from:"Uriel Carrasquilla"



Uriel -
   1) Please stop top-posting.

   2) I'm afraid I don't have any data on specifics of creating a
front-end.  I know how to do it, but my time these days is spent
writing lots of other projects that I have been doing.  :-)  I would
suggest you get SER and set it up - it's quite easy, and the
documentation on SER itself is very well written, and if you have a
good idea of how SIP works you should be able to patch together an
appropriate system.  However, if you aren't 100% familiar with how
SIP works, I would stick to just an Asterisk system; SER doesn't
allow for any of the shortcuts that Asterisk has.

   3) Use Google and do some searching.  I found some quick links with
a few of the keywords that would seem obvious, but I don't have
enough time to review them...

JT


John:
Thank you for responding.  I am in the process of installing SER and hope to
have it ready by this weekend.  I am in the process of installing some
equipment at a local colo.

I have to tell you, at the expense of offending you, that I use MS-Outlook
and the responses go to the tope of the messages.  At work I use Lotus Notes
and the same thing happens.  Before, I used PROFS (on mainframes) and the
same principle applied.  All in all, 20+ years of using this principle for
e-mails at both work and home.  As a matter of fact, I am of the opinion
that the response to E-mails should go at the top to save time.  However,
this is not about me but the * group and the well being of this list.  Does
anybody else have a strong opinion one way or the other?  If it is left to
John and myself we have a 1:1 vote.

Regards,
Uriel


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] X100P Echo Problems..What's going to happen?

John:
I
don't use MSN so I can't comment. I do know that when my connections are
pure VoIP (no analog PSTN connections), the quality is better if enough
bandwidth is available.
TCP is
a protocol that gets used when you want to make sure a packet arrives at the
other end. UDP is better for voice because you don't want packets to be
retransmitted and have to wait to assemble them on the other end in sequence so
the conversation makes sense.
Regards,
Uriel

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of John
MSent: Monday, October 13, 2003 11:18 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P
Echo Problems..What's going to happen?

What I dont
understand is why MSN messenger is perfect with no echo? I switch back and forth and still hear
a big difference. I believe MSN
is using TCP rather than UDP. Can
* run on TCP rather than UDP? I
think this makes senses and can eliminate that
echo.

-Original
Message-From: Uriel
Carrasquilla [mailto:[EMAIL PROTECTED] Sent: Monday, October 13,
2003 10:02
PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] X100P Echo
Problems..What's going to happen?

John:

I have
been around voice over data packets for quite a few years and I am still to
see the perfect system that works identical to circuit switching 100% of the
time. My opinion is that there is a lot more to the story than just
parameters. Packet loses, double compressions, faulty routers,
bandwidth, analog to digital and so on can get in the
way.

On the
other hand, if your customer understand the benefits, and I mean more than
cost, and can leave with 80% perfect, then you will be able to understand why
a lot of companieshaveopted for VoIP (or ATM or Frame
Relay).

Regards,

Uriel-Original
Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of John MSent: Monday, October 13,
2003 1:41
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] X100P Echo
Problems..What's going to happen?Importance: High

Ive read and experienced the
echo problems with the X100P.
Is Digium going to fix the problem or refund our money? I want to see this work because
myself and other small companies out there use analog lines. I would trade up to T1 but that
requires me to have at least 9 lines.
If I did trade up, do the T1 cards work perfectly with no echo at
all? I get echo with my
directly connected computer using Xten SIP. No matter with all the suggestions
to change the parameters, it still has echo.

Does anyone have the T1 and have
no problems at all? I would
surely appreciate you experiences.
Whats my option to get this too work
flawlessly?

John

RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

Andre:
This makes a lot of sense.  I had used Asterisk in the past to play the role
of Gatekeeper for directing traffic to the appropriate Asterisk acting as a
PSTN gateway.  IAX does a heck of a good job in that configuration.
However, with SIP, I have run into nothing but trouble with registrations
falling off.
I have read the SER manual I am going to jump into it, now that I know that
in practice it works and it is not only theory in a manual.
Thank you,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andres
Sent: Tuesday, October 14, 2003 12:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)


On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
 John:
 are you aware of any documentation on how to configre SER to be a
front-end
 to Asterisk?
Hi Uriel,

At TeleSIP we run a cluster of several geographically distributed SER
Servers
that hande all our SIP Routing.   SER is a robust, fast and stable platform
which has worked flawlessly for us.  We use * as our company PBX and PSTN
Gateway.  Basically what you need to do is to device a numbering plan so
that
SERs routing logic can forward the call to * when it needs to.

For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
  if (method==INVITE) {
 if (uri=~sip:[EMAIL PROTECTED]) {
  log(1, This is a Long Distance Call\n);
  route(6);
  break;
  };
  };
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] */SER/FW

Steve:
Unless Asterisk is on the public side of the Internet, you will run into
problems if the UA (SIP phones) are behind a NAT.
In the scenario you presented, I think SER would be used for all calls
between SIP phones and they would only go to Asterisk when you need to
Gateway into the PSTN some of the calls.

Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, October 14, 2003 6:43 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] */SER/FW


Hi,

I've just read the postings regarding the interworking between * and SER.
As these persons seem quite knowledgeable on this, I would like to have
their advise on my planned installation:

- I have broadband cable access
- I plan to install a SIP-aware router
- I plan to install a Linux server with Digium analog IF card(s) for
connection to my analog line (incoming and outgoing)
- I plan to install Asteriks on that server
- I plan to install a SIP-proxy,registrar on the same server (I've been
looking at iptel's SER)
- I plan to use the Budgetone SIP phones
- I plan to have a public (static) IP address

All this to have my own little phone company for me and my family/friends
as we are spread over Europe (high international phone costs!). Calling
eachother on our SIP phones and also being able to use eachother's PBX's to
make local calls. I would host the SIP Registrar (in stead of outsourcing).

My main question lies in the interworking between iptel's SER and Asteriks.
Not only on the configuration side, but also on the network side (here I
mean: can both run on the same server, or do they need to have different IP
addresses, ...).

Does anybody have a diagram or any practical guide on how to install this?

Thanks,

Steven

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring

Don't forget to reverse the FXO/FXS in the TA750.  They are opposite to the
asterisk config files.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason Piterak
Sent: Tuesday, October 14, 2003 5:13 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T100P to Adtran TA750 - No dialtone or ring

Hello all,

  I've got a T100P connected to an Adtran TA750 with a T1 crossover...
This connects to a patch panel with phone ports. The Adtran is fully
populated with FXS cards.
  All I get on any phone port is a fast clicking noise... No dialtone.
  Asterisk 'sees' the card, (the channels show up in /proc/zaptel).
Incoming calls are routed to the zap/x channel, but no ring.

I'm hoping I'm overlooking something stupid.

Thanks ahead of time...

--Jason

  Here are some (possibly) relevant snippits from various places:

o T100 LED shows green...
o Not showing any errors in /var/log/asterisk/messages (debug logging
enabled)
o Adtran config is set to:
  -- 2. Provisioning Templates
  -- 1. Factory Default (ESF,B8ZS,Loopstart)'

o OS/hardware:
  System OS:
   debian testing/unstable
   kernel: Custom 2.4.22-ck
(Con Kolivas patch set:
  o Preempt
  o Low-latency
  o AA vm hacks
  o RL2 Desktop Tuning
  o Debian logo in FB)

  Digium Cards:
T100P --FXS
X100P --FXO

o Asterisk version:
  Asterisk CVS-10/02/03-17:52:20

--
asterisk:~# cat /proc/zaptel/1
Span 1: WCT1/0 Digium Wildcard T100P T1/PRI Card 0 B8ZS/ESF
IRQ misses: 21904

 1 WCT1/0/1 FXOKS (In use)
 2 WCT1/0/2 FXOKS (In use)
 3 WCT1/0/3 FXOKS (In use)
 4 WCT1/0/4 FXOKS (In use)
 5 WCT1/0/5 FXOKS (In use)
 6 WCT1/0/6 FXOKS (In use)
 7 WCT1/0/7 FXOKS (In use)
 8 WCT1/0/8 FXOKS (In use)
 9 WCT1/0/9 FXOKS (In use)
10 WCT1/0/10 FXOKS (In use)
11 WCT1/0/11 FXOKS (In use)
12 WCT1/0/12 FXOKS (In use)
13 WCT1/0/13 FXOKS (In use)
14 WCT1/0/14 FXOKS (In use)
15 WCT1/0/15 FXOKS (In use)
16 WCT1/0/16 FXOKS (In use)
17 WCT1/0/17 FXOKS (In use)
18 WCT1/0/18 FXOKS (In use)
19 WCT1/0/19 FXOKS (In use)
20 WCT1/0/20 FXOKS (In use)
21 WCT1/0/21 FXOKS (In use)
22 WCT1/0/22 FXOKS (In use)
23 WCT1/0/23 FXOKS (In use)
24 WCT1/0/24 FXOKS (In use)

asterisk:~# cat /proc/zaptel/2
Span 2: WCFXO/0 Wildcard X101P Board 1
 25 WCFXO/0/0 FXSKS (In use)

---

asterisk:/etc/asterisk# cat /etc/zaptel.conf
#T1:
span=1,0,0,esf,b8zs
fxoks=1-24
loadzone = us
defaultzone=us

#X100P - Single-line FXO card
fxsks=25

-

asterisk:/etc/asterisk# cat zapata.conf
...
[channels]
;T1-fxo (incomming channels) on the channel bank
;-
; Section commented out until we have an fxo card in the CB
;context = bell
;language = en
;signalling = fxs_ks
;usecallerid = yes
;hidecallerid = no
;echocancel = yes
;echocancelwhenbridged = no
;;if immediate is set to yes, asterisk will automatically answer the line
;;and jump to the 's' extension for the context.
;;immediate = yes
;group = 1
;channel = 1

;T1-fxs (inside handsets) on the channel bank
context = local
language = en
signalling = fxo_ks
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = no
;callwaitingcallerid=yes --Change if Callwaiting is yes
threewaycalling = yes
transfer = yes
cancelforward = yes
callreturn = no
echocancel = yes
echocanelwhenbridged = no
immediate = no
rxgain=0.0
txgain=0.0
channel = 1-24
...

;SinglePort-fxo (incomming channels)
context = bell
language = en
signalling = fxs_ks
usecallerid = yes
hidecallerid = no
echocancel = yes
echocancelwhenbridged = no
;if immediate is set to yes, asterisk will automatically answer the line
;and jump to the 's' extension for the context.
immediate = yes
group = 1
channel = 25

-

Console during a call being routed to zap/1:
-- Starting simple switch on 'Zap/25-1'
-- Executing BackGround(Zap/25-1, thankyou) in new stack
-- Playing 'thankyou'
  == CDR updated on Zap/25-1
-- Executing Goto(Zap/25-1, mainmenu|s|2) in new stack
-- Goto (mainmenu,s,2)
-- Executing BackGround(Zap/25-1, greeting-announcements) in new
stack
-- Playing 'greeting-announcements'
  == CDR updated on Zap/25-1
-- Executing Goto(Zap/25-1, routing|300|1) in new stack
-- Goto (routing,300,1)
-- Executing Macro(Zap/25-1, oneline|300|Zap/1) in new stack
-- Executing DBget(Zap/25-1, fwdexten=CFU/300) in new stack
-- DBget: varname=fwdexten, family=CFU, key=300
-- DBget: Value not found in database.
-- Executing Goto(Zap/25-1, s|4) in new stack
-- Goto (macro-oneline,s,4)
-- Executing Dial(Zap/25-1, Zap/1||) in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
  == Spawn extension (macro-oneline, s, 4) exited non-zero on 'Zap/25-1' in
macro 'oneline'
  == Spawn extension (routing, s, 1) exited non-zero on 'Zap/25-1'

asterisk:~# lsmod
Module  Size  Used by

RE: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was: No sound with SIP Phones on the Internet))

Excellent points in the printed world.  I am not certain that from mail to
eMail I would use the same principles.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: Tuesday, October 14, 2003 6:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was:
No sound with SIP Phones on the Internet))

On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote:
 I have to tell you, at the expense of offending you, that I use
 MS-Outlook and the responses go to the tope of the messages.  At work
 I use Lotus Notes and the same thing happens.  Before, I used PROFS
 (on mainframes) and the same principle applied.  All in all, 20+
 years of using this principle for e-mails at both work and home.  As
 a matter of fact, I am of the opinion that the response to E-mails
 should go at the top to save time.  However, this is not about me but
 the * group and the well being of this list.  Does anybody else have
 a strong opinion one way or the other?  If it is left to John and
 myself we have a 1:1 vote.

This is all you really need to know:

http://learn.to/quote/

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

OK OK OK, I got it.  See my response inside the body of your E-mail.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw
Sent: Tuesday, October 14, 2003 8:44 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)

On 15/10/03 00:15, Uriel Carrasquilla wrote:

 Does anybody else have a strong opinion one way or the other? If it
 is left to John and myself we have a 1:1 vote.

See how much easier it is to follow the thread of conversation if you
quote just enough of the e-mail you're responding to so people know
what's going on without having to read through pages of text?

[URIEL] - I have to learn how to quote with Outlook.

Please see RFC 1855:
  - http://www.faqs.org/rfcs/rfc1855.html

Decent mail clients that behave sensibly regarding quoting are easy to
come by. You can even set up Outlook to behave vaguely properly and
quote using .

 As a matter of fact, I am of the opinion that the response to E-mails
 should go at the top to save time.

So, it's not worth *your* time organizing your e-mail sensibly, but it's
worth everyone else's time having to dig through lines of text to work
out what the context is? I find that selfish, at best.

[URIEL] you are absolutely right and I do apologize.  Ignorance is not an
excuse.

Please see the following page (strong words warning). It pretty much
sums it all up nicely:
  - http://thegestalt.org/simon/quoterant.html

[URIEL] Thank you.

--
Al Maw

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] X100P Echo Problems..What's going to happen?

John:
I have
been around voice over data packets for quite a few years and I am still to see
the perfect system that works identical to circuit switching 100% of the
time. My opinion is that there is a lot more to the story than just
parameters. Packet loses, double compressions, faulty routers, bandwidth,
analog to digital and so on can get in the way.
On the
other hand, if your customer understand the benefits, and I mean more than cost,
and can leave with 80% perfect, then you will be able to understand why a lot of
companieshaveopted for VoIP (or ATM or Frame
Relay).
Regards,
Uriel-Original
Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of John
MSent: Monday, October 13, 2003 1:41 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] X100P Echo
Problems..What's going to happen?Importance:
High

Ive read and experienced the echo
problems with the X100P. Is Digium going to fix the problem or refund our money? I want to see this work because myself and other small companies out there use analog
lines. I would trade up to T1 but
that requires me to have at least 9 lines. If I did trade up, do the T1 cards
work perfectly with no echo at all?
I get echo with my directly connected computer using Xten SIP. No
matter with all the suggestions to change the parameters, it still has echo.

Does anyone have the T1 and have
no problems at all? I would
surely appreciate you experiences.
Whats my option to get this too work
flawlessly?

John

RE: [Asterisk-Users] No sound with SIP Phones on the Internet

Dunca:
I am not sure I understand your statemnet.
SIP devices (UA) on the other side of the Internet behnid a NAT communicate
to * on the public Internet.  Then this Asterisk connects to other Asterisks
(via IAX) that can be behind Firwalls (or NATS).  am I understanding
correctly?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of duncan
Sent: Monday, October 13, 2003 12:25 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No sound with SIP Phones on the Internet



This is bull... I can't believe that...
Must be a solution...

sip is very tricky to get working behind firewalls.  sip clients work quite
well with nat, just make sure nat=yes is in the sip profile in sip.conf

my solution has always been to put an asterisk box behind the firewall and
make all the sip clients connect to that, then IAX out of the firewall to
the other machines.  i spent a few days trying unsuccessfully to find a
decent sip proxy that worked the way i wanted and decided that the asterisk
solution was much better.



duncan

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

Chris:
I am glad to see someone else asking the same question I have been asking
myself.
As soon as I get my public IP address, I will install SER on the public side
and Asterisk behind a NAT (with dynamic IP) to see if I can get around
problems I have when my SIP (UA) behind their own NAT on the other side of
my Internet connection.
If you make any progress, please share.  I will do the same.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Albertson
Sent: Monday, October 13, 2003 7:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)



I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.

SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.

SER is GPL'd and has very good user documentation.  Don't know how well
the above will work.  The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive

One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing  to edit *.conf files.

See here for details http://www.iptel.org/ser/


=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
http://shopping.yahoo.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
I suspect it is very inexpensive to put a SER server in a hosting facility
to forward traffic to multiple Asterisks based on Least Cost Routing.
My problem is that my experience is with Asterisk and not with SER.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Monday, October 13, 2003 8:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)


I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.

SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.

SER is GPL'd and has very good user documentation.  Don't know how well
the above will work.  The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive

One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing  to edit *.conf files.

See here for details http://www.iptel.org/ser/


=
Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK

SER is an excellent option as a front end to Asterisk.  It is a
true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been
the primary focus of Asterisk development.  In fact, Asterisk's SIP
implementation is very limited (though it is extremely pragmatic.)

However, moving to SER does not solve any of the issues about the
proxy being behind a NAT, and I believe that SER will have the same
problems (though I could be wrong on this; I haven't experimented
with SER's ability to work from behind a NAT.)   SIP clients work
well enough behind NAT (most of them, anyway) but the servers are a
different story.

I really like SER's third-party addons for account administration;
Asterisk is significantly more complex, and probably would not be as
easily converted to such a front end.  In fact, SER has a very
complex routing/scripting language that is not easily administered
with a web front end, so I think that SER and Asterisk suffer from
the same problems.  If someone were to come up with a simple way to
administer voicemail.conf and sip.conf from a web tool, that would go
far to making Asterisk a bit more user-accessible...

JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / IAX over satellite

are you using Frame Relay? how big are the packets? I don't think you would
be using ATM over a satellite VSAT modem.  Also don't be fooled, satellite
modems don't speak IP.  There is normally an IP edge device that makes you
believe it is IP.
The 500 ms delay is the speed of light.  You need to add the time for the
protocols used by the satellite links.  On top of that, you have the delays
of IAX itself when it assembles the packets.
All in all, you might be experiencing delays closer to 800 ms or more if
VSAT modem is poorly configured.
The other question to ask is if you are fully duplex.  In the satellite
world you do need two independ up/down links.
IP over satellite is crazy.  By nature satellites are broadcasting. Voice
over satellite is already a challenge.  I have done IAX over satellite but
using small frame relay packets (256) and stopping any error retransmissions
on the satellite modems.  The result can be acceptable if you train your
users to the 1/2 second delay and the fact that two people cannot talk at
the same time (a challenge for us in South America).
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: Saturday, October 11, 2003 7:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / IAX over satellite



On Saturday, Oct 11, 2003, at 17:04 America/Chicago, Paul Mahler wrote:

 Which satellite system?

 I think you need some specialized support, even special hardware.
 Check
 out

 http://www.groundcontrol.com/igvoip_001.htm

 You may need to replace TCP/IP

 http://www.mentat.com/skyx/skyx-gateway.html

I don't know why he'd need to replace TCP/IP when both SIP as well as
IAX use UDP/IP.  There may be substantial latency, however.

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / IAX over satellite

Dream on about the 100 ms or less.  Once you get to the satellite, it is the
same time regardless of where you are going on the foot print of the
satellite.   Speed of Light does not understand American speed limits.  Of
for that matter Europeans.  The speed of light is constant.  Just pick up
your calculator, take the distance to travel and you will see your 1/4
second each way. That does not include all the delays caused by modems,
routers, packetizing streams, etc.
Regards
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, October 11, 2003 7:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / IAX over satellite


I was looking into using satelitte for a backup internet connection at one
stage, iirc, its:

- 500ms transmit/recieve latency
- if yours sat connection terminates in the us, you should be able to reach
  most place in 30ms
- if you're going to europe (from the termination of the sats in the .us),
it
  will most likely be 80ms.

Those figures I got from about 2 different sales people. But this is from
memory.

(That said, in all cases it was better than a landline internet
connection,
due to the country and surrounding countries.)

Thanks,
Andrew Griffiths

On Sat, Oct 11, 2003 at 06:03:24PM -0500, Tilghman Lesher wrote:

 On Saturday, Oct 11, 2003, at 17:04 America/Chicago, Paul Mahler wrote:

 Which satellite system?
 
 I think you need some specialized support, even special hardware.
 Check
 out
 
 http://www.groundcontrol.com/igvoip_001.htm
 
 You may need to replace TCP/IP
 
 http://www.mentat.com/skyx/skyx-gateway.html

 I don't know why he'd need to replace TCP/IP when both SIP as well as
 IAX use UDP/IP.  There may be substantial latency, however.

 -Tilghman

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] T100P Phones Configuration

I am not sure I understand the comments but please allow me to simplify.
1) 1xT1 in the T400P goes to the Telco provided T1 connection.
2) 1xT1 in the T400P goes to the Channel Bank.
3) The channel bank breaks up the FXO or FXS analog.
I would suggest you stay away from a Bank Channel to receive the Telco T1.
It is just using extra equipment which makes things more complex.  Hope you
subscribe to the KISS principle,  I do.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Saturday, October 11, 2003 10:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] T100P  Phones Configuration


So...

I would need as you noted two T100P cards or a T400P.  The T1 goes into
the * Server and the second port of a T400P goes back to the asterisk
server. Then the extensions get broken out from the Channel bank?

Geoff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Sharp
Posted At: Saturday, October 11, 2003 12:02 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] T100P  Phones Configuration
Subject: Re: [Asterisk-Users] T100P  Phones Configuration


 Below you will find, what I believe to be a typical setup with a T100P

 card.  My question is -

 1. Is this correct?

Possibly.  Depends on if you use a channel bank that can do add/drop and
you're not using a PRI.

You'll take your incoming T1 and go into 1 T100P and use another T100P
to feed out to your channel bank...or you can get a T400P and just have
one card in the system.



 2. What kind of phones would be needed here... (Would you have to use 
 Digital phones)  And if so what would you recommend.


You can use anything from a $9 WalMart phone to a $300 ADSI analog
phone.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
attachment: winmail.dat

RE: [Asterisk-Users] No sound with SIP Phones on the Internet




is 
your SIP phone behind a NAT? is* behind a NAT? 

Uriel

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Chris 
  HarigaSent: Sunday, October 12, 2003 10:42 PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] No sound 
  with SIP Phones on the Internet
  
  Hi,
  
  I need some help with my sip 
  phones. I have a Xten softphone and a Budge Tone 101 from Grandstream.
  If Im connected from my LAN all 
  is fine but from the Internet I connect the phone but I dont have the 
  sound.
  Asterisk SLI show me this when I 
  try to call my voicemail:
  
  localhost*CLI
   -- Executing VoiceMailMain("SIP/chariga-c067", "105") in new 
  stack
   == Parsing '/etc/asterisk/voicemail.conf': == Parsing '/etc/asterisk/voicemail.conf': Found
   -- Playing 'vm-password'
   == Spawn extension (internal, 205, 1) 
  exited non-zero on 'SIP/chariga-c067'
   -- Unregistered SIP 'chariga'
  localhost*CLI
  
  Any help is 
  welcome.
  
  Best 
  regards,
  
  Chris 
  Hariga

RE: [Asterisk-Users] T100P Phones Configuration

2003-10-11 Thread Uriel Carrasquilla

The PRI goes right into the T100P.
I forgot if the T100P is the one with more than 1xT1, if so, the 2nd T1 in
the T100P goes to the Channel Bank.
Uriel 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of PBX
Sent: Friday, October 10, 2003 11:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] T100P  Phones Configuration

Below you will find, what I believe to be a typical setup with a T100P
card.  My question is - 

1. Is this correct?

2. What kind of phones would be needed here... (Would you have to use
Digital phones)  And if so what would you recommend.

PRI/T1-
   |
   |
   |

  |   |
  | Channel Bank  |
  |   |

  |  | ||
  |  |Amphenol |  24 Port Patch | 
  |  --|   Panel|
  |||
  -||
  ||
  | * Server   |  ||  ||
  | T100P  |  ||  ||
  ||  ||  ||
  - Phone Phone Phone Phone

Thank you,

Geoff
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
attachment: winmail.dat

[Asterisk-Users] SER versus Asterisk for WAN SIP Phones

2003-10-08 Thread Uriel Carrasquilla




I have researched 
the topic and would be interested to see the "theory versus practice" of using 
Asterisk versus Asterisk with SER to handle SIP to SIP and SIP to PSTN 
calls.
Config:
1) SIP  
NAT - Internet -- SER - Internet - NAT  
SIP (i.e. no Asterisk in the picture for SIP to SIP 
calls)
2) PSTN (phone call) 
 Asterisk (via SIP) -- SER (i.e. any time traffic is 
to be sent/received to/from PSTN, use Asterisk as GW)

Both SER and 
ASTERISK are in the same LAN.
Anybody with 
experience in this type of set up willing to share war 
stories?
Would this 
configuration help those members of this list complaining Asterisk is dropping 
registered Budgetone/SIP phones?

Regards,
Uriel

RE: [Asterisk-Users] Re: DB virtualization for multiple database support - Was Re: [Asterisk-Users] How to use vmdb.sql in voicemail.conf/extension.conf

2003-10-07 Thread Uriel Carrasquilla

If you have the lattest PHP (4.1.3+), then, look for the PEAR subdirectory.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Garry Adkins
Sent: Monday, October 06, 2003 11:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: DB virtualization for multiple
database support - Was Re: [Asterisk-Users] How to use vmdb.sql in
voicemail.conf/extension.conf

Not familiar with it...  You have a URL?

- Original Message - 
From: Matteo Brancaleoni [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 05, 2003 4:52 PM
Subject: [Asterisk-Users] Re: DB virtualization for multiple database
support - Was Re: [Asterisk-Users] How to use vmdb.sql in
voicemail.conf/extension.conf

 Like what PEARS (php libs) do for db backends?

 Matteo.

 Garry Adkins wrote:

 I am trying a scenerio where the * will take the email and mailbox
  number from the Mysql database for sendming mail to a voicemail user. I
  have seen vmdb.sql file but is not able to determine its use.

You can't anymore MySQL was ripped from Asterisk because the client
libs
are GPL.

  I would be more than happy to help write a DB Virtualization function
  for *.

  I *love* the way it works in Java, but that's not a real possibility.
  It wouldn't need to be as complicated as JDBC but it's a nice model.

  We could however:
  1)  Abstract out the schema from the database calls
  2)  Develop a pluggable driver interface to translate to whatever DB
  you're using.

  This way...  You want MySQL, you develop a translation driver that
  maps * db calls to MySql.  (fairly trivial)
  Same for Postgres  (I'd suggest making this the default, as no GPL
  issues for mark, etc.)
  Same for Oracle, etc.

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] the g729 situation

2003-10-03 Thread Uriel Carrasquilla

If we avoid g729, what options do we have for SIP/Budgetone/Grandstream when
communicating to * via the Internet and still have something comparable to
GSM?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter
Sent: Thursday, October 02, 2003 2:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] the g729 situation

 LDM == Louis-David Mitterrand [EMAIL PROTECTED] writes:
 LDM Having purchased a license for 5 g729 channels on Digium's web
 LDM shop I thought registration and installation would be a snap. NOT.

 LDM I followed registration instructions to the letter but it failed
 LDM with that message:

 LDM ERROR! Your Internet connection is probably behind a proxy and the
 LDM Registration program can't communicate with our server

 LDM Which is stupid as my * box is a firewall and sits directly on the
 LDM Internet whith no restrictions from in-out.

I must say I'm impressed that people are brave enough to (1) accept the
long, restrictive and sometimes outright scary (did you read the parts
about credit card charges, or the definition of G.729 software in
connection with Improvement by Licensee?) licensing agreement and
(2) run a binary module that touches strange parts of the machine and
communicates that information over the network to a third party.

I also feel sorry for Digium, because they have to take the heat from
unhappy users.

IMHO this codec should be avoided at all cost.

--J.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Help with GPL license of Asterisk

2003-10-03 Thread Uriel Carrasquilla

So, is Astrisk being changed to an OSI-compliant license without the
anti-patent clause?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter
Sent: Thursday, October 02, 2003 2:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help with GPL license of Asterisk

 Mark == Mark Spencer [EMAIL PROTECTED] writes:
[...]
 Mark No problem, it's easy to get confused :) I would, however, take
 Mark issue with the GPL being evil.  It's not my *ideal* license,
 Mark but it certainly is good enough.

Just for the reference, while we're at it. GPL does have an issue, which
can cause problems to some people or companies. It is often overlooked,
because the open source issues seem much more controversial.

Having worked with GPL software quite a bit, also in the commercial
world, and having gotten some legal advice, I believe that the
anti-patent clauses in the GPL and LGPL are quite possibly the biggest
problem preventing the use of GPL'd software by commercial entities,
much bigger than the pass on the source and the rights requirement.

An excerpt from the GPL:

 7. If, as a consequence of a court judgment or allegation of patent
   infringement or for any other reason (not limited to patent issues),
   conditions are imposed on you (whether by court order, agreement or
   otherwise) that contradict the conditions of this License, they do not
   excuse you from the conditions of this License.  If you cannot
   distribute so as to satisfy simultaneously your obligations under this
   License and any other pertinent obligations, then as a consequence you
   may not distribute the Program at all.  For example, if a patent
   license would not permit royalty-free redistribution of the Program by
   all those who receive copies directly or indirectly through you, then
   the only way you could satisfy both it and this License would be to
   refrain entirely from distribution of the Program.
 [...]
 8. If the distribution and/or use of the Program is restricted in
   certain countries either by patents or by copyrighted interfaces, the
   original copyright holder who places the Program under this License
   may add an explicit geographical distribution limitation excluding
   those countries, so that distribution is permitted only in or among
   countries not thus excluded.  In such case, this License incorporates
   the limitation as if written in the body of this License.

As I understand it (and as my legal counsel advises me) this effectively
means that if I distribute GPL/LGPL code, I have to make sure that its
distribution and re-distribution is not restricted by patents (or other
restrictions).

If the code in question contains parts which some patents lay claim to,
restricting distribution, then I must not distribute the code at
all. Furthermore, by distributing the code I breach the GPL and expose
myself to legal threat of a lawsuit from the FSF.

It is needless to mention that it is impossible to me to verify that no
patents (worldwide!) lay claim to the code I'm distributing and impose
restrictions upon its distribution. Sooner or later I'm going to find
out that I do not comply with the GPL, because I distribute GPLd code
even though there are patent restrictions that apply to it.

An example of a particularly clear case of this problem is the XviD code
(http://www.xvid.org/), which is GPL-licensed. It seems to me that the
authors (copyright holders, to be precise) may distribute the software
under any license they choose, but nobody else is allowed to
re-distribute it, because they would be violating section 7 of the GPL,
as the MPEG-4 compression is (in some countries) covered by patents
requiring royalties to be paid.

This is an issue which is very often overlooked in the hot GPL
debates. However, in the commercial world, it is possibly the most
important one.

Conclusion (IMHO of course): if you have the choice, use a license that
is OSI-compliant but does not have the anti-patent clause. Or has it
phrased differently.

--J.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Grandstream Phone Issue

2003-09-30 Thread Uriel Carrasquilla

Are the SIP phones behind a NAT/Router? is * behind a NAT or Firewall?
can we see your sip.conf file?
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kevin
Sent: Tuesday, September 30, 2003 8:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Phone Issue

When I dial with my Grandstream 101 telephone to another sip phone or
Zap FXS, the call rings, but no audio is passed.  Eventually the call
gets disconnected.  The same thing happens if I dial the Grandstream.  

Any Suggestions?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR Web Search Frontend

2003-09-29 Thread Uriel Carrasquilla

How about including VoiceMail viewer/retriever.
URiel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Evans
Sent: Monday, September 29, 2003 5:37 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CDR Web Search Frontend

 Or do something really smart like the Perl guys and have a 
 backend-mostly-independent DB infrastructure.  Hell I think that PHP 
 finally smartened up and went this way, too.

Hi Guys

I am happy to do this and send the code back. Database independence
isn't to hard to achieve. It would be nice if a group of us could get
together and discuss how we can make this great app even better and
possibly look at getting a small team together to merge this and
phpconfig into a single application. Will possible access to cvs for the
developers.

Thoughts?

Mark Evans
SiteTel

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR Web Search Frontend

2003-09-29 Thread Uriel Carrasquilla

Why couldn't we just use PEAR?
it is there, it works and it does provide the abstraction layer for any of
the RDMBS systems discussed so far.
URiel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Evans
Sent: Monday, September 29, 2003 11:10 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CDR Web Search Frontend

Hi All

I am happy to go with SF. Jamie do you want to apply for an account

What I was thinking for the DB layer is the following

We have a main class which contains a basic DB implementation. We then
create subclasses which extend the main class for each DB we want
supported.

This code would be released under the GPL getting rid of any licencing
issues that might arise.

Thoughts?

I am happy to get a basic DB abstraction layer ready for people to look
at if its agreed this is a goer.

 BTW, I'm not a coder, I'm just an idea man.

BTW I am a coder and rarely get any good ideas :)

Regards

Mark

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR Web Search Frontend

Jamie:
thank you.  I will install later today.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl
Sent: Saturday, September 27, 2003 11:17 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] CDR Web Search Frontend

*This message was transferred with a trial version of CommuniGate(tm) Pro*

It's in the archives.  People on this list usually don't take too well
to repeating stuff. :)
(i'm not fussed tho)

http://asterisk.jazz-inc.net

Yes, the source is of course available for download.
:)

Enjoy!

J

 -Original Message-
 From: Uriel Carrasquilla [mailto:[EMAIL PROTECTED]
 Sent: Sunday, 28 September 2003 1:05 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] CDR Web Search Frontend

 *This message was transferred with a trial version of 
 CommuniGate(tm) Pro*
 does it include the source in PhP?
 what was the link again please?
 Uriel

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl
 Sent: Saturday, September 27, 2003 3:42 AM
 To: Asterisk Users (E-mail); Asterisk Dev (E-mail)
 Subject: RE: [Asterisk-Users] CDR Web Search Frontend

 *This message was transferred with a trial version of 
 CommuniGate(tm) Pro*

 Hey all,

 New versions available.  Now written in PHP with totals for Billing
 Seconds and Duration.

 Help yourselves and please send me more suggestions!!!
 Thanx!

 J

  -Original Message-
  From: Dimitri Bellini [mailto:[EMAIL PROTECTED]
  Sent: Friday, 26 September 2003 10:40 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] CDR Web Search Frontend

  *This message was transferred with a trial version of 
  CommuniGate(tm) Pro*
  Hi Carl
  i see web frontend i action is very good!! The total 
  time at end is good 
  thing.
  Thanks for great work. Can you put the script in some place 
  to download. 

  Dimitri

   *This message was transferred with a trial version of 
  CommuniGate(tm) Pro*

   Hey all,

   I've just done a quick (but functional) web front end for 
  searching the
   CDRs in a MySQL database.  Anyone interested in trying it 
 out?  I'm
   wondering what to add to it next.

   So far you can seach using source, destination, CLI, 
  channel and date
   ranges.  It also displays ALL fields in the database table.

   If interested, email me on [EMAIL PROTECTED]  Do not reply 
  directly to
   this email, it will bounce.  Depending on the level of 
  interest, I may
   post this somewhere for your free downloading pleasure.

   Regards,

   Jamie Carl
   Jazz Inc.
   http://www.jazz-inc.net
   Email: [EMAIL PROTECTED]
   JID: [EMAIL PROTECTED]
   Phone: +61-414-365466

   ___
   Asterisk-Users mailing list
   [EMAIL PROTECTED]
   http://lists.digium.com/mailman/listinfo/asterisk-users

  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] how stable is dynextendb

Jeremy:
where can I find retrieve_extensions_from_mysql.pl?
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Saturday, September 27, 2003 9:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] how stable is dynextendb




 I'm looking for a way to manage large dial plans.

 Blitz on IRC mentioned  DynExtenDB

 I'm wondering how stable it is since its not been
 updated since 2002-12-15


 Any other ideas ??
 I want to have my dial plan in a SQL database


 I actually just stumbled upon this today, and looks very interesting
 and useful.  Is anyone actually using this to do what John is looking
 to do?


IMHO, DynExtenDB is the absolute wrong way to deal with Extensions from
a database.Have you actually looked in the asterisk source tree?
See retrieve_extensions_from_mysql.pl. Granted it is not perfect or
quite how I would do it, but it does get the job done without any craziness.


Jeremy McNamara






___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] NAT/SIP solution?

I might be putting words into Stig's message but I think what he means to
ask was the following scenario that causes problems:
SIP --- NAT --- Internet --- NAT --- Asterisk
Nikotel has a solution and one participant in thi list is doing a trial on a
SIP/NAT router (claiming to be the first one in this realm).
To answer Stig's question as I understand it:
I don't think anybody is working on a solution in this list since the
by-pass is to put Asterisk directly on the Internet with its own public-IP
address.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: Sunday, September 28, 2003 2:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT/SIP solution?


Stig Hess wrote:

 Greetings,

 I was wondering if somebody is working on a solution to the
 NAT/SIP-issues? It seems to me that the problem has been identified,
 is that correct?

 Just hoping that someone with more skills will provide us with a
 solution sooner or later...

 Regards,

 Stig

The solution is to use nat=yes in your sip.conf.. so far this has
worked great for me..

Later..

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] NAT/SIP solution?

I had been warned about British sense of humour, but this even a South
American like myself find funny.

Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: Sunday, September 28, 2003 3:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT/SIP solution?

Stig Hess wrote:

 I meant where Asterisk is behing a NAT... sorry for the confusion.

 Regards,

 Stig H.

Oh.. :)

Well thats a bigger problem.. and i doubt the Gods of SIP are going to
fix it any time soon.. :(

Later..

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] the g729 situation

I am using SIP/GrandStream connecting to Asterisk over the Internet.  I
prefer not to use g729 and uLaw/aLaw uses too much bandwidth.  What would be
the next logical choice for codec (given that GSM is not supported) and be
comparable to GSM?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Friday, September 26, 2003 5:04 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] the g729 situation


NuFone only had 3 G.729 licenses and when we went to add more it blew up
our system and now we have none.  Anyways, we are not very fond of the
VoiceAge licensing terms.  We prefer iLBC.


Jeremy McNamara





Thomas Moghnie wrote:

 Hi,

 On the same note, I am having a problem with G.729, having 4 *
 asterisk boxes 2 with 10 licenses and one with 2 licenses.

 The licenses installs fine, but the codec doesn't work as supposed to
 be. In path thru situation, where a UA (grandstream phone) is talking
 to the * that is connected to NuFone over IAX/2 seems to work. But
 when NuFone stopped supporting G729. The RTP path could not be
 established (G729-*-SPEEX). However, the following scenario works
 (G711/GSM-*-SPEEX)

 Thanks for your help

 */Mark Spencer [EMAIL PROTECTED]/* wrote:

  Having purchased a license for 5 g729 channels on Digium's web
 shop I
  thought registration and installation would be a snap. NOT.
 
  I followed registration instructions to the letter but it failed
 with
  that message:
 
  ERROR! Your Internet connection is probably behind a proxy and the
  Registration program can't communicate with our server

 You can call us for free support on G.729 if you purchased it from us.
 877-LINUX-ME just choose install support.

  Now I wrote to vonage as per the instructions further in the error
  message, requesting a certificate. I'm sure I'm not the only one
 going
  through all these hoops.

 I trust you mean Voiceage not Vonage but in any case neither will
 likely be useful. Definitely should contact us directly.

  - is there a cracked g729 b inary out there? (which I plan to
 use inside
  my license agreement)

 Not as far as I know.

  - is it true that * has to be run with -c when using g729 ?

 Yes, again we're trying to get Voiceage to fix the issue, but
 working with
 closed source, slow moving, intellectual property based vendors is
 generally a pretty miserable experience.

 Mark

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

 
 Do you Yahoo!?
 The New Yahoo! Shopping

http://shopping.yahoo.com/?__yltc=s%3A15443%2Cd%3A22708228%2Cslk%3Atext
%2Csec%3Amail
 - with improved product search



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR Web Search Frontend

does it include the source in PhP?
what was the link again please?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jamie Carl
Sent: Saturday, September 27, 2003 3:42 AM
To: Asterisk Users (E-mail); Asterisk Dev (E-mail)
Subject: RE: [Asterisk-Users] CDR Web Search Frontend

*This message was transferred with a trial version of CommuniGate(tm) Pro*

Hey all,

New versions available.  Now written in PHP with totals for Billing
Seconds and Duration.

Help yourselves and please send me more suggestions!!!
Thanx!

J

 -Original Message-
 From: Dimitri Bellini [mailto:[EMAIL PROTECTED]
 Sent: Friday, 26 September 2003 10:40 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] CDR Web Search Frontend

 *This message was transferred with a trial version of 
 CommuniGate(tm) Pro*
 Hi Carl
   i see web frontend i action is very good!! The total 
 time at end is good 
 thing.
 Thanks for great work. Can you put the script in some place 
 to download. 

 Dimitri

  *This message was transferred with a trial version of 
 CommuniGate(tm) Pro*

  Hey all,

  I've just done a quick (but functional) web front end for 
 searching the
  CDRs in a MySQL database.  Anyone interested in trying it out?  I'm
  wondering what to add to it next.

  So far you can seach using source, destination, CLI, 
 channel and date
  ranges.  It also displays ALL fields in the database table.

  If interested, email me on [EMAIL PROTECTED]  Do not reply 
 directly to
  this email, it will bounce.  Depending on the level of 
 interest, I may
  post this somewhere for your free downloading pleasure.

  Regards,

  Jamie Carl
  Jazz Inc.
  http://www.jazz-inc.net
  Email: [EMAIL PROTECTED]
  JID: [EMAIL PROTECTED]
  Phone: +61-414-365466

  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Follow Me

suggestion:
get call forwarding on your POTS line.  Then when the call comes-in, flash
and forward to your cell.
URiel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ben Wern
Sent: Friday, September 26, 2003 5:36 PM
To: Ernest W. Lessenger; [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Follow Me

Ernest,

Again, I really appreciate your help with this. Your solution looks like it
requires two POTS lines -- am I misreading it? My goal is to have a call
come in on a single POTS line and then have Asterisk try to track me down
via the same POTS line (3 way calling.)

Ben

At 12:30 PM 9/17/2003 -0700, Ernest W. Lessenger wrote:
At 06:48 PM 9/16/2003, you wrote:
cell phone into the call (or my office number, etc.) I understand the
selected numbers part of it, but not how to get it to use the three way.
If
I send it to Nufone first, I'm paying for a call to a local number (my
cell) that I don't need to.

This should work...

[default]
exten = s,1,Dial(Zap/3,20,t) ; This is your desk phone
exten = s,2,Dial(Zap/2/1234567,20,t) ; This is your secondary POTS line
calling your office
exten = s,3,Dial(Zap/2/3217654,20,t) ; This is your secondary POTS line
calling your cell phone
; I've never tried this one coming up, but I think it's worth a shot as it
works just fine for local extensions
exten = s,4,Dial(Zap/2/3217654Zap/3/3217654,20,t) ; This is your
secondary and tertiary POTS lines calling your cell phone anbd office

As long as none of these lines go to voicemail, they should fail over
properly in order. You can also make it more complicated with time-based
includes and gotos.

--Ernest

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CDR Web Search Frontend

Yes, thru PEAR you can make PhP independent of the DB (i.e. RDBMS
abstraction layer).
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Saturday, September 27, 2003 8:14 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CDR Web Search Frontend

 Since * and MySQL have had a licensing scuffle, is there a way to set it
 up so that we can specify wether or not it's in the mysql database, or
 use the plaintext file that * generates with cdr_csv.so?

Or do something really smart like the Perl guys and have a
backend-mostly-independent DB infrastructure.  Hell I think that PHP
finally smartened up and went this way, too.

Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Budgetone + NAT: Firmware Version?

Is there anyway to prevent the BudgetTone from just doing a BIOS upgrade
without consulting?
I would be scared if my PC upgraded its BIOS at will.
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Saturday, September 27, 2003 8:11 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Budgetone + NAT: Firmware Version?


 All I can suggest is to just go over your settings and just make sure
 that they are all correct and that nothing was accidentally changed
 during the upgrade..

I'd go one further than that and have all the phones go to factory default,
and then re-set all the parameters to what you want them to be.  Perhaps
locations changed in the NVRAM and while it seems to look like it's
configured right, it isn't.

Think of it as a PC BIOS upgrade -- you must do a factory default reset and
then set your parameters again or you might end up with mostly working but
infuriating little problems.

Regards,
Andrew
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] how stable is dynextendb

I spent a lot of time on Dynextendb and made it work (I am actually still
using it at home with 3 POTS and 4 extensions) after extensive modifications
to the original source code.  However, it does not scale and I am about to
pull the plug.  It is not worth it but it was an interesting exercise.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Brown
(CV)
Sent: Saturday, September 27, 2003 6:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] how stable is dynextendb


I'm looking for a way to manage large dial plans.

Blitz on IRC mentioned  DynExtenDB

I'm wondering how stable it is since its not been
updated since 2002-12-15


Any other ideas ??

I want to have my dial plan in a SQL database


thanks



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP/ Grandstream Issues

Try on the Grandstream DTMF via INFO.
Also use uLaw for codec.
If behind the NAT just say NAT=YES and REINVITE=NO.
It works like a champ.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lists
Sent: Saturday, September 27, 2003 7:01 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP/ Grandstream Issues



I just got a grandstream SIP phone

Here is my sip.conf for the phone

[mlh]
type=friend
insecure=yes
username=mlh
secret=mlh
host=dynamic
canreinvite=no

The phone as the default config on it.


If I use the phone to call a Zap interface (a tdm card) the voice sounds 
all choppy.

If I use the phone to call a x100p card, it does not dial what I dial (no 
DTMF)

I don't know what else to try.should I change the vocoder (it is on 
PCMU at the momemnt)

I am using the phone on a LAN so bandwidth is not an issue.

Any Help would be great,

Michael

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] RTP routing..

2003-09-26 Thread Uriel Carrasquilla

rtp.conf?
what does it do?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Low, Adam
Sent: Friday, September 26, 2003 8:47 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] RTP routing..

I can restrict the RTP ports used with my Cisco 79xx phones and on my Cisco
AS5300 and I think you can with Asterisk by using the rtp.conf but I'm not
completely sure, I'd suggest diving into the source for that one ...

 -Original Message-
 From: Andre Lomonaco [mailto:[EMAIL PROTECTED]
 Sent: 26 September 2003 14:31
 To: '[EMAIL PROTECTED]'
 Subject: RES: [Asterisk-Users] RTP routing..

 Hi,

 Sorry for my bad english but I´ll try to explain my problem

 I got an Asterisk running in my house with ADSL...
 I´m using X100P and TDM400P cards

 My intention is get calls via PSTN to my house and
 Redirect to my computer in my work using X-Lite by SIP...

 Here´s the map with Firewalls

 Call for anyone to my house = PSTN = X100P = EXTENSIONS =
 SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE

 It´s working very nice, but I had to disable iptables in my
 Asterisk Box(Home)...

 I was using my linux with PPPoe Client, DynamicDnsClient and
 IPTABLES...

 I´d like to know if is possible to using IPTABLES again.
 My stupid question is: Can I restrict the ports that Asterisk uses
 to transmit RTP.

 When I was using IPTABLES with only port 5060 open , the SIP
 registration
 works nice but I didn´t receive sound...

   Andre Lomonaco

 -Mensagem original-
 De: Low, Adam [mailto:[EMAIL PROTECTED]
 Enviada em: Friday, September 26, 2003 9:06 AM
 Para: '[EMAIL PROTECTED]'
 Assunto: RE: [Asterisk-Users] RTP routing..

 WipeOut,

 I just started to whiteboard this and had some realisations/questions:

 1. I guess/hope your ADSL connection is not NAT'd ?
 2. You will need two NIC's as I assume you will have two
 separate next hop
 gateways with each ADSL connection!
 3. How would you load balance the inbound calls over the two
 connections
 (ensuring each doesn't exceed capacity)?

 The more I think about this the more I feel that a better
 solution would be
 to place a router between the Asterisk server and the two
 ADSL modems with
 some kind of NAT setup ...

 Adam

 * DISCLAIMER *

 This message and any attachment are confidential and may be
 privileged or
 otherwise protected from disclosure and may include
 proprietary information.
 If you are not the intended recipient, please telephone or
 email the sender
 and delete this message and any attachment from your system.
 If you are not
 the intended recipient you must not copy this message or attachment or
 disclose the contents to any other person

 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

* DISCLAIMER *

This message and any attachment are confidential and may be privileged or
otherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
disclose the contents to any other person

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

FW: RE: [Asterisk-Users] AntiSpam UOL




Every 
time I send an e-mail to the * list, I receive this "AntiSpam UOL" E-mail. 
is anybody else experiencing the same?
How 
can I get rid of it?
Uriel
-Original Message-From: AntiSpam UOL 
[mailto:[EMAIL PROTECTED]Sent: Wednesday, September 24, 
2003 11:51 PMTo: [EMAIL PROTECTED]Subject: RE:RE: 
[Asterisk-Users] SIP / GrandStream Configuration

  
  



  

  


  
  Olá,Você enviou uma mensagem para 
[EMAIL PROTECTED]Para que sua mensagem seja 
encaminhada, por favor, clique aqui
  

  Esta confirmação é necessária porque 
[EMAIL PROTECTED] usa o Antispam UOL, um programa que 
elimina mensagens enviadas por robôs, como pornografia, propaganda e 
correntes.As próximas mensagens enviadas para 
[EMAIL PROTECTED] não precisarão ser 
confirmadas*.*Caso você receba outro pedido de confirmação, por favor, 
peça para [EMAIL PROTECTED] incluí-lo em sua lista de 
autorizados.

  
  
Atenção! Se você não 
  conseguir clicar no atalho acima, acesse este 
  endereço:http://tira-teima.as.uol.com.br/challengeSender.html?data="">
  

  

  

  


  
  Hi,You´ve just sent a message to 
[EMAIL PROTECTED]In order to confirm the sent 
message, please click here
  

  This confirmation is necessary because 
[EMAIL PROTECTED] uses Antispam UOL, a service that 
avoids unwanted messages like advertising, pornography, viruses, and 
spams.Other messages sent to 
[EMAIL PROTECTED] won't need to be 
confirmed*.*If 
you receive another confirmation request, please ask 
[EMAIL PROTECTED] to include you in his/her authorized e-mail 
list.

  
  
Warning! If the link 
  doesn´t work, please copy the address below and paste it on 
  your 
  browser:http://tira-teima.as.uol.com.br/challengeSender.html?data="">
  
Use o AntiSpam UOL e proteja sua caixa 
  postal

RE: [Asterisk-Users] SIP / GrandStream Configuration




Michael:
could 
you share how you configured your GrandStream? for example, did you say 
"yes" to NAT (without a STUN)?
how 
about in SIP.CONF, how did you configure the remote 
GrandStream?
Regards,
Uriel

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Michael 
  KoehlerSent: Thursday, September 25, 2003 10:42 AMTo: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / 
  GrandStream ConfigurationSorry, but my * is behind NAT 
  and i have no problems with SIP, and it even works with NAT to NAT and without 
  forwarding ports or similar effords.MichaelStephen 
  Varga wrote:
  On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote:
  
Adam:
in reference to my first message, the NAT on the SIP/GS (a D-Link router)
has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
forwarded to the Sip/GS.
The Asterisk server, also behind another NAT (Linksys), has the same ports
opened and forwarded.
is it still impossible?
URiel

Nope, it is not currently possible. * behind a NAT for SIP does not work
because the * real IP address is placed in the SDP information,
therefore the 'outside' phone can not send the media stream to *. See my
answers over the last week for the more details and possible work
arounds.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / GrandStream Configuration




Michael:
is it 
a D-link on both NAT? the one for * and the one for the Grand 
Stream?
Uriel

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Michael 
  KoehlerSent: Thursday, September 25, 2003 12:55 PMTo: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / 
  GrandStream ConfigurationA plain wireless dlink dsl 
  router.Stephen Varga wrote:
  On Thu, 2003-09-25 at 10:42, Michael Koehler wrote:
  
Sorry, but my * is behind NAT and i have no problems with SIP, and it
even works with NAT to NAT and without forwarding ports or similar
effords.


Michael


What kinda box/device is doing the NAT? 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / GrandStream Configuration




Michael:
I am 
working in a second language and I might be loosing some subtle points. 
Please over communicate to make your points.
are 
you saying that two garden variety D-Link NAT routers working on two ends of the 
Interent with one end running a SIP/GrandStream IP-Phone and the other running * 
will work?
This 
is where Stephen stated that it will NOT. You seem to be saying it will 
work.
Is 
Nikotel doing anything special that allows them to work this type of 
configuration?
Please 
elaborate.
Regards,
Uriel

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Michael 
  KoehlerSent: Thursday, September 25, 2003 3:41 PMTo: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / 
  GrandStream ConfigurationIt is not a feature of the 
  router, it is the way SIP is handled with nikotel.comI recently wrote 
  that i'm using just a plain router with my natted asterisk because "Stephen 
  Varga" wrote that SIP behindNAT (in relation to asterisk) is impossible. 
  It is possible because i'm using asterisk this way.There is also 
  nothing special to setup with the router for nikotel and NAT, except you have 
  a firewall and needstraight rules, then you may use port 
  forwarding.MichaelStephen Varga wrote:
  On Thu, 2003-09-25 at 12:54, Michael Koehler wrote:
  
A plain wireless dlink dsl router.

Do you know the model number and the software version? 

I am trying to understand how it is making the appropriate adjustments
to allow the connection to work.

Thanks,
Steve

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / GrandStream Configuration

What we need are the nuclear scientists at Nikotel sharing their solution.
I am wondering if they are using a Linux/NAT.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga
Sent: Thursday, September 25, 2003 4:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration

On Thu, 2003-09-25 at 15:41, Michael Koehler wrote:
 It is not a feature of the router, it is the way SIP is handled with
 nikotel.com

 I recently wrote that i'm using just a plain router with my natted
 asterisk because Stephen Varga wrote that SIP behind
 NAT (in relation to asterisk) is impossible. It is possible because
 i'm using asterisk this way.

 There is also nothing special to setup with the router for nikotel and
 NAT, except you have a firewall and need
 straight rules, then you may use port forwarding.

Ok maybe I was being to broad in my original statement, so let me
clarify.

There orginal question was does the scenario

 SIP Phone --- NAT --- Internet --- NAT --- Asterisk 

work.

In general this can not be easily accomplished, because of the real ip
address of the devices get embedded in SDP message during the INVITE
process. Most phones can be changed to use the NAT address in this
process, so this solves one side of the conversation. However I have not
found away to do this in the asterisk software, thus SDP message needs
modified to change the ip address to the NATed one outside of * for this
to work. For this I have not discovered a reasonable solution.

In Mike's case, I am guessing the SDP message is being modified when the
packet arrives at the Nikotel's gateway. Which makes this a specialized
case.

So that still leaves us with a general problem of SIP and NATing on both
sides, for the rest of us not having the benefit of the software that
nikotel is using to make this scenario work.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP / GrandStream Configuration




Hi 
there!
I installed the 
BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it 
to another location using a D-Link NAT.
I opened 5060 (SIP) 
and 5000 to 5008 for RTP. I also fixed the IP address of the 
BudgetTone.
When I receive a 
call on my Asterisk, it would ring my FXS as before. However, after I pick 
up, it hangs within a few seconds (Hungup Zap1-1 in the 
log).
The configuration 
I have in * is the following:
sip.conf
---
[general]
port=5060
context=sip
maxexpirey=3600
defaultexpirey=60
disallow=all
allow=ulaw
allow=gsm
[1000]
contet=sip
type=friend
username=1000
secret=? 
(not the real one)
host=dynamic
mailbox=1000
canreinvite=yes
dtmfmode=rfc2833

I did not change the 
above configuration when I moved the budgetTone from the LAN to the Internet 
(Wan).
I am not using a 
"register" statement in the sip.conf and I am wondering if I need 
to.
I did change the sip 
server IP address in the Grandstream configuration.

I suspect my problem 
is with the router (NAT). I don't quite understand the symetric 
discussions but I downloaded a paper to learn more. Right now, all my 
public and private ports are the same.

Regards,
Uriel

RE: [Asterisk-Users] SIP / GrandStream Configuration

Very valuable help.  It is now working like a champ.

This is a solution with SIP--NAT---Internet---Asterisk.  No problems here.

What I would like to do next is to move Asterisk behind a NAT as follows
SIP---NAT---Internet---NAT---Asterisk
do I need a STUN server? is there a chance this could work?
The Google results seems to indicate that I will get an ulcer attempting
this step.  is that true?

Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: Wednesday, September 24, 2003 9:05 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration


Try adding nat=yes to your config..

Also if you want to make SIP to SIP extension calls and don't want to fight
with the NAT set canreinvite=yes to canreinvite=no..

Finally set dtmfmode=info for the GS phones..

Later..

 Hi there!
 I installed the BudgetTone (GrandStream) on my LAN without any problems.
 Then, I moved it to another location using a D-Link NAT.
 I opened 5060 (SIP) and 5000 to 5008 for RTP.  I also fixed the IP address
 of the BudgetTone.
 When I receive a call on my Asterisk, it would ring my FXS as before.
 However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in
 the log).
 The configuration I  have in * is the following:
 sip.conf
 ---
 [general]
 port=5060
 context=sip
 maxexpirey=3600
 defaultexpirey=60
 disallow=all
 allow=ulaw
 allow=gsm
 [1000]
 contet=sip
 type=friend
 username=1000
 secret=?  (not the real one)
 host=dynamic
 mailbox=1000
 canreinvite=yes
 dtmfmode=rfc2833

 I did not change the above configuration when I moved the budgetTone from
 the LAN to the Internet (Wan).
 I am not using a register statement in the sip.conf and I am wondering
if
 I need to.
 I did change the sip server IP address in the Grandstream configuration.

 I suspect my problem is with the router (NAT).  I don't quite understand
the
 symetric discussions but I downloaded a paper to learn more.  Right now,
all
 my public and private ports are the same.

 Regards,
 Uriel


--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr

Powered by Outblaze
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP / GrandStream Configuration

Adam:
in reference to my first message, the NAT on the SIP/GS (a D-Link router)
has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being
forwarded to the Sip/GS.
The Asterisk server, also behind another NAT (Linksys), has the same ports
opened and forwarded.
is it still impossible?
URiel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Hart
Sent: Wednesday, September 24, 2003 7:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration


How will the packets get to the asterisk server? You'd need to forward ports
on the NAT device, otherwise it's impossible

- Original Message -
From: Uriel Carrasquilla [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 25, 2003 9:48 AM
Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration


 Very valuable help.  It is now working like a champ.

 This is a solution with SIP--NAT---Internet---Asterisk.  No problems here.

 What I would like to do next is to move Asterisk behind a NAT as follows
 SIP---NAT---Internet---NAT---Asterisk
 do I need a STUN server? is there a chance this could work?
 The Google results seems to indicate that I will get an ulcer attempting
 this step.  is that true?

 Regards,
 Uriel

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
 Sent: Wednesday, September 24, 2003 9:05 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration


 Try adding nat=yes to your config..

 Also if you want to make SIP to SIP extension calls and don't want to
fight
 with the NAT set canreinvite=yes to canreinvite=no..

 Finally set dtmfmode=info for the GS phones..

 Later..

  Hi there!
  I installed the BudgetTone (GrandStream) on my LAN without any problems.
  Then, I moved it to another location using a D-Link NAT.
  I opened 5060 (SIP) and 5000 to 5008 for RTP.  I also fixed the IP
address
  of the BudgetTone.
  When I receive a call on my Asterisk, it would ring my FXS as before.
  However, after I pick up, it hangs within a few seconds (Hungup Zap1-1
in
  the log).
  The configuration I  have in * is the following:
  sip.conf
  ---
  [general]
  port=5060
  context=sip
  maxexpirey=3600
  defaultexpirey=60
  disallow=all
  allow=ulaw
  allow=gsm
  [1000]
  contet=sip
  type=friend
  username=1000
  secret=?  (not the real one)
  host=dynamic
  mailbox=1000
  canreinvite=yes
  dtmfmode=rfc2833
 
  I did not change the above configuration when I moved the budgetTone
from
  the LAN to the Internet (Wan).
  I am not using a register statement in the sip.conf and I am wondering
 if
  I need to.
  I did change the sip server IP address in the Grandstream configuration.
 
  I suspect my problem is with the router (NAT).  I don't quite understand
 the
  symetric discussions but I downloaded a paper to learn more.  Right now,
 all
  my public and private ports are the same.
 
  Regards,
  Uriel
 

 --
 __
 http://www.linuxmail.org/
 Now with e-mail forwarding for only US$5.95/yr

 Powered by Outblaze
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users


 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Check and restart script..

yes,
it is not cron but a daemon.  Iactually got the suggestion from this list.

You can get all the glory details from:
http://cr.yp.to/daemontools.html
Dr. Bernstein tools.
I have been using it with asterisk successfully for 4 months.
Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: Wednesday, September 24, 2003 11:02 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Check and restart script..


Has anyone written a script that can be used as a cron job or similar that
will test if Asterisk is running and if not restart it??

I have just had an issue where asterisk crashed and someone was trying to
call me.. it would be nice if it could have been automatically restarted..

I was thinking of a simple bash script something like running ps -aux |grep
asterisk and then some kind of if to say that if the result is nothing
then execute asterisk.. Problem with that theory is that the ps command
will show up as well so i will have to work out a way to drop that..

Of course I may be missing a simpler or far better solution so thats why I
am asking here first..

Later..
--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr

Powered by Outblaze
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] MY Sql CDR

I have a question regarding MySQL CDR's:
For a given extension I need to limit the number of minutes it can use in a
given week.
I was thinking about using the CDR information in the MySQL table to see the
usage for the week and then if exceeded, STOP the call and play a message.
does anybody have a suggestion on how to query the database so I don't have
to add-up all the minutes this particular extension have used during the
week?

Regards,
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Sunday, September 21, 2003 10:20 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] MY Sql CDR


Look here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cdr+mysql
/Olle

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] MY Sql CDR

Agree, I can run an AGI script after the outbound call.
But where do I invoke the AGI script?
it can't be in extensions.conf since, I believe, when either party hang-up,
the next priority is not invoked, or am I mistaken?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Crick
Sent: Sunday, September 21, 2003 8:33 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] MY Sql CDR


You could have an AGI script that runs after an outbound call to update a
running-total figure with the amount of either the last call or all calls to
date in the current period?

That way you're just checking a stored value before allowing/denying an
outbound call?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] RE: Very bad echo (appears that...)

You are kidding,I hope.
This typo would manifest itself as an echo problem?
May be the parser needs to put out a warning of some kind.
That is my 2cents.
URiel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk PBX
Sent: Sunday, September 21, 2003 8:39 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: Very bad echo (appears that...)


My partner found it!!

Problem solved...

The error was a syntax error in the zapata.conf

channel=1

Should have been written as:

channel=1



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] MY Sql CDR