember a thread about this exact problem a few weeks ago.
You need to upgrade the phone XML config files to the ones that come with the 
version of firmware you are using.

Doug.

> -Original Message-
> From: Kenneth Shaw [mailto:[EMAIL PROTECTED]
> Sent: Monday, May 01, 2006 3:24 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Polycom SoundPoint 501 + Asterisk
> 
> 
> Guys,
> 
> I recently purchased some new SoundPoint 501 phones. The new 
> ones that I
> purchased had a newer firmware than the 501s we had deployed at the
> time. I updated the firmware on the older 501s, and now these phones
> have some very odd echoes with Asterisk, etc. I assume this is due to
> changes in the config file somehow (I love Polycom!), but was 
> wondering
> if anyone else had experienced odd audio problems with the 501s after
> upgrading to the latest bootrom and sip images.
> 
> Any info would be much appreciated! Thanks!
> 
> -- 
> Kenneth Shaw
> Director of Technology
> ExpiTrans, Inc.
> 129 W. Wilson St., Suite 204
> Costa Mesa, CA 92627
> tel: 949 278 7288
> fax: 949 642 6044
> [EMAIL PROTECTED]
> 
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RE: [Asterisk-Users] CallerID Name problem

Title: RE: [Asterisk-Users] CallerID Name problem



That worked GREAT 
Thank you so so MUCH for your 
help!!


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Monday, May 01, 2006 5:06 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
CallerID Name problem

You don't need the answer, But you need the wait. 
CallerID Name comes over the FACILITY messge many times and it takes a slpit 
second for it to come in.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Hall, 
  Eric M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] CallerID Name problem
  
  Do you wait before or after the answer? Do you even need the 
  answer? -Original Message-From:   
  Alexander Lopez [mailto:[EMAIL PROTECTED]]Sent:   
  Mon May 01 14:26:49 2006To: Asterisk Users Mailing 
  List - Non-Commercial 
  DiscussionSubject:    RE: 
  [Asterisk-Users] CallerID Name problemHow are the calls coming into 
  the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP 
  channel.> -Original Message-> From: 
  [EMAIL PROTECTED] [mailto:asterisk-users-> 
  [EMAIL PROTECTED] On Behalf Of Hall, Eric M.> Sent: Monday, May 
  01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial 
  Discussion> Subject: [Asterisk-Users] CallerID Name 
  problem>>> I'm having trouble getting callerid name to 
  show up on my phones(Cisco> 7960 and a few softphones)> When 
  I look in the CDR database I see the name but not on any phonewhen> 
  being called.>> I'm running> Asterisk SVN-trunk-r7498 
  built on 2006-04-30 15:11:39 UTC>>> Any help would be 
  great !> ___> 
  --Bandwidth and Colocation provided by Easynews.com -->> 
  Asterisk-Users mailing list> To UNSUBSCRIBE or update options 
  visit:>    http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth 
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RE: [Asterisk-Users] CallerID Name problem

2006-05-01 Thread Josh McAllister


>From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - 
>Aspendora
>Sent: Monday, May 01, 2006 3:06 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] CallerID Name problem
>
>Do you get caller ID number?  If so, WAITing is not going to help, since you 
>already get the info.  If you >get caller ID number, then your telco is not 
>>sending the name.

This is not necessarily true. I've always gotten cID number, but only recently 
when I added a wait(1) did I start getting channel vars populated with cID 
Name. Same as Eric, I was getting cID Name in the CDR records all along as well.

Eric -- Go ahead and give it a shot... even if you are getting the cID number. 
This will likely fix your problem.

Josh McAllister
 

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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Doug Lytle wrote:

  

Jon,

I don't know.  I went into my voicemail.conf and put a semicolon in
front of that option,

Re-attached to the Asterisk console and did a reload and the option was
no longer available from Advanced Options.

I'm running 1.2.7.1



Do you mean the option was not offered to you or it did not work when
offered. I am also on 1.2.7.1


  

The option is no longer offered.

Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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[Asterisk-Users] Music on Hold from Soundcard

2006-05-01 Thread Alex Robar

Hey 
all,


I've been trying to get MoH to work 
from the line-in on my soundcard, but as of yet have had no success. I found 
this script that should allow for it to happen:http://www.sineapps.com/news.php?rssid=722


The script, when run as the asterisk 
user, works properly and streams sound to stdin. But when Asterisk starts MoH it 
stops it immediately afterwards with no explanation. Has anyone gotten this to 
work? Or does anyone have any ideas on how to debug why MoH stops immediately 
after starting?

Thanks in advance!Alex Robar
 
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Alex Robar,  Technical Support, 
  GearyTech Inc.


3075 Fourteenth Avenue, Unit 3, 
Markham, Ontario L3R 
0G9
Markham: 905-513-8000  x 223  
Fax: 905-513-8040
Toronto
: 416-226-3614      
Toll Free: 888-890-3499
[EMAIL PROTECTED]
 
www.gearytech.com
 
Strategic management of technology 
for business.
 
Have you seen our new 
Voice over IP telephone systems for business? 
Visit 
http://www.gearytech.com/VoIP.asp
 
for more information.-- Alex Robar[EMAIL PROTECTED]
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Re: [Asterisk-Users] billing realtime

2006-05-01 Thread Thameem Ansari

Hello All,

I had the same question when I was writing my own billing software in
java. Here is what I am doing to track multiple calls at a time from
the prepaid account.

1. Keep on db table for balance and reserver_balance.
2. First call coming to agi, check the balance - Sum of all the reserve_balance of that account code.
3. Check the destination and allowed minutes for that balance amount from step 2.
4. Reserve balance table will contain destination, amount, reserved secs columns
5. If the avaialable balance is <= 0 then announce not enough credit and hangup.
6. If the available balance is > 0 but seconds allowed to talk is
less than reserved secs (see step 8 for more details about what this is)
then set absolutetimeout for those seconds.
7. Otherwise the allowed seconds is more than the allowed seconds, set
absolute time out for the reserved seconds and make the call.
8. Reserved secs is a custom constant seconds, say you can reserve fund
for 3 minutes (180 seconds). if the account has balance for only 2
minutes (120 seconds) then the absolute time out will be 120 seconds.
9. Once the channel status changed to reserved, insert an record to
reserve_balance table with uniqueid, accountcode, amount, reserved_secs
information.

The above steps will handle one call so far now...and lets see how the dial plan should be,
10. In your dial plan, add an AbsoluteTimeout extension "T" and call
another AGI script which will just to reset the absolute timeout.
11. When the particular timeout is reached asterisk will transfer the
call to 'T" extension which will in turn call another agi.
12. The agi will receive all the information about the channel
including uniqueid, repeat the steps 2- 7 (except dial) and reset the
abstimeout and this process will repeat until the channel hangup.
13. Once the channel hangup, you can either use Manager to receive the
cdr event or you can set "h" extension (not reliable and not
recommended) to calculate the real balance and update the balance
table. Once you update the balance table, remove the record from
reserve_balance table for the uniqueid, channel and accountcode. (these
three are enough to find out the entry in that table).

Now lets take the scenario for second call when the first call was active,
14. When the second call comes in, start from step 2. In step 2, we are
doing finalBalance = Balance - Sum(reserve_balance) for that account
code. If there is already a call on this accountcode, then this table
will have one entry and the reserved amount. Get the finalBalance by
subtracting the amounts. Follow step 3 and allow or deny the caller.
The above said solution is very stable and doesn't overflow the memory
or session and not using any threads. The only restriction here is, if
we have the scenario,

Call -1
balance = $0.10
destination= 1 (which is US)
rate = $0.02 per minute
reserveSecs = 10 minutes (600secs)
finalBalance = $0.10 - $0 (consider this is first call and no entry in reserve_balance table) = $0.10
allowedMints = $0.10/$0.02 = 5 minutes = 300 seconds.
AbsoluteTimeout = 300 seconds (this is less than the default reserveSecs so set this as abstimeout)

Call -2
balance = $0.10

destination= 1 (which is US)

rate = $0.02 per minute

reserveSecs = 10 minutes (600secs)

finalBalance = $0.10 - $0.10 (consider this is second call and already an entry in reserve_balance table) = $0.0

allowedMints = 0 seconds.
announce the denied ivr.

So, the reserveSecs is critical to avoid how much threshold amount the
caller should have to make two calls. If they have $10 in their account
as per the above two algorithms, they can make as many simultaneous
calls.

I hope this solves most of your problems. I looked at ASTCC, A2Billing
etc and they are not doing this way and not know whether they work
properly. But this works for me. Shoot me your questions if you have
one.

I am developing my own billing and routing app (in java) and I need a
name for that.. guys pls suggest one.. i may put that in sourceforge if
i feel confident.

Thanks,
Thameem
On 4/27/06, JP Carballo <[EMAIL PROTECTED]> wrote:
Dovid Bender wrote:>A while back some one posted some code that he used>that took out the flag in astcc that kept track if>there was a call in progress for that pin or not. Dont>know if it wil work for real time though.
>>Dovid>>I don't know if you were pertaining to what I posted in the message"ASTCC: How to reset "in-use" flag automatically ?".The setinuse() routine already exists in ASTCC.
One simply has to use that routine to disable the inuse flag when a callbegins and ASTCC will allow multiple calls for the same account.However, I too have no idea if this will work for real time.--
JP Carballohttp://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.___
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RE: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO

2006-05-01 Thread Chris Bagnall

I've no experience with the A200, so I can't really pass judgement on it. We
have 2 sites with TDM400P cards currently deployed (one with 3 FXO, the
other with 1 FXO and 3 FXS). The site in Northampton with 3 FXO has been an
absolute nightmare over the last 9 months the system's been in place.
Initially we had big problems with echo that no amount of tweaking of the
echo parameters would fix.

Once asterisk 1.2 was deployed, things improved remarkably. I took the time
on-site to properly configure the txgain and rxgain using a milliwatt tone
coming from one of our other servers in a datacentre (rather than relying on
using ztmonitor -v whilst the client was making/receiving calls - you can't
guarantee whether the party they're talking to has *their* gains properly
configured, which means you end up with completely different results on
every call). Once I'd got the gains properly configured, fairly standard
echo cancellation settings of "on" and "echotraining=800" seem to work
relatively well.

One thing worth checking with BT - if you can find someone who can give you
an accurate answer - is whether the Featureline will give you disconnect
supervision. If it will, so much the better, as * 1.2 seems to have usable
support for it. If it works reliably, you can increase the number of busy
tones to detect before a channel is killed (this will massively reduce
incidences of false hangups).

It's still not as good as the building next door where they have 2 ISDN
BRIs, but it works, and I've not had any complaints for the last month since
their box was upgraded to 1.2.

Hope that helps.

(shameless plug - I have a spare TDM400 card here if you're looking to
acquire one at a reduced price - discuss off-list if interested)

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] Polycom SoundPoint 501 + Asterisk

2006-05-01 Thread Kenneth Shaw

Guys,

I recently purchased some new SoundPoint 501 phones. The new ones that I
purchased had a newer firmware than the 501s we had deployed at the
time. I updated the firmware on the older 501s, and now these phones
have some very odd echoes with Asterisk, etc. I assume this is due to
changes in the config file somehow (I love Polycom!), but was wondering
if anyone else had experienced odd audio problems with the 501s after
upgrading to the latest bootrom and sip images.

Any info would be much appreciated! Thanks!

-- 
Kenneth Shaw
Director of Technology
ExpiTrans, Inc.
129 W. Wilson St., Suite 204
Costa Mesa, CA 92627
tel: 949 278 7288
fax: 949 642 6044
[EMAIL PROTECTED]

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RE: Spam? Re: [Asterisk-Users] CallerID Name problem

Title: RE: Spam? Re: [Asterisk-Users] CallerID Name problem






I'm getting Number but when I look at the CDR database. I do see the name



 -Original Message-
From:   Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 17:10:26 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    Spam? Re: [Asterisk-Users] CallerID Name problem

Do you get caller ID number?  If so, WAITing is not going to help, since you
already get the info.  If you get caller ID number, then your telco is not
sending the name.

On 5/1/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:
>
>  Do you wait before or after the answer? Do you even need the answer?
>
>
>
>
>  -Original Message-
> From:   Alexander Lopez [mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> ]
> Sent:   Mon May 01 14:26:49 2006
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:    RE: [Asterisk-Users] CallerID Name problem
>
> How are the calls coming into the PBX. PRI? If so add a Wait(1) before
> your try ringing the SIP channel.
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> > Sent: Monday, May 01, 2006 12:37 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] CallerID Name problem
> >
> >
> > I'm having trouble getting callerid name to show up on my phones
> (Cisco
> > 7960 and a few softphones)
> > When I look in the CDR database I see the name but not on any phone
> when
> > being called.
> >
> > I'm running
> > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
> >
> >
> > Any help would be great !
> > ___
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>
>


--
Lacy Moore
Aspendora, Inc.




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Re: [Asterisk-Users] voicemail dialout



Doug Lytle wrote:

> Jon,
> 
> I don't know.  I went into my voicemail.conf and put a semicolon in
> front of that option,
> 
> Re-attached to the Asterisk console and did a reload and the option was
> no longer available from Advanced Options.
> 
> I'm running 1.2.7.1

Do you mean the option was not offered to you or it did not work when
offered. I am also on 1.2.7.1


-- 
Jon Farmer
Telford, Shropshire, UK
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RE: [Asterisk-Users] CallerID Name problem

Title: RE: [Asterisk-Users] CallerID Name problem






Thanks will try that tonight.

Thanks again



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 17:07:43 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users] CallerID Name problem

You don't need the answer, But you need the wait. CallerID Name comes
over the FACILITY messge many times and it takes a slpit second for it
to come in.





    From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Hall, Eric
M.
    Sent: Monday, May 01, 2006 4:34 PM
    To: Asterisk Users Mailing List - Non-Commercial Discussion
    Subject: RE: [Asterisk-Users] CallerID Name problem
   
   

    Do you wait before or after the answer? Do you even need the
answer?
   
   
   
 -Original Message-
    From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
    Sent:   Mon May 01 14:26:49 2006
    To: Asterisk Users Mailing List - Non-Commercial Discussion
    Subject:    RE: [Asterisk-Users] CallerID Name problem
   
    How are the calls coming into the PBX. PRI? If so add a Wait(1)
before
    your try ringing the SIP channel.
   
   
    > -Original Message-
    > From: [EMAIL PROTECTED]
[mailto:asterisk-users-
    > [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
    > Sent: Monday, May 01, 2006 12:37 PM
    > To: Asterisk Users Mailing List - Non-Commercial Discussion
    > Subject: [Asterisk-Users] CallerID Name problem
    >
    >
    > I'm having trouble getting callerid name to show up on my
phones
    (Cisco
    > 7960 and a few softphones)
    > When I look in the CDR database I see the name but not on any
phone
    when
    > being called.
    >
    > I'm running
    > Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
    >
    >
    > Any help would be great !
    > ___
    > --Bandwidth and Colocation provided by Easynews.com --
    >
    > Asterisk-Users mailing list
    > To UNSUBSCRIBE or update options visit:
    >    http://lists.digium.com/mailman/listinfo/asterisk-users
    ___
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    To UNSUBSCRIBE or update options visit:
       http://lists.digium.com/mailman/listinfo/asterisk-users
   





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Re: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO

2006-05-01 Thread John Novack

Though many will probably disagree, you will be LOTS better off with the 
Sangoma A200
It is MUCH more forgiving regarding Motherboards and the PCI 2.2 
requirement, even though the software installation instructions leave 
much to be desired for the inexperienced, it does work, and Sangoma  
fixed a problem I discovered with their FXS module software.
My first experience with Digium and the TDM400 was, when it couldn't be 
seen in a motherboard that said it was PCI 2.2,  Digium's response was 
"try another motherboard"


John Novack


Nick Chalk wrote:


Evening all.

I'm looking at building an Asterisk system for one
of the projects of the Charity where I'm the
SysAdmin.

The project has two analogue phone lines - BT
Featureline Compact, we're in the UK - that I'd
like Asterisk to handle.

My current quandary is which FXO interface to use.
I've been looking at the Digium TDM400P and
Sangoma A200, which are similar prices for two FXO
lines. Can anyone comment on the relative merits
of those two devices? Are there any other options
in the same price range?

As we're a small Charity, price is a major
consideration. I don't think we can afford the
A200's hardware echo-canceller, for instance.

Thanks for your help.

Nick.

 


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[Asterisk-Users] app_icd

2006-05-01 Thread lenz


Hello list,

I am seeing that the current version of AMP / FreePBX sports app_icd
instead of app_queue as the default call distribution method - see
http://sourceforge.net/projects/icd

I was wondering if anyone had hands-on experience with app_icd and if
there are good selling points that make the transition worthwile - it is
not surely used in as many installations as app_queue, so I would bet it
is not as tested.

Anybody got first-hand experience?
Yours,
l.

--
Assum est, versa et manduca.
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Re: [Asterisk-Users] CallerID Name problem

2006-05-01 Thread Lacy Moore - Aspendora

Do you get caller ID number?  If so, WAITing is not going to help, since you already get the info.  If you get caller ID number, then your telco is not sending the name.
On 5/1/06, Hall, Eric M. <[EMAIL PROTECTED]> wrote:


Do you wait before or after the answer? Do you even need the answer?
 -Original Message-From:   Alexander Lopez [
mailto:[EMAIL PROTECTED]]Sent:   Mon May 01 14:26:49 2006To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users] CallerID Name problemHow are the calls coming into the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP channel.
> -Original Message-> From: [EMAIL PROTECTED]
 [mailto:asterisk-users-> 
[EMAIL PROTECTED]] On Behalf Of Hall, Eric M.> Sent: Monday, May 01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: [Asterisk-Users] CallerID Name problem
>>> I'm having trouble getting callerid name to show up on my phones(Cisco> 7960 and a few softphones)> When I look in the CDR database I see the name but not on any phonewhen
> being called.>> I'm running> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC>>> Any help would be great !> ___
> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users___
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 --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. 
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RE: [Asterisk-Users] CallerID Name problem

Title: RE: [Asterisk-Users] CallerID Name problem



You don't need the answer, But you need the wait. 
CallerID Name comes over the FACILITY messge many times and it takes a slpit 
second for it to come in.
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Hall, 
  Eric M.Sent: Monday, May 01, 2006 4:34 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] CallerID Name problem
  
  Do you wait before or after the answer? Do you even need the 
  answer? -Original Message-From:   
  Alexander Lopez [mailto:[EMAIL PROTECTED]]Sent:   
  Mon May 01 14:26:49 2006To: Asterisk Users Mailing 
  List - Non-Commercial 
  DiscussionSubject:    RE: 
  [Asterisk-Users] CallerID Name problemHow are the calls coming into 
  the PBX. PRI? If so add a Wait(1) beforeyour try ringing the SIP 
  channel.> -Original Message-> From: 
  [EMAIL PROTECTED] [mailto:asterisk-users-> 
  [EMAIL PROTECTED] On Behalf Of Hall, Eric M.> Sent: Monday, May 
  01, 2006 12:37 PM> To: Asterisk Users Mailing List - Non-Commercial 
  Discussion> Subject: [Asterisk-Users] CallerID Name 
  problem>>> I'm having trouble getting callerid name to 
  show up on my phones(Cisco> 7960 and a few softphones)> When 
  I look in the CDR database I see the name but not on any phonewhen> 
  being called.>> I'm running> Asterisk SVN-trunk-r7498 
  built on 2006-04-30 15:11:39 UTC>>> Any help would be 
  great !> ___> 
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[Asterisk-Users] Asterisk-Users Digest, Vol 22, Issue 1

2006-05-01 Thread Terence Burnard

Hey,

Thanks for the input Andrew. I did all you suggested but noticed that 
when I did the loopback test, the output *was not* there as you 
mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!").

In fact, the same message as before kept repeating every second or so:

>> Unnumbered frame:
>> SAPI: 00  C/R: 0 EA: 0
>>  TEI: 000EA: 1
>>   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode

extended) ]

>> 0 bytes of data

Sending Set Asynchronous Balanced Mode Extended

This was true on both of our Te411 cards so I'd be surprised if both 
were defective.  The cisco gateway that this Asterisk box is replacing 
is dms100, esf, b8zs so I just set the asterisk box with the same 
configuration.  I spoke to the telco and they said everything looks good 
on their end, and since this a production T1 it doesn't need turning up.

Now that I know support comes with these cards, I might give Digium a 
shout. :)

Terence

Date: Mon, 1 May 2006 07:20:51 -0400
From: Andrew Kohlsmith <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] PRI Issue: D-Channel woes
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;  charset="iso-8859-1"

On Monday 01 May 2006 01:42, Terence Burnard wrote:

Module  Size  Used by
wcusb  21760  0
wctdm  36512  0
wcfxo  13408  0
wcte11xp   24896  0
wct1xxp16544  0
wct4xxp97664  24
tor2   89856  0

First of all, don't load every Asterisk module under the sun.  Load the
modules for the hardware you have, and if you're using something like [EMAIL 
PROTECTED]
which loads everything, edit your /etc/modules.conf to alias the ones 
you do

NOT have to 'off' to prevent them from being loaded.

# cat /etc/zaptel.conf
span=1,0,0,esf,b8zs

That should read '1,1,0' but is otherwise great.  You want to 
synchronize to

the telco.

[channels]
context=pri_inbound
switchtype=dms100
signalling=pri_cpe
group=1
channel => 1-23

I generally add
pridialplan=unknown
priindication=outofband
overlapdial=no
resetinterval=86400

to that as well (i.e. before the "channel =>" line) to make things a little
cleaner and clearer.  Only the first line would have any effect on 
mitigating

your problem, though.

*CLI> pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active

That's the problem: your D channel is not up.  Either your switchtype is 
wrong
(it is NOT set to the physical switch make/model on the other side, it 
is set
to the signaling system it is emulating), the telco's brought the line 
down,

or the two sides just can't see each other.  Since you're running
green-light, you're seeing each other.

You can do a very simple test by plugging a loopback plug into the span
instead of the PRI.  just take an RJ45 end and two pieces of wire.  plug 
one

wire into pins 1 and 4, and the other into 2 and 5.  Crimp it down and plug
it in.  The light on the back of the card should go green and the system
should be in the exact same state it's in now.  If you change your 
signalling
to pri_net, you should see messages on the Asterisk console complaining 
about

"I'm set to pri_net, but the other side thinks it is pri_net!".

Verify your switchtype...  The Cisco should have a setting and you 
should set
Asterisk in a similar fashion.  Failing that I would try loading the 
wct4xxp

module with the "vpmsupport=0" parameter to disable the voice processing
module just to see if that helps (i.e. possible card issue), and failing 
all
of that, I would contact Digium technical support, as the price of that 
card
included technical assistance and it is this type of problem that is a 
great

use of that support.

-A.
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Re: [Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO

2006-05-01 Thread Mike Clark


Nick Chalk wrote:


Evening all.

I'm looking at building an Asterisk system for one
of the projects of the Charity where I'm the
SysAdmin.

The project has two analogue phone lines - BT
Featureline Compact, we're in the UK - that I'd
like Asterisk to handle.

My current quandary is which FXO interface to use.
I've been looking at the Digium TDM400P and
Sangoma A200, which are similar prices for two FXO
lines. Can anyone comment on the relative merits
of those two devices? Are there any other options
in the same price range?

As we're a small Charity, price is a major
consideration. I don't think we can afford the
A200's hardware echo-canceller, for instance.

Thanks for your help.

Nick.

 


Nick:

If you aren't going for the echo cancellation, then I think either card 
will do fine.We have over twenty five systems deployed in rural eastern 
North Carolina locations with six analog lines each and the majority 
work fine with TDM400s, though we have had to replace several with 
Sangoma A200 with echo cancellation to eliminate echo. We are now 
deploying only the A200 because we never know if echo will be an issue 
or if it can be "tuned" away and it is just easier for support to not 
have that worry.


Mike Clark


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RE: [Asterisk-Users] CallerID Name problem

Title: RE: [Asterisk-Users] CallerID Name problem






Do you wait before or after the answer? Do you even need the answer?



 -Original Message-
From:   Alexander Lopez [mailto:[EMAIL PROTECTED]]
Sent:   Mon May 01 14:26:49 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:    RE: [Asterisk-Users] CallerID Name problem

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Name problem
>
>
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
>
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
>
>
> Any help would be great !
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Re: [Asterisk-Users] integrated voip originator, to digitize audio once and only once?

2006-05-01 Thread Tom Engleward

Bruce Reeves <[EMAIL PROTECTED]> wrote:
> I use teliax.com and exgn.net to do my initial test
> of toll free calls into
> my system.
How's your experience been with their audio quality,
and with their inbound call completion reliability?


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Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk

2006-05-01 Thread Matt Roth


Steve Totaro wrote:

Not sure if cheesy is the right word.  Sound solution may be a better 
adjective.  Adding two NICs, one to each machine and connecting them 
directly via crossover cable on a totally separate network may be my 
best solution.  No FTP traffic would even hit the NIC or the network 
used for VoIP and everything else.


Unless there is a setting in Linux somewhere (still holding out hope)


Steve,

Part of your problem may be that you are mixing batches of leg files on 
the Asterisk server every five minutes.  Mixing the leg files is 
processor intensive and I'd be surprised if mixing them in batches 
didn't degrade your call quality.  In general, it is better to perform 
tasks such as this on a remote server.


If the problem is purely bandwidth related, the batch processing is 
probably aggravating that as well.  It is relatively simple to transfer 
the leg files as each individual call terminates.  Staggering the 
transfers of the files may provide all of the throttling you need.


I am currently administrating an Asterisk server that functions as the 
switch for an inbound call center.  On a busy day it handles over 13,000 
digital recordings in the PCM format.  This post caught my eye, because 
I use a dedicated NIC and a crossover cable to transfer them via NFS.  
From my observations, this method adds very little load to the server 
(although I am considering NAPI and other methods to throttle the number 
of interrupts generated by the NICs).  The entire setup is documented here:




I've made some small alterations in our production environment since the 
writing of that post.  The most important one concerns the use of the 
"MONITOR_EXEC" variables in the dialplan to trigger the transferring of 
the leg files.  That method turned out to be unreliable.  Hijacking 
soxmix, as documented in the following post, handles this task reliably:




See "show application monitor" in the Asterisk CLI for details on 
getting your custom "soxmix" script invoked at the end of each call.


As I mentioned, offloading tasks such as mixing the leg files is a *very 
good idea*.  I have a set of four scripts on the digital recording 
server that take care of mixing the leg files, indexing them by date and 
time, exporting a list of recordings to our mainframe, archiving the 
recordings, and deleting old recordings to free up disk space.  I'd be 
happy to help you offload some of these tasks from your Asterisk server 
if you are interested.


I hope this was helpful

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[Asterisk-Users] Digium TDM400P vs Sangoma A200 for 2 x FXO

2006-05-01 Thread Nick Chalk

Evening all.

I'm looking at building an Asterisk system for one
of the projects of the Charity where I'm the
SysAdmin.

The project has two analogue phone lines - BT
Featureline Compact, we're in the UK - that I'd
like Asterisk to handle.

My current quandary is which FXO interface to use.
I've been looking at the Digium TDM400P and
Sangoma A200, which are similar prices for two FXO
lines. Can anyone comment on the relative merits
of those two devices? Are there any other options
in the same price range?

As we're a small Charity, price is a major
consideration. I don't think we can afford the
A200's hardware echo-canceller, for instance.

Thanks for your help.

Nick.

-- 
Nick Chalk . once a Radio Designer
 Confidence is failing to understand the problem.

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RE: [Asterisk-Users] FreePBX in production?

2006-05-01 Thread Remco Barende

On Mon, 1 May 2006, Kerry Garrison wrote:

> There are still some basic things missing (for example if you
don't use
> voicemail it is not possible to set a destination for the call if
not
> answered, you have to create a ring group for each extension to
work
> around it, this is a major issue)

Remco - take a look at the Follow Me module I added. It is basically
a presonal ring group for each extension. If you want to do the above, just
define the Follow-Me settings to ring your own extension (or more if you
want) and then choose any destination you want. It effectively does 'creat a
ring group for each extensions' that wants one, but it does it in such a way
as to be separate and work side by side with normal ringgroups, and there is
a direct link between it and the extension (or user) so that navigation is
very easy as you can bounce back and forth with a single mouse click.

Sounds very interesting, and looks like a good solution to the problem.

I believe that many of the "shortcomings" of FreePBX are caused by lack of 
documentation, i.e. you do not immediately see how to solve a certain 
problem.

Thanks to all the develeopers for all the work they are putting 
in FreePBX!

I'll have a look at the new functions, thanks!
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RE: [Asterisk-Users] Problems with zaptel and TE210P









Did your Telco tell you what switch they
are using?

 

Also change your timing settings in the /etc/zaptel.conf
file to have this PRI do primary timing.

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 2:59 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P



 

Thank you for the reply Alexandar.

 

After I restarted my dev machine I
recieved these messages from asterisk:

 

chan_zap.c:2290 pri_find_dchan: No
D-channels available!  Using Primary channel 24 as D-channel anyway!



chan_zap.c:8202 pri_dchannel: PRI got
event: No more alarm (5) on Primary D-channel of span 1



May  1 10:23:27 WARNING[3882]:
chan_zap.c:2290 pri_find_dchan: No D-channels available!  Using Primary
channel 24 as D-channel anyway!

 

So you are right on the D-Chan.  If
my telco says everything is ok, what should I look at next?

 

AFAIK this PRI was in working condition
before I moved it to the asterisk test machine.

 

Thanks!

 

-Dan

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 11:53
AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P

Looks like your D-channel is down.

 

Ztcfg reports all is ok, b/c as far as iut
is concerned, it is talking to your card just fine. LibPri handles the PRI
implemetaton.

 

Since you are able to see the pri commands
from the CLI, Isdn supprt is linked into your asterisk core.

 

Call your telco and ask if they have your
D-channel in a loop.

 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 2:48 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P



 

Some more info to my problem:

 

ipt-dev01*CLI> zap show status

Description 
Alarms IRQ   
bpviol CRC4

T2XXP (PCI) Card 0 Span
1   
OK
0 
0  0



 



ipt-dev01*CLI> pri show span 1

Primary D-channel: 24

Status: Provisioned, Down, Active

Switchtype: National ISDN

Type: CPE

Window Length: 0/7

Sentrej: 0

SolicitFbit: 0

Retrans: 0

Busy: 0

Overlap Dial: 0

T200 Timer: 1000

T203 Timer: 1

T305 Timer: 3

T308 Timer: 4000

T313 Timer: 4000

N200 Counter: 3



 



Any ideas? ZTCFG looks good.

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 10:41
AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problems
with zaptel and TE210P



Hello,





I'm just starting out with asterisk and I'm playing around
with the system.  Currently I have a Digium TE210P connected to a PRI on
the Asterisk server.  I have a SIP soft phone on my laptop for testing
that is working fine.  When I try to place a call from my soft phone I get
this from Asterisk:





 





May  1 09:11:41 NOTICE[20098]: app_dial.c:1029
dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel
congestion)





  == Everyone is busy/congested at this time (1:0/1/0)





  == Auto fallthrough, channel 'SIP/test-3a26' status
is 'CONGESTION'





 





 





#/etc/zaptel.conf:





span=1,0,0,esf,b8zs





bchan=1-23





dchan=24





 





#/etc/asterisk/zapata.conf:





[channels]





switchtype=national





context=default





signalling=pri_cpe





group=1





channel => 1-23





 





#/etc/asterisk/extensions.conf:





[general]





static=yes





writeprotect=no





autofallthrough=yes





 





[default]





exten => 123,1,Answer()





exten => 123,2,Playback(hello-world)





exten => 123,3,Hangup()





 





exten => _9NXX,1,Dial(Zap/g1)





 





 





Any ideas?  Thank you in advance, your help is greatly
appreciated.





 





-Dan





 





 












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RE: [Asterisk-Users] Problems with zaptel and TE210P









His PRI span is showing down, If you
forget to add the ${EXTEN} as you said it would show as connecting and he _should_ get an intercept from the telco.

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Monday, May 01, 2006 2:53 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P



 



Just a stab:





 





exten =>
_9NXX,1,Dial(ZAP/g1/${EXTEN})





 





Note uppercase ZAP and explicitly
specifying the dialled number. 





 





hth





-Original Message-
From: Dan Brummer
[mailto:[EMAIL PROTECTED]
Sent: Monday, May 01, 2006 12:48 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P

Some more info to my problem:

 

ipt-dev01*CLI> zap show status

Description 
Alarms IRQ    bpviol
CRC4

T2XXP (PCI) Card 0 Span
1   
OK
0 
0  0



 



ipt-dev01*CLI> pri show span 1

Primary D-channel: 24

Status: Provisioned, Down, Active

Switchtype: National ISDN

Type: CPE

Window Length: 0/7

Sentrej: 0

SolicitFbit: 0

Retrans: 0

Busy: 0

Overlap Dial: 0

T200 Timer: 1000

T203 Timer: 1

T305 Timer: 3

T308 Timer: 4000

T313 Timer: 4000

N200 Counter: 3



 



Any ideas? ZTCFG looks good.

 







From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 10:41
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problems
with zaptel and TE210P



Hello,





I'm just starting out with asterisk and I'm playing around
with the system.  Currently I have a Digium TE210P connected to a PRI on
the Asterisk server.  I have a SIP soft phone on my laptop for testing
that is working fine.  When I try to place a call from my soft phone I get
this from Asterisk:





 





May  1 09:11:41 NOTICE[20098]: app_dial.c:1029
dial_exec_full: Unable to create channel of type 'Zap' (cause 34 -
Circuit/channel congestion)





  == Everyone is busy/congested at this time (1:0/1/0)





  == Auto fallthrough, channel 'SIP/test-3a26' status
is 'CONGESTION'





 





 





#/etc/zaptel.conf:





span=1,0,0,esf,b8zs





bchan=1-23





dchan=24





 





#/etc/asterisk/zapata.conf:





[channels]





switchtype=national





context=default





signalling=pri_cpe





group=1





channel => 1-23





 





#/etc/asterisk/extensions.conf:





[general]





static=yes





writeprotect=no





autofallthrough=yes





 





[default]





exten => 123,1,Answer()





exten => 123,2,Playback(hello-world)





exten => 123,3,Hangup()





 





exten => _9NXX,1,Dial(Zap/g1)





 





 





Any ideas?  Thank you in advance, your help is greatly
appreciated.





 





-Dan





 





 












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Re: [Asterisk-Users] Is there a way to monitor DTMF tones in a channel?

2006-05-01 Thread C F


On 5/1/06, Obelix <[EMAIL PROTECTED]> wrote:


Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?

I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc


take a look at features.conf
what do you mean by "arbitrary", beyond the 16 DTMF being used? if so
you will have to edit the code for the others and it should work.




/Obelix
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Re: [Asterisk-Users] Problems with zaptel and TE210P

2006-05-01 Thread James Texter

Title: Re: [Asterisk-Users] Problems with zaptel and TE210P



Shouldn’t your zapata.conf be 

span=1,1,0,esf,b8zs

As it stands, you are not taking timing from the PRI.  Changing the second digit of the span entry to 1 will tell Asterisk to use that line as the clock master.

HTH,

James


On 5/1/06 1:58 PM, "Dan Brummer" <[EMAIL PROTECTED]> wrote:

Thank you for the reply Alexandar.
 
After I restarted my dev machine I recieved these messages from asterisk:
 
chan_zap.c:2290 pri_find_dchan: No D-channels available!  Using Primary channel 24 as D-channel anyway!
chan_zap.c:8202 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1
May  1 10:23:27 WARNING[3882]: chan_zap.c:2290 pri_find_dchan: No D-channels available!  Using Primary channel 24 as D-channel anyway!
 
So you are right on the D-Chan.  If my telco says everything is ok, what should I look at next?
 
AFAIK this PRI was in working condition before I moved it to the asterisk test machine.
 
Thanks!
 
-Dan

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Monday, May 01, 2006 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P

Looks like your D-channel is down.
 
Ztcfg reports all is ok, b/c as far as iut is concerned, it is talking to your card just fine. LibPri handles the PRI implemetaton.
 
Since you are able to see the pri commands from the CLI, Isdn supprt is linked into your asterisk core.
 
Call your telco and ask if they have your D-channel in a loop.
 
 
 





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Problems with zaptel and TE210P
 
Some more info to my problem:
 
ipt-dev01*CLI> zap show status
Description  Alarms IRQ    bpviol CRC4
T2XXP (PCI) Card 0 Span 1    OK 0  0  0

 
ipt-dev01*CLI> pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3

 
Any ideas? ZTCFG looks good.
 





From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 10:41 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problems with zaptel and TE210P

Hello,

I'm just starting out with asterisk and I'm playing around with the system.  Currently I have a Digium TE210P connected to a PRI on the Asterisk server.  I have a SIP soft phone on my laptop for testing that is working fine.  When I try to place a call from my soft phone I get this from Asterisk:

 

May  1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)

  == Everyone is busy/congested at this time (1:0/1/0)

  == Auto fallthrough, channel 'SIP/test-3a26' status is 'CONGESTION'

 

 

#/etc/zaptel.conf:

span=1,0,0,esf,b8zs

bchan=1-23

dchan=24

 

#/etc/asterisk/zapata.conf:

[channels]

switchtype=national

context=default

signalling=pri_cpe

group=1

channel => 1-23

 

#/etc/asterisk/extensions.conf:

[general]

static=yes

writeprotect=no

autofallthrough=yes

 

[default]

exten => 123,1,Answer()

exten => 123,2,Playback(hello-world)

exten => 123,3,Hangup()

 

exten => _9NXX,1,Dial(Zap/g1)

 

 

Any ideas?  Thank you in advance, your help is greatly appreciated.

 

-Dan

 

 

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RE: [Asterisk-Users] Problems with zaptel and TE210P

2006-05-01 Thread Dan Brummer




Thank you for the reply Alexandar.
 
After I restarted my dev machine I recieved these messages 
from asterisk:
 
chan_zap.c:2290 pri_find_dchan: No D-channels 
available!  Using Primary channel 24 as D-channel 
anyway!
chan_zap.c:8202 pri_dchannel: PRI got event: No more alarm (5) on Primary 
D-channel of span 1
May  1 10:23:27 WARNING[3882]: chan_zap.c:2290 
pri_find_dchan: No D-channels available!  Using Primary channel 24 as 
D-channel anyway!
 
So you are right on the D-Chan.  If my telco says 
everything is ok, what should I look at next?
 
AFAIK this PRI was in working condition before I moved it 
to the asterisk test machine.
 
Thanks!
 
-Dan


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander 
LopezSent: Monday, May 01, 2006 11:53 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
Problems with zaptel and TE210P


Looks like your 
D-channel is down.
 
Ztcfg reports all is 
ok, b/c as far as iut is concerned, it is talking to your card just fine. LibPri 
handles the PRI implemetaton.
 
Since you are able to 
see the pri commands from the CLI, Isdn supprt is linked into your asterisk 
core.
 
Call your telco and ask 
if they have your D-channel in a loop.
 
 
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dan BrummerSent: Monday, May 01, 2006 2:48 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] Problems with 
zaptel and TE210P
 
Some more info to my 
problem:
 
ipt-dev01*CLI> zap 
show status
Description  
Alarms IRQ    
bpviol CRC4
T2XXP (PCI) Card 0 Span 
1    
OK 
0  
0  
0

 
ipt-dev01*CLI> pri 
show span 1
Primary D-channel: 
24
Status: 
Provisioned, Down, 
Active
Switchtype: National 
ISDN
Type: 
CPE
Window Length: 
0/7
Sentrej: 
0
SolicitFbit: 
0
Retrans: 
0
Busy: 
0
Overlap Dial: 
0
T200 Timer: 
1000
T203 Timer: 
1
T305 Timer: 
3
T308 Timer: 
4000
T313 Timer: 
4000
N200 Counter: 
3

 
Any ideas? ZTCFG looks 
good.
 



From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dan BrummerSent: Monday, May 01, 2006 10:41 
AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problems with 
zaptel and TE210P

Hello,

I'm just starting out with asterisk 
and I'm playing around with the system.  Currently I have a Digium TE210P 
connected to a PRI on the Asterisk server.  I have a SIP soft phone on my 
laptop for testing that is working fine.  When I try to place a call from 
my soft phone I get this from Asterisk:

 

May  1 09:11:41 NOTICE[20098]: 
app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 
- Circuit/channel congestion)

  == Everyone is busy/congested 
at this time (1:0/1/0)

  == Auto fallthrough, channel 
'SIP/test-3a26' status is 'CONGESTION'

 

 

#/etc/zaptel.conf:

span=1,0,0,esf,b8zs

bchan=1-23

dchan=24

 

#/etc/asterisk/zapata.conf:

[channels]

switchtype=national

context=default

signalling=pri_cpe

group=1

channel => 
1-23

 

#/etc/asterisk/extensions.conf:

[general]

static=yes

writeprotect=no

autofallthrough=yes

 

[default]

exten => 
123,1,Answer()

exten => 
123,2,Playback(hello-world)

exten => 
123,3,Hangup()

 

exten => 
_9NXX,1,Dial(Zap/g1)

 

 

Any ideas?  Thank you in 
advance, your help is greatly appreciated.

 

-Dan

 

 
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RE: [Asterisk-Users] Problems with zaptel and TE210P

2006-05-01 Thread Colin Anderson




Just a 
stab:
 
exten 
=> _9NXX,1,Dial(ZAP/g1/${EXTEN})
 
Note 
uppercase ZAP and explicitly specifying the dialled number. 
 
hth

  -Original Message-From: Dan Brummer 
  [mailto:[EMAIL PROTECTED]Sent: Monday, May 01, 2006 12:48 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: RE: [Asterisk-Users] Problems with zaptel and 
  TE210P
  Some more info to my problem:
   
  ipt-dev01*CLI> zap show status
  Description  
  Alarms IRQ    
  bpviol CRC4
  T2XXP (PCI) Card 0 Span 
  1    
  OK 
  0  
  0  0
   
  ipt-dev01*CLI> pri show span 1
  Primary D-channel: 24
  Status: Provisioned, Down, 
  Active
  Switchtype: National ISDN
  Type: CPE
  Window Length: 0/7
  Sentrej: 0
  SolicitFbit: 0
  Retrans: 0
  Busy: 0
  Overlap Dial: 0
  T200 Timer: 1000
  T203 Timer: 1
  T305 Timer: 3
  T308 Timer: 4000
  T313 Timer: 4000
  N200 Counter: 3
   
  Any ideas? ZTCFG looks good.
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dan 
  BrummerSent: Monday, May 01, 2006 10:41 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problems 
  with zaptel and TE210P
  
  Hello,
  I'm just starting 
  out with asterisk and I'm playing around with the system.  Currently I 
  have a Digium TE210P connected to a PRI on the Asterisk server.  I have a 
  SIP soft phone on my laptop for testing that is working fine.  When I try 
  to place a call from my soft phone I get this from 
  Asterisk:
   
  May  1 
  09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create 
  channel of type 'Zap' (cause 34 - Circuit/channel 
  congestion)
    == Everyone 
  is busy/congested at this time (1:0/1/0)
    == Auto 
  fallthrough, channel 'SIP/test-3a26' status is 
'CONGESTION'
   
   
  #/etc/zaptel.conf:
  span=1,0,0,esf,b8zs
  bchan=1-23
  dchan=24
   
  #/etc/asterisk/zapata.conf:
  [channels]
  switchtype=national
  context=default
  signalling=pri_cpe
  group=1
  channel => 
  1-23
   
  #/etc/asterisk/extensions.conf:
  [general]
  static=yes
  writeprotect=no
  autofallthrough=yes
   
  [default]
  exten => 
  123,1,Answer()
  exten => 
  123,2,Playback(hello-world)
  exten => 
  123,3,Hangup()
   
  exten => 
  _9NXX,1,Dial(Zap/g1)
   
   
  Any ideas?  
  Thank you in advance, your help is greatly appreciated.
   
  -Dan
   
   
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RE: [Asterisk-Users] Problems with zaptel and TE210P









Looks like your D-channel is down.

 

Ztcfg reports all is ok, b/c as far as iut
is concerned, it is talking to your card just fine. LibPri handles the PRI
implemetaton.

 

Since you are able to see the pri commands
from the CLI, Isdn supprt is linked into your asterisk core.

 

Call your telco and ask if they have your
D-channel in a loop.

 

 

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 2:48 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Problems with zaptel and TE210P



 

Some more info to my problem:

 

ipt-dev01*CLI> zap show status

Description 
Alarms IRQ   
bpviol CRC4

T2XXP (PCI) Card 0 Span
1   
OK
0 
0  0



 



ipt-dev01*CLI> pri show span 1

Primary D-channel: 24

Status: Provisioned, Down, Active

Switchtype: National ISDN

Type: CPE

Window Length: 0/7

Sentrej: 0

SolicitFbit: 0

Retrans: 0

Busy: 0

Overlap Dial: 0

T200 Timer: 1000

T203 Timer: 1

T305 Timer: 3

T308 Timer: 4000

T313 Timer: 4000

N200 Counter: 3



 



Any ideas? ZTCFG looks good.

 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Brummer
Sent: Monday, May 01, 2006 10:41
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problems
with zaptel and TE210P



Hello,





I'm just starting out with asterisk and I'm playing around
with the system.  Currently I have a Digium TE210P connected to a PRI on
the Asterisk server.  I have a SIP soft phone on my laptop for testing that
is working fine.  When I try to place a call from my soft phone I get this
from Asterisk:





 





May  1 09:11:41 NOTICE[20098]: app_dial.c:1029
dial_exec_full: Unable to create channel of type 'Zap' (cause 34 -
Circuit/channel congestion)





  == Everyone is busy/congested at this time (1:0/1/0)





  == Auto fallthrough, channel 'SIP/test-3a26' status
is 'CONGESTION'





 





 





#/etc/zaptel.conf:





span=1,0,0,esf,b8zs





bchan=1-23





dchan=24





 





#/etc/asterisk/zapata.conf:





[channels]





switchtype=national





context=default





signalling=pri_cpe





group=1





channel => 1-23





 





#/etc/asterisk/extensions.conf:





[general]





static=yes





writeprotect=no





autofallthrough=yes





 





[default]





exten => 123,1,Answer()





exten => 123,2,Playback(hello-world)





exten => 123,3,Hangup()





 





exten => _9NXX,1,Dial(Zap/g1)





 





 





Any ideas?  Thank you in advance, your help is greatly
appreciated.





 





-Dan





 





 










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RE: [Asterisk-Users] FreePBX in production?

2006-05-01 Thread Kerry Garrison

> There are still some basic things missing (for example if you
don't use 
> voicemail it is not possible to set a destination for the call if
not 
> answered, you have to create a ring group for each extension to
work 
> around it, this is a major issue)

Remco - take a look at the Follow Me module I added. It is basically
a presonal ring group for each extension. If you want to do the above, just
define the Follow-Me settings to ring your own extension (or more if you
want) and then choose any destination you want. It effectively does 'creat a
ring group for each extensions' that wants one, but it does it in such a way
as to be separate and work side by side with normal ringgroups, and there is
a direct link between it and the extension (or user) so that navigation is
very easy as you can bounce back and forth with a single mouse click.
 

Many people have talked about limitations of freePBX and how you cant do
custom things. Both Phillip and I attacked the follow-me function this week
with his using personal ring groups and mine using personal call queues (see
article at http://voipspeak.net) to simulate the functionaly of the locate
function from CallManager. Both solutions used only freePBX functionality to
acocmplish two relativly complex tasks that many people have been struggling
to create with just the config files. That speaks to the flexibility of
using the system. Sure there are bound to be limitations that may prevent
some type of functionality, no system is perfect, but it does cover far more
than just a few small businesses.

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



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RE: [Asterisk-Users] Problems with zaptel and TE210P

2006-05-01 Thread Dan Brummer




Some more info to my problem:
 
ipt-dev01*CLI> zap show status
Description  
Alarms IRQ    
bpviol CRC4
T2XXP (PCI) Card 0 Span 
1    
OK 
0  
0  0
 
ipt-dev01*CLI> pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, 
Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
 
Any ideas? ZTCFG looks good.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dan 
BrummerSent: Monday, May 01, 2006 10:41 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problems 
with zaptel and TE210P

Hello,
I'm just starting 
out with asterisk and I'm playing around with the system.  Currently I have 
a Digium TE210P connected to a PRI on the Asterisk server.  I have a SIP 
soft phone on my laptop for testing that is working fine.  When I try to 
place a call from my soft phone I get this from Asterisk:
 
May  1 09:11:41 
NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 
'Zap' (cause 34 - Circuit/channel congestion)
  == Everyone 
is busy/congested at this time (1:0/1/0)
  == Auto 
fallthrough, channel 'SIP/test-3a26' status is 'CONGESTION'
 
 
#/etc/zaptel.conf:
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
 
#/etc/asterisk/zapata.conf:
[channels]
switchtype=national
context=default
signalling=pri_cpe
group=1
channel => 
1-23
 
#/etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
 
[default]
exten => 
123,1,Answer()
exten => 
123,2,Playback(hello-world)
exten => 
123,3,Hangup()
 
exten => 
_9NXX,1,Dial(Zap/g1)
 
 
Any ideas?  
Thank you in advance, your help is greatly appreciated.
 
-Dan
 
 
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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Doug Lytle wrote:

  

What do you see at the console when someone presses 4 from voice mail?



-- Executing VoiceMailMain("SIP/502-ac3f", "s502") in new stack
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-tomakecall' (language 'en')
-- Playing 'vm-starmain' (language 'en')


  

Jon,

I don't know.  I went into my voicemail.conf and put a semicolon in 
front of that option,


Re-attached to the Asterisk console and did a reload and the option was 
no longer available from Advanced Options.


I'm running 1.2.7.1

Doug

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Re: [Asterisk-Users] FreePBX in production?

2006-05-01 Thread Philippe Lindheimer

Rich Adamson wrote: > Let's see if I can summarize various recent postings relative to the > broader topic of whether FreePBX/AAH is production-ready. It's not proper to put FreePBX/AAH in the same breath. AAH puts FreePBX ontop of their build, along with a bunch of other software. Although AAH gets 'most of the credit,' the 'value' that most users are exposed to is FreePBX (or AMP). Not to trivialize the 'integration' work that AAH does. However - I say don't put them together because all too often I have seen AAH drop things on top that break FreePBX. If you want a pure system, build your own ISO and drop FreePBX on top of it. Rich Adamson wrote: > Seems the general consensus is that AAH and/or FreePBX is considered > production ready if the functionality embedded in AMP (primarily) > happens to fit the specific small
business requirements... and "Craig" wrote: > Too many limitations in terms of having a flexible diaplan. What would be nice> though is if they were to produce a 'lite' version that gave a gui interface> to add/change/move things - sip.conf, voicemail.conf, meetme.conf but> staying well away from extensions.conf What one considers 'prodcution ready' is a very subjective evaluation. However, I will say that I have yet to find something I can't do on a system that FreePBX has. What I mean by that is that I can modify any macro or part of the core dial plan I don't like by overriding it in the _custom file as well as add any custom dialplans or other configuration that I need. I'm sure there is something out there I will run into where this doesn't work - but I haven't hit it yet. (And if I do, I'll do the needed changes and submit it back to FreePBX).
I think it is fair to say that if you really understand Asterisk and then spend a small amount of time understanding what FreePBX is doing, you can easily accomplish the best of both worlds. If you don't have that level of understanding, you may be thankful for those 'magic scripts' that are making a working system behind the scene from the FreePBX GUI. If you are not interested in the fundamental functionality and 'fat/rich' dailplan, then you are better off using something else though. Remco Barende wrote: > There are still some basic things missing (for example if you don't use > voicemail it is not possible to set a destination for the call if not > answered, you have to create a ring group for each extension to work > around it, this is a major issue) Remco - take a look at the Follow Me module I added. It is
basically a presonal ring group for each extension. If you want to do the above, just define the Follow-Me settings to ring your own extension (or more if you want) and then choose any destination you want. It effectively does 'creat a ring group for each extensions' that wants one, but it does it in such a way as to be separate and work side by side with normal ringgroups, and there is a direct link between it and the extension (or user) so that navigation is very easy as you can bounce back and forth with a single mouse click. p
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RE: [Asterisk-Users] CallerID Name problem

How are the calls coming into the PBX. PRI? If so add a Wait(1) before
your try ringing the SIP channel.


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Hall, Eric M.
> Sent: Monday, May 01, 2006 12:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Name problem
> 
> 
> I'm having trouble getting callerid name to show up on my phones
(Cisco
> 7960 and a few softphones)
> When I look in the CDR database I see the name but not on any phone
when
> being called.
> 
> I'm running
> Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC
> 
> 
> Any help would be great !
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RE: [Asterisk-Users] CallerID Name problem

Using SIP and SCCP. The softphone uses SIP.

Doing a debug  I see no name being sent. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kevin ling
Sent: Monday, May 01, 2006 2:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CallerID Name problem

Hi,

What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone. 

Kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric
M.
Sent: Tuesday, May 02, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CallerID Name problem

I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones) When I look in the CDR database I see the
name but not on any phone when being called.

I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC 

Any help would be great !

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RE: [Asterisk-Users] CallerID Name problem

2006-05-01 Thread kevin ling

Hi,

What protocol for your 7960 phone? SCCP or SIP? You can turn on the SIP
debug on CLI to make sure the callerid and name pass to your phone. 

Kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Tuesday, May 02, 2006 12:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CallerID Name problem

 
I'm having trouble getting callerid name to show up on my phones (Cisco 7960
and a few softphones) When I look in the CDR database I see the name but not
on any phone when being called.

I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC 


Any help would be great !



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Re: [Asterisk-Users] Listening on one IP and binding to other IP - isthis possible ?

2006-05-01 Thread Bartosz Jozwiak





Hello

Im new to Asterisk and generally to IPPBX.

Quick specification of my SOHO system:
1. 2 Interfaces: WAN and LAN
4. 5 VoIP phones configured with SIP attached to LAN
5. 1 external SIP line configured as "gateway" for VoIP phones to allow
outbound calls.

What I want is to set secure Asterisk platform. I would like Asterisk to
allow connections only from LAN but I would also like to use 1 SIP
account which requires registration on my provider sip server which
needs Internet connection.

When I set sip.conf "bindaddr" option to my LAN addres I was sure that
Asterisk will use WAN interface for Internet connections but i was wrong
:|.

When I set "bindaddr" back to "0.0.0.0" then all is ok except that
Asterisk allow to connect and register SIP accounts from WAN.

My general question is:
How to make Asterisk listen only on LAN interface and use WAN interface
for Internet connections ?

I could set NAT from LAN => WAN but this would be ugly, so I prefer to
handle this problem using Asterisk configuration.



To solve it quickly... set up some kind of firewall with iptables and 
allow/disallow sip access from certain IP addresses.


Bartosz 


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RE: [Asterisk-Users] Cepstral , options to read the contents of a file

2006-05-01 Thread kevin ling

Hi,

You can call an agi script to convert the text file to wave format.

Example:
http://www.voip-info.org/wiki/view/swift.agi

Kevin


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Joseph
Sent: Monday, May 01, 2006 7:08 PM
To: Asterisk Users
Subject: [Asterisk-Users] Cepstral , options to read the contents of a file

Hi 
I had installed Cepstral , and it is working in Asterisk ,  it workfine
for exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,Cepstral( This is
Just a test ) exten => s,4,Cepstral(Hope u are getting this voices)

but instead of the text contents  for Cepstral , can I use the file name
location , where  it can read the file 
  Thanks 
  Joseph John 




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[Asterisk-Users] Listening on one IP and binding to other IP - is this possible ?

2006-05-01 Thread Lukasz Wojciechowski


Hello

Im new to Asterisk and generally to IPPBX.

Quick specification of my SOHO system:
1. 2 Interfaces: WAN and LAN
4. 5 VoIP phones configured with SIP attached to LAN
5. 1 external SIP line configured as "gateway" for VoIP phones to allow
outbound calls.

What I want is to set secure Asterisk platform. I would like Asterisk to
allow connections only from LAN but I would also like to use 1 SIP
account which requires registration on my provider sip server which
needs Internet connection.

When I set sip.conf "bindaddr" option to my LAN addres I was sure that
Asterisk will use WAN interface for Internet connections but i was wrong
:|.

When I set "bindaddr" back to "0.0.0.0" then all is ok except that
Asterisk allow to connect and register SIP accounts from WAN.

My general question is:
How to make Asterisk listen only on LAN interface and use WAN interface
for Internet connections ?

I could set NAT from LAN => WAN but this would be ugly, so I prefer to
handle this problem using Asterisk configuration.

Thaks in advance for any help

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Re: [Asterisk-Users] voicemail dialout



Doug Lytle wrote:

> What do you see at the console when someone presses 4 from voice mail?

-- Executing VoiceMailMain("SIP/502-ac3f", "s502") in new stack
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-tomakecall' (language 'en')
-- Playing 'vm-starmain' (language 'en')


-- 
Jon Farmer
Telford, Shropshire, UK
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[Asterisk-Users] Problems with zaptel and TE210P

2006-05-01 Thread Dan Brummer




Hello,
I'm just starting 
out with asterisk and I'm playing around with the system.  Currently I have 
a Digium TE210P connected to a PRI on the Asterisk server.  I have a SIP 
soft phone on my laptop for testing that is working fine.  When I try to 
place a call from my soft phone I get this from Asterisk:
 
May  1 09:11:41 
NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 
'Zap' (cause 34 - Circuit/channel congestion)
  == Everyone 
is busy/congested at this time (1:0/1/0)
  == Auto 
fallthrough, channel 'SIP/test-3a26' status is 'CONGESTION'
 
 
#/etc/zaptel.conf:
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
 
#/etc/asterisk/zapata.conf:
[channels]
switchtype=national
context=default
signalling=pri_cpe
group=1
channel => 
1-23
 
#/etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
 
[default]
exten => 
123,1,Answer()
exten => 
123,2,Playback(hello-world)
exten => 
123,3,Hangup()
 
exten => 
_9NXX,1,Dial(Zap/g1)
 
 
Any ideas?  
Thank you in advance, your help is greatly appreciated.
 
-Dan
 
 
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RE: [Asterisk-Users] auto-dail for ZAP channel, the application gets executed before the call attended

2006-05-01 Thread Michael Silveus

I'm still trying to accomplish the same thing and a solution might come from
Newman Telecom but I'm still waiting to see. Note: I have no affiliation
with Newman Telecom but they're application code appears to be the closest
to what I'm looking for.

The problem is because asterisk needs a way of emulating answer supervision
on the FXO circuit, which it does not currently have. If you tested your
callout on a local extension or VOIP it works because it traps events
internally for all extensions. But once you do a handoff to an outside line
that's the end of asterisk knowledge base until someone out there creates an
addon patch to do answer supervision for those outside lines. The problem is
that world wide there is no standard signaling tones for ringing, busy, fax,
answering machines, etc. The indications file does a great job of defining
the basics but let's say you're in the US so * is defined for US but if you
are calling internationally the ring back tones you receive will vary
greatly.

Mike

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Joseph
Sent: Monday, May 01, 2006 9:18 AM
To: Asterisk Users
Subject: [Asterisk-Users] auto-dail for ZAP channel,the application gets
executed before the call attended 



Hi All 
   when I try to use auto-dial to connect to
outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use 
Channel: ZAP/1/050745 in my sample.call file , if 
I use Channel:SIP/326 ,  it works  fine 
  my "sample.call" file contains 
Channel: ZAP/1/050745
Callerid: Asterisk
MaxRetries: 2
RetryTime: 10
WaitTime: 60
Context: outboundmsg1
Extension: s
Priority: 1

and my "extensions.conf" file  contains
[outboundmsg1]
exten => s,1,DigitTimeout,5
exten => s,2,ResponseTimeout,30   
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Cepstral(Hi ,, Hope u are getting this
voices, It is text to Voice  conversation , did u hear
it  )  
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup

  Guidance requested , to know the
reason , why the  "Ceptral " application is executed ,
before the call is attended when I user ZAP channel .
But  for  SIP channel (only SIP internal call , I made
) it is working fine 
Thanks 
Joseph John 



 





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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Doug Lytle wrote:

  

Let's see that section of your voicemail.conf



  

What do you see at the console when someone presses 4 from voice mail?

Doug

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Re: [Asterisk-Users] auto-dail for ZAP channel, the application gets executed before the call attended

2006-05-01 Thread Moises Silva


The problem is that "Answer"  for Zap channels, means that the FXO
card has accepted the call and is in process of making the call, but
that does not means that the other end (ie. the phone in the PSTN) has
already answered. Personally I have the same problem, but I have not
looked further on the code to solve it because I havent had the time.
I suggest you 3 options:

1. Wait 2 or 3 weeks, so I have some free time and came up with a fix.
2. Ask in bugs.digium.com for a feature to know this.
3. Keep looking and questioning about it in asterisk-users, may be
someone has a workaround.

Regards

On 5/1/06, John Joseph <[EMAIL PROTECTED]> wrote:


Hi All
   when I try to use auto-dial to connect to
outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use
Channel: ZAP/1/050745 in my sample.call file , if
I use Channel:SIP/326 ,  it works  fine
  my "sample.call" file contains
Channel: ZAP/1/050745
Callerid: Asterisk
MaxRetries: 2
RetryTime: 10
WaitTime: 60
Context: outboundmsg1
Extension: s
Priority: 1

and my "extensions.conf" file  contains
[outboundmsg1]
exten => s,1,DigitTimeout,5
exten => s,2,ResponseTimeout,30
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Cepstral(Hi ,, Hope u are getting this
voices, It is text to Voice  conversation , did u hear
it  )
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup

  Guidance requested , to know the
reason , why the  "Ceptral " application is executed ,
before the call is attended when I user ZAP channel .
But  for  SIP channel (only SIP internal call , I made
) it is working fine
Thanks
Joseph John









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Re: [Asterisk-Users] Softphone ready to go installed on USB flash drive

2006-05-01 Thread Ronald Wiplinger

Bruce Reeves wrote:
I do this with the windows version of idefisk from Asteriskguru.com 
. The configuration is stored in the dir with 
the program and dll. I have actually configured it and emailed it to 
users. There is no installer and a simple shortcut or autoplay menu 
should take care of the rest.

It is a nice phone, but it is IAX. I would like to use a SIP phone.
The reason for that is that there is no IAX server for the mass, but 
openSER 

bye

Ronald Wiplinger

On 5/1/06, *Time Bandit* <[EMAIL PROTECTED] 
> wrote:

> How can I install a softphone on my USB flash drive like Xlite
and have
> it ready to go when I plug  it in at any Windows XP computer?
> (Same for a Linux softphone, both on one USB flash drive).
I believe Dan's softphone is suitable for this. See
http://www.laser.com/dante/diax/diax.html

Actually, I should do that with my softphone instead of using the
registry :(

hth
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--
Bruce
Nortex Networks

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Re: [Asterisk-Users] voicemail dialout



Doug Lytle wrote:

> Let's see that section of your voicemail.conf

; tz=central; Timezone from zonemessages above.  Irrelevant
if envelope=no.
; attach=yes; Attach the voicemail to the notification email
*NOT* the pager email
; saycid=yes; Say the caller id information before the
message. If not described,
; or set to no, it will be in the envelope
; cidinternalcontexts=intern; Internal Context for Name Playback
instead of extension digits when saying caller id.
; sayduration=no; Turn on/off the duration information before
the message. [ON by default]
; saydurationm=2; Specify the minimum duration to say. Default
is 2 minutes
;dialout=from-vm; Context to dial out from [option 4 from the
advanced menu]
; if not listed, dialing out will not be
permitted
sendvoicemail=no; Context to Send voicemail from [option 5 from
the advanced menu]
; if not listed, sending messages from inside
voicemail will not be
; permitted
; searchcontexts=yes; Current default behavior is to search only the
default context
; if one is not specified.  The older behavior
was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; callback=fromvm   ; Context to call back from
; if not listed, calling the sender back
will not be permitted
; review=yes; Allow sender to review/rerecord their message
before saving it [OFF by default
operator=no ; Allow sender to hit 0 before/after/during
leaving a voicemail to
; reach an operator  [OFF by default]
; envelope=no   ; Turn on/off envelope playback before message
playback. [ON by default]
; This does NOT affect option 3,3 from the
advanced options menu
; delete=yes; After notification, the voicemail is deleted
from the server. [per-mailbox only]
; This is intended for use with users who
wish to receive their voicemail ONLY by email.
; nextaftercmd=yes  ; Skips to the next message after hitting 7 or 9
to delete/save current message.
; [global option only at this time]
; forcename=yes ; Forces a new user to record their name.  A new
user is
; determined by the password being the same as
; the mailbox number.  The default is "no".
; forcegreetings=no ; This is the same as forcename, except for
recording
; greetings.  The default is "no".
; hidefromdir=yes   ; Hide this mailbox from the directory produced
by app_directory
; The default is "no".



Jon Farmer
Telford, Shropshire, UK
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[Asterisk-Users] Can i use same group with 2 or more hfc-cards ?

2006-05-01 Thread office

 
Can i use same group in zapata.conf with 2 or more hfc-cards ?

Is this possible :

instaed of Dial(Zap/g1/10&Zap/g2/10)

Dial(Zap/g1/10) - with g1 fuer both cards.


cu

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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Doug Lytle wrote:

  

You'll also need to do a stop/start of Asterisk.



Done that also, no difference

  

Let's see that section of your voicemail.conf

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Re: [Asterisk-Users] Softphone ready to go installed on USB flashdrive

2006-05-01 Thread Bruce Reeves

Check out the zip version at under a meg.On 5/1/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:





The current versions of IDEFISK use a Windows installer, 
wether it is required or not now I dont know.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] On Behalf Of Bruce 
  ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Softphone ready to go installed on USB 
  flashdrive
  I do this with the windows version of idefisk from Asteriskguru.com. The configuration is 
  stored in the dir with the program and dll. I have actually configured it and 
  emailed it to users. There is no installer and a simple shortcut or autoplay 
  menu should take care of the rest. 
  On 5/1/06, Time 
  Bandit <[EMAIL PROTECTED]> 
  wrote:
  > 
How can I install a softphone on my USB flash drive like Xlite and 
have> it ready to go when I plug  it in at any Windows XP 
computer?> (Same for a Linux softphone, both on one USB flash 
drive).I believe Dan's softphone is suitable for this. See http://www.laser.com/dante/diax/diax.html
Actually, 
I should do that with my softphone instead of using the registry 
:(hth___ 
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-- BruceNortex Networks 

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http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
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RE: [Asterisk-Users] Softphone ready to go installed on USB flashdrive

2006-05-01 Thread John covici

Does idefisk support that iogear usb phone that I think you were
talking about?

on Monday 05/01/2006 Kerry Garrison([EMAIL PROTECTED]) wrote
 > The current versions of IDEFISK use a Windows installer, wether it is
 > required or not now I dont know.
 > -Kerry
 >  
 > 
 > 
 >   _  
 > 
 > From: [EMAIL PROTECTED]
 > [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
 > Sent: Monday, May 01, 2006 9:13 AM
 > To: Asterisk Users Mailing List - Non-Commercial Discussion
 > Subject: Re: [Asterisk-Users] Softphone ready to go installed on USB
 > flashdrive
 > 
 > 
 > I do this with the windows version of idefisk from Asteriskguru.com. The
 > configuration is stored in the dir with the program and dll. I have actually
 > configured it and emailed it to users. There is no installer and a simple
 > shortcut or autoplay menu should take care of the rest. 
 > 
 > 
 > On 5/1/06, Time Bandit <[EMAIL PROTECTED]> wrote: 
 > 
 > > How can I install a softphone on my USB flash drive like Xlite and have
 > > it ready to go when I plug  it in at any Windows XP computer?
 > > (Same for a Linux softphone, both on one USB flash drive).
 > I believe Dan's softphone is suitable for this. See 
 > http://www.laser.com/dante/diax/diax.html
 > 
 > Actually, I should do that with my softphone instead of using the registry
 > :(
 > 
 > hth
 > ___ 
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 > 
 > Asterisk-Users mailing list
 > To UNSUBSCRIBE or update options visit:
 >http://lists.digium.com/mailman/listinfo/asterisk-users
 > 
 > 
 > 
 > 
 > 
 > -- 
 > Bruce
 > Nortex Networks 
 > 
 > 
 > 
 > 
 > 
 > 
 >  color=#ff size=2>The current versions of IDEFISK use a Windows 
 > installer, 
 > wether it is required or not now I dont know.
 >  color=#ff size=2>-Kerry
 >  class=138453616-01052006> 
 >  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #ff 2px solid; 
 > MARGIN-RIGHT: 0px">
 >   
 >   
 >   From: [EMAIL PROTECTED] 
 >   [mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
 >   ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk 
 >   Users Mailing List - Non-Commercial DiscussionSubject: Re: 
 >   [Asterisk-Users] Softphone ready to go installed on USB 
 >   flashdrive
 >   I do this with the windows version of idefisk fromhref="http://Asteriskguru.com";>Asteriskguru.com. The configuration is 
 >   stored in the dir with the program and dll. I have actually configured it 
 > and 
 >   emailed it to users. There is no installer and a simple shortcut or 
 > autoplay 
 >   menu should take care of the rest. 
 >   On 5/1/06, Time 
 >   Bandit <   href="mailto:[EMAIL PROTECTED]">[EMAIL PROTECTED]> 
 >   wrote:
 >  style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: 
 > rgb(204,204,204) 1px solid">> 
 > How can I install a softphone on my USB flash drive like Xlite and 
 > have> it ready to go when I plug  it in at any Windows 
 > XP 
 > computer?> (Same for a Linux softphone, both on one USB flash 
 > drive).I believe Dan's softphone is suitable for this. See  
 > href="http://www.laser.com/dante/diax/diax.html";>http://www.laser.com/dante/diax/diax.htmlActually,
 >  
 > I should do that with my softphone instead of using the registry 
 > :(hth___ 
 > --Bandwidth and Colocation provided by  href="http://Easynews.com";>Easynews.com --Asterisk-Users 
 > mailing 
 > listTo UNSUBSCRIBE or update options visit:    
 > href="http://lists.digium.com/mailman/listinfo/asterisk-users";>http://lists.digium.com/mailman/listinfo/asterisk-users  
 >   clear=all>-- BruceNortex Networks 
 > ___
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 > To UNSUBSCRIBE or update options visit:
 >http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]
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Re: [Asterisk-Users] voicemail dialout



Doug Lytle wrote:

> You'll also need to do a stop/start of Asterisk.

Done that also, no difference

-- 
Jon Farmer
Telford, Shropshire, UK
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[Asterisk-Users] 7941G - Any success stories?

2006-05-01 Thread Aaron Daniel

Has anyone successfully gotten the 7941G working on Asterisk?  We're 
looking at getting some of those instead of the 7940's, but there's really 
not much info out there about them.


--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Doug Lytle wrote:

  

It's enabled/disabled via the voicemail.conf



I have commented out

dialout=from-vm

but the option is still given even though any number dialled results in
unobtainable. So I dont want the option given.


  

You'll also need to do a stop/start of Asterisk.

Doug

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RE: [Asterisk-Users] Softphone ready to go installed on USB flashdrive

2006-05-01 Thread Kerry Garrison




The current versions of IDEFISK use a Windows installer, 
wether it is required or not now I dont know.
-Kerry
 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Bruce 
  ReevesSent: Monday, May 01, 2006 9:13 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Softphone ready to go installed on USB 
  flashdrive
  I do this with the windows version of idefisk from Asteriskguru.com. The configuration is 
  stored in the dir with the program and dll. I have actually configured it and 
  emailed it to users. There is no installer and a simple shortcut or autoplay 
  menu should take care of the rest. 
  On 5/1/06, Time 
  Bandit <[EMAIL PROTECTED]> 
  wrote:
  > 
How can I install a softphone on my USB flash drive like Xlite and 
have> it ready to go when I plug  it in at any Windows XP 
computer?> (Same for a Linux softphone, both on one USB flash 
drive).I believe Dan's softphone is suitable for this. See http://www.laser.com/dante/diax/diax.htmlActually, 
I should do that with my softphone instead of using the registry 
:(hth___ 
--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing 
listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks 
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[Asterisk-Users] CallerID Name problem

 
I'm having trouble getting callerid name to show up on my phones (Cisco
7960 and a few softphones)
When I look in the CDR database I see the name but not on any phone when
being called.

I'm running
Asterisk SVN-trunk-r7498 built on 2006-04-30 15:11:39 UTC 


Any help would be great !
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Re: [Asterisk-Users] voicemail dialout



Doug Lytle wrote:

> It's enabled/disabled via the voicemail.conf

I have commented out

dialout=from-vm

but the option is still given even though any number dialled results in
unobtainable. So I dont want the option given.


-- 
Jon Farmer
Telford, Shropshire, UK
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Re: [Asterisk-Users] voicemail dialout


Jon Farmer wrote:

Hi

How do i disable dialling out from voicemail?


  

It's enabled/disabled via the voicemail.conf

Doug

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Re: [Asterisk-Users] auto-dail for ZAP channel,


John Joseph wrote:
Hi All 
   when I try to use auto-dial to connect to

outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use 
Channel: ZAP/1/050745 in my sample.call file , if 
  


My fix for this is to loop an audio file, requiring the answering party 
to press a key before playing a message.  If it loops for a number of 
times, without a response, hangup.


Doug

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[Asterisk-Users] voicemail dialout