[Asterisk-Users] dbget & dbput

[Marian Danisek]

Hi,

do i need some other software than asterisk to use database commands -
dbput and dbget in asterisk ? 

regards 

Marian


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Re: [Asterisk-Users] IAXTEL toll-free

[Mark Spencer]

It's being replaced now with a new Dell.  Details will be made available
soon.

Mark

On Wed, 9 Jul 2003, Paul Cheng wrote:

> Hi,
>
> Has anyone been able to place a call via IAXTEL toll-free termination
> lately? I had it working at one time, but now it doesn't seem to work
> anymore. www.iaxtel.com also appears dead. Is this the server problem
> again or is it my config? Haven't been able to find any references in
> the list.
>
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RE: [Asterisk-Users] TDM400P noise?

[Steven Critchfield]

Then why not just get a new powersupply? If it is a standard ATX
powersupply then it should be able to be replaced for $25-40.

On Tue, 2003-07-08 at 19:41, Kevin Herzig wrote:
> Thanks for the response, Steven.
> 
> I've tried various configurations of with / without x and/or
> framebuffer.  No difference.
> 
> I'm running on Debian Woody with a kernel I compiled from the kernel
> source.  Maybe I'll try to download the stock kernel source and compile
> that next.
> 
> I've tried unmasking interrupts from the hard drive, with and without
> apm.o installed, etc, etc...
> 
> It sounds to me like the same problem referenced here:
> 
> http://www.mail-archive.com/[EMAIL PROTECTED]/msg02434.htm
> l
> 
> The other end of the conversation hears the same noise I do.  The
> conversation is intelligible, but barely.
> 
> Too bad, I guess I'm going to have to get another machine to do this
> with.  I can get another but it's getting to be expensive. :)
> 
> Kevin
> 
> On Sat, 2003-07-05 at 21:18, Kevin Herzig wrote:
> > Hi all.  I bought Digium's dev kit and a used IBM PL300 PC to try it
> out
> > in.  The X100P works fine, but with the TDM400P I get what I can best
> > describe as 'interrupt noise'... noise whenever I type a key on the
> > keyboard, or when something accesses the disk drive, uses cpu, etc.
> 
> 
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Re: [Asterisk-Users] voip

[wasim]

On Tue, 8 Jul 2003, marrandy wrote:

> Well I now have asterisk installed.
> 
> I've printed out the asterisk web site.
> 
> I've printed the draft Asterisk handbook V2
> 
> I've printed the Introduction to the asterisk open source pbx

oh, no, the trees, the humanity... try and not print, its easier to cut
paste from a console than print the buggers out, besides the pace of *
development at times has been so fast, those printed pages are probably
obsolete by now :)

> 1)   VOIP inside the private LAN from one computer to another.  e.g. 
> 192.168.1.1 to 192.168.1.2 etc.

great... IAX2 will do that for ya

> 2)   VOIP to someone outside in the U.S.

ask nufone, or setup your own gateway

> 3)   VOIP to someone overseas e.g. U.K.

ask nufone or taan or someone like that

> 4)   Get a hardware card for the incoming line

get an X100P from digium

> 5)   Some extensions, perhap's Four (4).

get a TDM400P from digium (or use soft phones)

> I don't see much in relation to point 1.

why not? keep looking, it'll become clearer

> What software (linux) can be used to connect VOIP to the astericks server ??

sip|h323|iax|mgcp?|skinny?
a whole buncha codecs

- wasim
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[Asterisk-Users] voip

[marrandy]

Hello.

Well I now have asterisk installed.

I've printed out the asterisk web site.

I've printed the draft Asterisk handbook V2

I've printed the Introduction to the asterisk open source pbx


Because I'm experimenting, I would like to do things in a certain order :-

1)   VOIP inside the private LAN from one computer to another.  e.g. 
192.168.1.1 to 192.168.1.2 etc.

2)   VOIP to someone outside in the U.S.

3)   VOIP to someone overseas e.g. U.K.

4)   Get a hardware card for the incoming line

5)   Some extensions, perhap's Four (4).

I don't see much in relation to point 1.

What software (linux) can be used to connect VOIP to the astericks server ??

Regards...Martin

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Attention TDM400P users (was Re: [Asterisk-Users] PCI Master Abort)

[Mark Spencer]

I believe the PCI master abort issue may also have something to do with
power.  As some of you know, we are currently working to resolve power
issues on the TDM40B which have caused some people grief with poor audio
or the "PCI Master Abort" syndrome.  We (Wil and I) have hand modified a
couple of boards to try to cleanup the PC power supply and are very
pleased with the results (until just recently, we didn't have a machine
with the noise problem in our hands).

I believe our modifications represent a general panacea for a range of
power supply related issues including:

* PCI Master abort
* "Interrupt noise"
* Static
* Buzzing

Martin is making a list of people who have trouble with bad audio and/or
PCI master abort.  In the next couple of days, we would like to send some
boards to people that have these problems *in the U.S.* and who are
capable of installing and testing these cards in a *timely fashion*.  If
you are interested, please contact Martin (256-428-6161 or martinp) and be
sure he has you "on the list" and knows what your symptom is and what your
hardware configuration is.  Once we have confirmed these fixes work
properly, we will be spinning a revised version of the TDM400P board.

Once the TDM400P replacement boards are ready, anyone who wants to upgrade
will be able to exchange their old board for a new one at no cost (you
will only be responsible for shipping us back your old board).  If you
are having no trouble with the board, there will be no need to send it
back, although we will allow the boards to be exchanged for the forseeable
future (so long as we produce anything like the TDM400P) so there will be
no hurry in swapping out your board.

Mark

On Tue, 8 Jul 2003 [EMAIL PROTECTED] wrote:

> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont
> > Sent: Monday, July 07, 2003 4:15 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] PCI Master Abort
> >
> > I am always getting multiple PCI Master Abort messages when I try to
> > plug in a second TDM400P.
> > I have asked this question before, but I nothing really solved my
> > problem and I just put it on the back burner for a while.
> > I am at a point where this is a crucial issue.
>
> I do have the same PCI Master abort message with a Wildcard S400P
>
> It seems this is NOT an IRQ problem :
> I did change the IRQ in the BIOS :
> - manual assignation for all the PCI boards
> - automatic assignation for the Wildcard
> then verificate the status of the PCI devices :
> - more /proc/pci
> - just to list the IRQ really assigned, the memory I/O addresses, ...
> All seems to be correct
>
> Then launch the modules :
> modprobe zaptel
> modprobe wcfxo
> modprobe wcfxs
>
> But here the error message still appears !
>
> I'm interested in your solution if you solve the pb !
>
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RE: [Asterisk-Users] TDM400P noise?

[Kevin Herzig]

Thanks for the response, Steven.

I've tried various configurations of with / without x and/or
framebuffer.  No difference.

I'm running on Debian Woody with a kernel I compiled from the kernel
source.  Maybe I'll try to download the stock kernel source and compile
that next.

I've tried unmasking interrupts from the hard drive, with and without
apm.o installed, etc, etc...

It sounds to me like the same problem referenced here:

http://www.mail-archive.com/[EMAIL PROTECTED]/msg02434.htm
l

The other end of the conversation hears the same noise I do.  The
conversation is intelligible, but barely.

Too bad, I guess I'm going to have to get another machine to do this
with.  I can get another but it's getting to be expensive. :)

Kevin

On Sat, 2003-07-05 at 21:18, Kevin Herzig wrote:
> Hi all.  I bought Digium's dev kit and a used IBM PL300 PC to try it
out
> in.  The X100P works fine, but with the TDM400P I get what I can best
> describe as 'interrupt noise'... noise whenever I type a key on the
> keyboard, or when something accesses the disk drive, uses cpu, etc.


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Re: [Asterisk-Users] Need a recommendation on a good motherboard/processor combination

[Mathew Frank]

I seem to remember you sent this the other day?  If so I`m guessing you
didn`t get much replies, so...

>I need a recommendation on a good motherboard/processor combination.
>I would like a motherboard that has lots of PCI slots and works well with
>Asterisk without problems getting drivers working, etc.  Onboard LAN
>would be nice to keep from using a slot.  Plan to use RedHat 8 for the OS.

I, and a reseller/consultant friend of mine, used to use ASUS a lot due to
reliability. (note:  I still have a working Asus P100 board running as a
firewall here - probably run for a few years more fault free - they really
used to be fantastic)  In the last few years I personally have had way too
much hassle with faulty boards (DOAs, firmware problems that never got fixed
and intermittant faults specifically for me), and so did this friend of mine
with his custumers, so we both will never touch them with a barge pole.
Maybe they do score well on TH for speed, but I`m in business - I need
reliability as a priority.

I personally have and still do use Gigabyte and have never had a problem -
though the way they have implemented "dual bios" defeats itself. (meaning a
virus could concievably blow both bios chips away because they are both
writable, and nvram is shared between the two)  The last ASUS board I ever
bought was extremely unreliable (strange intermittant boot problems among
other things) so I replaced it with a Gigiabyte board with the same (VIA
333) chipset and it was solid as a rock (still using it now in fact)
Gigabyte have a lot of boards with 5+ PCI slots, too.

My aforementioned reseller friend is a fan of Intel boards due to
reliability, just good design, and a _very_ good returns policy (very
important when you are reselling to businesses)

Well that`s how I see it - I`m sure others will dispute my claims, but they
are purely experience-based.

Hope this helps.

Cheers,
Mathew


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[Asterisk-Users] RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs

[Alex Lopez]

I asked on the IRC channel last night and was told the IAXTEL had been
down for a few months now.  It had a very poor uptime.. Maybe someone
can tell us why the uptime was so poor.

Alex


Message: 9
Date: Wed, 9 Jul 2003 01:05:00 +0200
From: Paul Cheng <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAXTEL toll-free
Reply-To: [EMAIL PROTECTED]

Hi,

Has anyone been able to place a call via IAXTEL toll-free
termination 
lately? I had it working at one time, but now it doesn't seem to
work 
anymore. www.iaxtel.com also appears dead. Is this the server
problem 
again or is it my config? Haven't been able to find any
references in 
the list.



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Re: [Asterisk-Users] Call Accounting

[Steven Critchfield]

You actually need to do

exten => _9NXXNXX,1,dial(Zap/1/${ETEN:1}) ; local
exten => _91NXXNXX,1,dial(Zap/1/${EXTEN:1}) ; LD

If you read the documentation and followed the examples you would
understand that the underscore starts a pattern, N is for digits 2-9, X
is for 0-9. 

On Tue, 2003-07-08 at 14:55, Erik Kendall wrote:
> Matteo,
> 
> Thank you.  I'll give it a try.
> 
> Erik
> 
> --- Brancaleoni Matteo <[EMAIL PROTECTED]> wrote:
> > That's the reason why no number is logged, since
> > you simply connect the user to the telco, and let
> > him
> > dial. To have the number logged, asterisk must
> > dial, so you can use this exten matching:
> > 
> > exten => _9.,2,Dial(Zap/1/${EXTEN:1})
> > 
> > the '.' after the nine just match anything you dial.
> > after the timeout (3 or five secs, don't remind),
> > asterisk dials the entire number to Zap/1 ,
> > stripping away the lead digit 9 , since EXTEN:1
> > means 'what have you dialled, beside the first
> > digit'
> > 
> > Matteo.
> > 
> > Il mar, 2003-07-08 alle 20:55, Erik Kendall ha
> > scritto:
> > > Yes, I'm using a 9.  I tried the following found
> > in
> > > sample extension.conf, but it didn't work because
> > I
> > > have 10 digit local dialing.
> > > 
> > > #exten => _9NXX,1,StripMSD,1
> > > #exten => _NXX,2,Dial,Zap/1/BYEXTENSION
> > > 
> > > So, I started using the following with success:
> > > 
> > > #ignorepat => 9
> > > #exten => 9,1,Dial,Zap/1/
> > > 
> > > What should I use for 10-digit local and 11-digit
> > long
> > > distance dialing?
> > > 
> > > Thanks,
> > > Erik
> > > 
> > > --- Brancaleoni Matteo <[EMAIL PROTECTED]>
> > wrote:
> > > > are you using some exten to get an external
> > > > dialtone?
> > > > like 0 -> give a dialtone
> > > > then compose the number
> > > > 
> > > > In this way numbers could not be logged, since
> > > > are dialled 'natively' on the telco dialtone by
> > the
> > > > user,
> > > > not by asterisk.
> > > > 
> > > > Matteo.
> > > > 
> > > > Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
> > > > scritto:
> > > > > Why doesn't the CDR show outgoing numbers?  I
> > need
> > > > a
> > > > > record of outbound digits dialed to reconcile
> > my
> > > > phone bills.
> > > > > 
> > > > > __
> > > > > Do you Yahoo!?
> > > > > SBC Yahoo! DSL - Now only $29.95 per month!
> > > > > http://sbc.yahoo.com
> > > > >
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> > > > >
> > > >
> > >
> >
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> > > > 
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> > > http://sbc.yahoo.com
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> 
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[Asterisk-Users] DID number assignment to SIP phones

[Mark Street]

The concept is fuzzy...  If one were to terminate a T1 PRI(s) into a T100P or 
T400P.  How are DID numbers assigned to IP phones on the inside through a SIP 
gateway.

What am I missing here?...

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Re: [Asterisk-Users] TDM400P noise?

[Steve]

On Saturday 05 July 2003 10:18 pm, Kevin Herzig wrote:
> Hi all.  I bought Digium's dev kit and a used IBM PL300 PC to try it out
> in.  The X100P works fine, but with the TDM400P I get what I can best
> describe as 'interrupt noise'... noise whenever I type a key on the
> keyboard, or when something accesses the disk drive, uses cpu, etc.
>
> In my other PC it works fine, no noise and sounds great.  But I cannot
> dedicate the other PC to it.
>
> Anyone have any ideas?   I'd be grateful.  So far I've had good success
> getting * to dial out via IConnectHere.  Neat stuff!
>
> Kevin

When I used * on an IBM PIII 600 (6565-85U) it also made a nice noise. I 
switched to a dell poweredge and it's gone. It seems to be related to the 
power supply. I'd try adding sime decent size capacitors (5000 u+) to 5 and 
12V, if I had the time to mess with it. Check to see if it is certified for 
home use which is harder than office.

-- 

Steve
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[Asterisk-Users] codec problems with asterisk

[Reece Anderson]

We appear to be having a problem with our asterisk setup.
We have a cisco AS5300 with pri lines coming in and passing the calls onto
asterisk then too the sip phones.

the phone call from the sip phones (7960's) appears to be ok nice and clear
including the user who has called in.

but if your the user who has called in its all crackley sounds really bad
when they speak.

i believe this problem is a codec problem as far as i can see we use ulaw
across the board, the 5300 currently supports 12 different codecs however
asterisk only like too work with ulaw or alaw it tends to not except the
call if the other codecs are used.

  clear-channel  Clear Channel 64000 bps
  g711alaw   G.711 A Law 64000 bps
  g711ulaw   G.711 u Law 64000 bps
  g723ar53   G.723.1 ANNEX-A 5300 bps
  g723ar63   G.723.1 ANNEX-A 6300 bps
  g723r53G.723.1 5300 bps
  g723r63G.723.1 6300 bps
  g726r16G.726 16000 bps
  g726r24G.726 24000 bps
  g726r32G.726 32000 bps
  g728   G.728 16000 bps
  g729br8G.729 ANNEX-B 8000 bps
  g729r8 G.729 8000 bps


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[Asterisk-Users] IAXTEL toll-free

[Paul Cheng]

Hi,

Has anyone been able to place a call via IAXTEL toll-free termination 
lately? I had it working at one time, but now it doesn't seem to work 
anymore. www.iaxtel.com also appears dead. Is this the server problem 
again or is it my config? Haven't been able to find any references in 
the list.

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Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider

[Paul Cheng]

Hi BK,

Using your configuration info, I now have Nikotel working again. Other  
than the fromuser=, it appears that one also now needs the auth=md5  
whereas before it was not necessary.

To disable incoming calling, just delete the register -> line for  
Nikotel. That way, no one can find you. You do not need to the register  
-> line for outgoing calls.

On Monday, July 7, 2003, at 11:16  AM, BK [address only for mailing  
lists] wrote:

Hi Paul,

thanks for your insights

On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote:

To dial a PSTN number through Nikotel used to work from Asterisk, but  
they had a very serious security issue (you could make calls anytime  
anywhere and their billing wouldn't charge it) and after I informed  
them of this, they changed their authentication mechanism and since  
then I have not gotten it to work (they didn't even thank me!).
This is what we have discovered last night. However, We have got it  
working now.

I will document this in detail and make it available, but briefly here  
a quick summary ...

First I had various glitches in my dial string. With the help of John  
Todd and some others on the IRC #asterisk channel I was able to fix  
those glitches. Thanks everybody who assisted.

Then I tried a number of things I had already experimented with  
before. When I turned on SIP debug and watched the datagrams, I could  
see Nikotel's response "account name does not match address of  
record". Together with the "from" part, this led me to fiddle with  
"fromuser" again and when I set it to the actual login name, it > worked.

Their tech people said it should work with a slight change: "yes, we  
changed it yesterday. Now the user part of the From: address has to  
be the same as the username in the Proxy-Authentication line. I don't  
know if the Asterisk can do that. The ATA186 does it b[y] default."

This CAN be done if you edit chan_sip.c,
It would seem you can do it a lot simpler:

in sip.conf
--- 
--
register => myusername:[EMAIL PROTECTED]

[nikotel]
username=myusername
fromuser=myusername
...
--- 
--

but when I did this, it billed me a few times for unconnected calls
Thanks for sharing this with us. I will watch this for a while and see  
if this happens here too.

 and I gave up trying to debug and switched to iConnect. iConnect is  
worse quality, but it is very easy to connect to.

I had much better quality with calls via Nikotel than iConnect, but  
their support is non-existent/bad at best. I sent them 3-4 e-mails  
about their security issue before they even responded.
Yes, support is not exactly their strength, is it?!

FYI. Registering with Nikotel was futile anyways, because I never  
figured out how anyone could call into me.
I don't want anybody to call in via Nikotel. Since they do not provide  
a telephone number for incoming calls, the only calls you could  
possibly get are from their public chat room. In the very best case  
you get a friendly test call from somebody who has just signed up and  
wants to try out the service, in the worst case you get prank calls in  
the middle of the night or indecent proposals and all the rest of it.

I will have to find a way to disable incoming calls from Nikotel  
entirely.


iConnect provides a PSTN-SIP dial in as an option, but I haven't  
tried it.
Yes, I have seen that. And at $8.95/mth it would seem reasonably  
priced, too.

Outbound calls do not require registering.

I can provide examples of iConnect connection scripts if you contact  
me offline.
Thanks, I will do that.

again many thanks to everybody who has helped solving this riddle
rgds
bk
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---
Paul Cheng
Mátyás király ut 10
H-1121 Budapest HUNGARY
[EMAIL PROTECTED]
mobile: +36 30 381-9311
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RE: [Asterisk-Users] Budgetone and Voicemail

[Brian Borders]

I never even noticed it worked fine without speakerphone.  Thanks for
pointing that out :)

Upgrading the firmware and changing the DMTF to Info corrected
everything.  Thanks so much!


Brian Borders
[EMAIL PROTECTED] 

-Original Message-
From: Brancaleoni Matteo [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, July 08, 2003 1:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Budgetone and Voicemail


using speakerphone with dtmf=inband? 
that's caused by a too loud dtmf tones when on
speakerphone.

upgrade to latest firmware (.72) and use
dtmf=info or rfc2833, should to the trick.

Matteo.

Il mar, 2003-07-08 alle 20:21, Brian Borders ha scritto:
> I have a problem with using voicemail on the Budgetone phones.  When 
> entering the mailbox and password, sometimes some keys will register 
> multiple times (as shown on console when it says no such user in 
> config
> file) and sometimes some keys won't even register at all.  It seems
> totally random.  Has anyone seen this problem?  Any recommendations
> would be greatly appreciated.  Thanks.
> 
> 
> Brian Borders
> [EMAIL PROTECTED]
> 
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[Asterisk-Users] SIP Problem (previous post) .. information might be relevant

[Dave Alan Caruana]

regarding my previous post about SIP outgoing calls
dropping with an error 481 ..

this is my output from  a SIP debug.
the call dropped occurs at the end.
Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my
control.

help :) please!!

Dave

Signal=0
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:57 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 117 INFO
Contact: 


10 headers, 0 lines
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 118 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:58 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 118 INFO
Contact: 


10 headers, 0 lines
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 119 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=2
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:58 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 119 INFO
Contact: 


10 headers, 0 lines
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 120 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250
 (no NAT) to 216.52.153.207:5060
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 121 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=6
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:59 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 121 INFO
Contact: 


10 headers, 0 lines
Retransmitting #1 (no NAT):
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 120 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=250

 to 216.52.153.207:5060
set_destination: Parsing  for address/port to
send to
set_destination: set destination to 216.52.153.207, port 5060
Reliably Transmitting:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 122 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=4
Duration=250
 (no NAT) to 216.52.153.207:5060
Sip read:
SIP/2.0 481 Invalid CSeq Number
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Call-ID: [EMAIL PROTECTED]
CSeq: 120 INFO
Content-Length: 0


7 headers, 0 lines
-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e
From: "21382890" ;tag=as6556b0d9
To: ;tag=26845C24-FDA
Date: Tue, 08 Jul 2003 22:22:59 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 122 INFO
Contact: 


10 headers, 0 lines
Sip read:
BYE sip:[EMAIL PRO

[Asterisk-Users] SIP disconnecting : response 481

[Dave Alan Caruana]

-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'


I am getting this error on an outgoing call to a SIP host.
The call just disconnects ..

is there any way around it ? 

thanks
Dave


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[Asterisk-Users] oh323 prob :)

[Dave Alan Caruana]

i'm getting Asterisk to dial an h323 call termination service ..

right now getting this message:

-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's' on channel 1, span 1
-- Executing Dial("Zap/1-1", "OH323/h323:[EMAIL PROTECTED]") in new
stack
  5:59.330 H323 Cleaner H323Connection
ip$localhost/18729 terminated.
ERROR[1230546240]: File chan_oh323.c, Line 704 (oh323_call): H323:0: Could
not call h323:[EMAIL PROTECTED]
-- Couldn't call h323:[EMAIL PROTECTED]
-- Hungup 'H323:0'
  == Everyone is busy at this time
-- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'


any idea what that can mean ?

I have my system currently working through SIP, however every now and then
it shows this
message

-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
  == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'

and drops the line which is the reason I am trying to use H323 instead,
maybe I can
get around that problem. Can anyone tell me what it means?

thanks
Dave


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Re: [Asterisk-Users] Budgetone and Voicemail

[Dan]

Hi,

The same problem with a Cisco 7960 phone too.
I don't think is phone related.

BR,
Dan

- Original Message - 
From: "WipeOut ." <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, July 08, 2003 9:42 PM
Subject: Re: [Asterisk-Users] Budgetone and Voicemail


> I have had double digits being passed every now and then once I am into
voicemail.. I haven't had a problem with the initial login stage.. I also
haven't had time to look into it yet..
>
> You could try changing the DTMF mode and see if it helps..
>
> Later..
>
> > I have a problem with using voicemail on the Budgetone phones.  When
> > entering the mailbox and password, sometimes some keys will register
> > multiple times (as shown on console when it says no such user in config
> > file) and sometimes some keys won't even register at all.  It seems
> > totally random.  Has anyone seen this problem?  Any recommendations
> > would be greatly appreciated.  Thanks.
> >
> >
> > Brian Borders
> > [EMAIL PROTECTED]
> >
> > ___
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Re: [Asterisk-Users] Call Accounting

[Erik Kendall]

Matteo,

Thank you.  I'll give it a try.

Erik

--- Brancaleoni Matteo <[EMAIL PROTECTED]> wrote:
> That's the reason why no number is logged, since
> you simply connect the user to the telco, and let
> him
> dial. To have the number logged, asterisk must
> dial, so you can use this exten matching:
> 
> exten => _9.,2,Dial(Zap/1/${EXTEN:1})
> 
> the '.' after the nine just match anything you dial.
> after the timeout (3 or five secs, don't remind),
> asterisk dials the entire number to Zap/1 ,
> stripping away the lead digit 9 , since EXTEN:1
> means 'what have you dialled, beside the first
> digit'
> 
> Matteo.
> 
> Il mar, 2003-07-08 alle 20:55, Erik Kendall ha
> scritto:
> > Yes, I'm using a 9.  I tried the following found
> in
> > sample extension.conf, but it didn't work because
> I
> > have 10 digit local dialing.
> > 
> > #exten => _9NXX,1,StripMSD,1
> > #exten => _NXX,2,Dial,Zap/1/BYEXTENSION
> > 
> > So, I started using the following with success:
> > 
> > #ignorepat => 9
> > #exten => 9,1,Dial,Zap/1/
> > 
> > What should I use for 10-digit local and 11-digit
> long
> > distance dialing?
> > 
> > Thanks,
> > Erik
> > 
> > --- Brancaleoni Matteo <[EMAIL PROTECTED]>
> wrote:
> > > are you using some exten to get an external
> > > dialtone?
> > > like 0 -> give a dialtone
> > > then compose the number
> > > 
> > > In this way numbers could not be logged, since
> > > are dialled 'natively' on the telco dialtone by
> the
> > > user,
> > > not by asterisk.
> > > 
> > > Matteo.
> > > 
> > > Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
> > > scritto:
> > > > Why doesn't the CDR show outgoing numbers?  I
> need
> > > a
> > > > record of outbound digits dialed to reconcile
> my
> > > phone bills.
> > > > 
> > > > __
> > > > Do you Yahoo!?
> > > > SBC Yahoo! DSL - Now only $29.95 per month!
> > > > http://sbc.yahoo.com
> > > >
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> > > >
> > >
> >
>
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> > > 
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> > >
> >
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> > 
> > 
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> >
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

On your place I would check separately if you can use both accounts. I
think that one of your accounts in disabled ...

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> First off, sorry for using a mail client without the "in-reply-to"
> function.
>
> Second:  I still can't make two calls using iconnecthere at the same
> time.
> Here is what I have tried:
>
> Attempt 1:
> >>exten=>_91NXXNXX,1,Dial,StripMSD
> >>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >>exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
>
> Attempt 2:
> >>exten=>_91NXXNXX,1,Dial,StripMSD
> >>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >>exten=>_1NXXNXX,103,Dial,SIP/[EMAIL PROTECTED]
>
> Attempt 3:
> >>exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> >>exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>
> Attempt 4:
> exten =>
> _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]&SIP/${EXTEN:[EMAIL PROTECTED]
>
>
> So far nothing has worked.
>
> Another question I have is about jitter buffer.  Is there a way to
> create a
> Jitter buffer in sip.conf?
> When I type sip show channels I get the following output:
>
> sip show channels
> Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> 213.137.73.176   xx  5752cb7a55f  00103/0  0ms  ms
> 4
>
> There is a section for Jitter, so I would imagine that there is some way
> to do it.
>
> Thank you for your time.
> Also, if anybody could suggest a good mail client for windows that is
> able to
> use the in-reply-to function, it would be helpful.
>
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[Asterisk-Users] Using multiple iconnecthere accounts

[Derek Beaumont]

First off, sorry for using a mail client without the "in-reply-to"
function.

Second:  I still can't make two calls using iconnecthere at the same
time.
Here is what I have tried:

Attempt 1:
>>exten=>_91NXXNXX,1,Dial,StripMSD
>>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
>>exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]

Attempt 2:
>>exten=>_91NXXNXX,1,Dial,StripMSD
>>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
>>exten=>_1NXXNXX,103,Dial,SIP/[EMAIL PROTECTED]

Attempt 3:
>>exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>>exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

Attempt 4:
exten =>
_91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]&SIP/${EXTEN:[EMAIL PROTECTED]


So far nothing has worked.

Another question I have is about jitter buffer.  Is there a way to
create a
Jitter buffer in sip.conf?
When I type sip show channels I get the following output:

sip show channels
Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
Format
213.137.73.176   xx  5752cb7a55f  00103/0  0ms  ms
4

There is a section for Jitter, so I would imagine that there is some way
to do it.

Thank you for your time.
Also, if anybody could suggest a good mail client for windows that is
able to 
use the in-reply-to function, it would be helpful.

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Re: [Asterisk-Users] Call Accounting

[Jeremy McNamara]

We've wrote our own Asterisk based billing system using the CDR 
interface and we have zero problems.

Jeremy McNamara



Erik Kendall wrote:

Why doesn't the CDR show outgoing numbers?  I need a
record of outbound digits dialed to reconcile my phone bills.
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Re: [Asterisk-Users] Call Accounting

[Brancaleoni Matteo]

That's the reason why no number is logged, since
you simply connect the user to the telco, and let him
dial. To have the number logged, asterisk must
dial, so you can use this exten matching:

exten => _9.,2,Dial(Zap/1/${EXTEN:1})

the '.' after the nine just match anything you dial.
after the timeout (3 or five secs, don't remind),
asterisk dials the entire number to Zap/1 ,
stripping away the lead digit 9 , since EXTEN:1
means 'what have you dialled, beside the first digit'

Matteo.

Il mar, 2003-07-08 alle 20:55, Erik Kendall ha scritto:
> Yes, I'm using a 9.  I tried the following found in
> sample extension.conf, but it didn't work because I
> have 10 digit local dialing.
> 
> #exten => _9NXX,1,StripMSD,1
> #exten => _NXX,2,Dial,Zap/1/BYEXTENSION
> 
> So, I started using the following with success:
> 
> #ignorepat => 9
> #exten => 9,1,Dial,Zap/1/
> 
> What should I use for 10-digit local and 11-digit long
> distance dialing?
> 
> Thanks,
> Erik
> 
> --- Brancaleoni Matteo <[EMAIL PROTECTED]> wrote:
> > are you using some exten to get an external
> > dialtone?
> > like 0 -> give a dialtone
> > then compose the number
> > 
> > In this way numbers could not be logged, since
> > are dialled 'natively' on the telco dialtone by the
> > user,
> > not by asterisk.
> > 
> > Matteo.
> > 
> > Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
> > scritto:
> > > Why doesn't the CDR show outgoing numbers?  I need
> > a
> > > record of outbound digits dialed to reconcile my
> > phone bills.
> > > 
> > > __
> > > Do you Yahoo!?
> > > SBC Yahoo! DSL - Now only $29.95 per month!
> > > http://sbc.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
> 
> 
> __
> Do you Yahoo!?
> SBC Yahoo! DSL - Now only $29.95 per month!
> http://sbc.yahoo.com
> ___
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RE: [Asterisk-Users] BudgeTone-100 Early Dial

[Paul Barrett]

Hi Stephen

Thanks for the reply

I am using inband DTMF and firmware version 1.0.3.72

Paul


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stephen R.
Besch
Sent: 07 July 2003 23:50
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] BudgeTone-100 Early Dial


Paul,

First, make sure that you use inband DTMF. As far as I know, out of band 
still does not work.  Second, make sure that the firmware is up to date. 
  The "silent DTMF" problem was fixed a few releases ago (at rev 
xx.xx.xx.60 I believe).

-- 
Stephen R. Besch, Ph.D.
SachsLab
320 Cary Hall
SUNY at Buffalo
Buffalo, NY 14214
(716) 829-3289 x106

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Re: [Asterisk-Users] Budgetone and Voicemail

[Brancaleoni Matteo]

using speakerphone with dtmf=inband? 
that's caused by a too loud dtmf tones when on
speakerphone.

upgrade to latest firmware (.72) and use
dtmf=info or rfc2833, should to the trick.

Matteo.

Il mar, 2003-07-08 alle 20:21, Brian Borders ha scritto:
> I have a problem with using voicemail on the Budgetone phones.  When
> entering the mailbox and password, sometimes some keys will register
> multiple times (as shown on console when it says no such user in config
> file) and sometimes some keys won't even register at all.  It seems
> totally random.  Has anyone seen this problem?  Any recommendations
> would be greatly appreciated.  Thanks.
> 
> 
> Brian Borders
> [EMAIL PROTECTED]
> 
> ___
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[BK [address only for mailing lists]]

On Wednesday, July 9, 2003, at 02:46 AM, Derek Beaumont wrote:

exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
What does EXTEN:1 do?  Why is StripMSD not used?
EXTEN:1 expands into the extension dialed without the first digit, 
that's why you don't need StripMSD.

rgds
bk
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Re: [Asterisk-Users] Call Accounting

[Erik Kendall]

Yes, I'm using a 9.  I tried the following found in
sample extension.conf, but it didn't work because I
have 10 digit local dialing.

#exten => _9NXX,1,StripMSD,1
#exten => _NXX,2,Dial,Zap/1/BYEXTENSION

So, I started using the following with success:

#ignorepat => 9
#exten => 9,1,Dial,Zap/1/

What should I use for 10-digit local and 11-digit long
distance dialing?

Thanks,
Erik

--- Brancaleoni Matteo <[EMAIL PROTECTED]> wrote:
> are you using some exten to get an external
> dialtone?
> like 0 -> give a dialtone
> then compose the number
> 
> In this way numbers could not be logged, since
> are dialled 'natively' on the telco dialtone by the
> user,
> not by asterisk.
> 
> Matteo.
> 
> Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
> scritto:
> > Why doesn't the CDR show outgoing numbers?  I need
> a
> > record of outbound digits dialed to reconcile my
> phone bills.
> > 
> > __
> > Do you Yahoo!?
> > SBC Yahoo! DSL - Now only $29.95 per month!
> > http://sbc.yahoo.com
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
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>
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Re: [Asterisk-Users] Budgetone and Voicemail

[WipeOut .]

I have had double digits being passed every now and then once I am into voicemail.. I 
haven't had a problem with the initial login stage.. I also haven't had time to look 
into it yet..

You could try changing the DTMF mode and see if it helps..

Later..

> I have a problem with using voicemail on the Budgetone phones.  When
> entering the mailbox and password, sometimes some keys will register
> multiple times (as shown on console when it says no such user in config
> file) and sometimes some keys won't even register at all.  It seems
> totally random.  Has anyone seen this problem?  Any recommendations
> would be greatly appreciated.  Thanks.
> 
> 
> Brian Borders
> [EMAIL PROTECTED]
> 
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

How about that:

exten => _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]&SIP/${EXTEN:[EMAIL 
PROTECTED]

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Asterisk has registered with both accounts:
>
> sip show registry
> Host  Username Refresh State
> 213.137.73.178:5060    120 Registered
> 213.137.73.178:5060    120 Registered
>
> I can make one call just fine, but when I try to make the second call,
> I get an invalid extension error.  When using the following
> configuration:
> >exten=>_91NXXNXX,1,StripMSD,1
> >exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> >exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
> I get the following output
>
> Executing Dial("Zap/4-1", "SIP/[EMAIL PROTECTED]") in new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/iconnect-cd45 is making progress passing it to Zap/4-1
>
> >show channels
> >Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> >213.137.73.178   xx  6631b1e766b  00103/0  0ms  ms
> 4
> >1 active SIP channel(s)
>
> This appears when I make the first call.  I notice that I have a 0ms
> Jitter buffer. I am now curious as to how I create a jitter buffer
> in sip.conf?  I have the following in the [general] section of sip.conf
>
> >jitterbuffer=yes
> >dropcount=3
> >maxjitterbuffer=2500
> >maxexccessbuffer=100
>
>
> Below is the output when I tried to call a second long distance number
>
> -- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> -- Got SIP response 480 "Temporarily not available" back from
> 213.137.73.178
> -- SIP/iconnect-fde9 is circuit-busy
>   == Everyone is busy at this time
> -- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> sip show channels
> Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
> Format
> 213.137.37.178   xx  7047ee1a76b  00102/0  0ms  ms
> 2
> 213.137.73.176   xx  7b782a7b3dd  00103/0  0ms  ms
> 4
> 2 active SIP channel(s)
> *CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
> retries exceeded on call
> [EMAIL PROTECTED] for seqno 102 (Request)
>   == No one is available to answer at this time
> -- Sent into invalid extension 'xxx' in context 'outgoing'
> on Zap/4-2
> -- Executing Playback("Zap/4-2", "TelError") in new stack
> -- Playing 'TelError'
> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
> exceeded on call [EMAIL PROTECTED] for
> seqno 102 (Request)
>
> Any help is appreciated.  Thank you for your time.
>
> OLD MESSAGE===
>
> >>Did asterisk register with both accounts ?
> >>"sip show registry"
> >>
> >>Can you post what happens on the console along with 'sip debug' ?
> >>
> >>Martin
>
> On Tue, 8 Jul 2003, Derek Beaumont wrote:
>
> > Has anybody out there tried to use two different iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the first is busy.
> > I have tried the following:
> >
> > exten=>_91NXXNXX,1,StripMSD,1
> > exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
> > first account
> > exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
> > the second account
> >
> > But that doesn't work.  Has anybody tried this before?
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
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Re: [Asterisk-Users] Budgetone and Voicemail

[James Sizemore]

Yes I have seen it. I had to change the digit time in the voicemail app 
and recompile.
There is a new voicemail2 app. I have not used it, but maybe it fixes 
this problem.
If you test it out, let me know how it works for you.

Brian Borders wrote:

I have a problem with using voicemail on the Budgetone phones.  When
entering the mailbox and password, sometimes some keys will register
multiple times (as shown on console when it says no such user in config
file) and sometimes some keys won't even register at all.  It seems
totally random.  Has anyone seen this problem?  Any recommendations
would be greatly appreciated.  Thanks.
Brian Borders
[EMAIL PROTECTED]
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[Asterisk-Users] Budgetone and Voicemail

[Brian Borders]

I have a problem with using voicemail on the Budgetone phones.  When
entering the mailbox and password, sometimes some keys will register
multiple times (as shown on console when it says no such user in config
file) and sometimes some keys won't even register at all.  It seems
totally random.  Has anyone seen this problem?  Any recommendations
would be greatly appreciated.  Thanks.


Brian Borders
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Call Accounting

[Brancaleoni Matteo]

are you using some exten to get an external dialtone?
like 0 -> give a dialtone
then compose the number

In this way numbers could not be logged, since
are dialled 'natively' on the telco dialtone by the user,
not by asterisk.

Matteo.

Il mar, 2003-07-08 alle 19:07, Erik Kendall ha scritto:
> Why doesn't the CDR show outgoing numbers?  I need a
> record of outbound digits dialed to reconcile my phone bills.
> 
> __
> Do you Yahoo!?
> SBC Yahoo! DSL - Now only $29.95 per month!
> http://sbc.yahoo.com
> ___
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Re: [Asterisk-Users] Call Accounting

[Erik Kendall]

I'm using Zap.  TDM10P and X100P

Thanks for any help.

--- Erik Kendall <[EMAIL PROTECTED]> wrote:
> Why doesn't the CDR show outgoing numbers?  I need a
> record of outbound digits dialed to reconcile my
> phone bills.
> 
> __
> Do you Yahoo!?
> SBC Yahoo! DSL - Now only $29.95 per month!
> http://sbc.yahoo.com
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
>
http://lists.digium.com/mailman/listinfo/asterisk-users


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[Asterisk-Users] Using multiple iconnecthere accounts

[Derek Beaumont]

>>Derek, tray this - it's working 100% with iconnect:
>>
>>exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>>exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
>>
>>
>>Best regards
>>Lubo

I tried that exact configuration, and now I can't even make a single
call.
What does EXTEN:1 do?  Why is StripMSD not used?

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Re: [Asterisk-Users] Call Accounting

[Steven Critchfield]

On Tue, 2003-07-08 at 12:07, Erik Kendall wrote:
> Why doesn't the CDR show outgoing numbers?  I need a
> record of outbound digits dialed to reconcile my phone bills.

What technology are you using? I have no problem with recording the
outbound numbers called on my zap channels.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Debug PRI!

[Steven Critchfield]

On Tue, 2003-07-08 at 12:06, Cristi wrote:
> This indicate that the connection with  the local provider PTSN it is ok? :
> 
>  > Called Number (len= 5) [ Ext: 1  TON: National Number (2)  NPI: 
> ISDN/Telephony Numbering Plan (E.164/E.163) (1) '10' ]

This is the interesting line. Does the number your calling to get into
this system end in 10? if so it would seem you are only getting the last
2 digits of the number and need to set up an extension for 10 in the
context you are dropping the call into.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Lot's of errors and warnings.

[marrandy]

Hadn't finished and sent the last mail by mistake.

Anyway...

Mandrake v9.1

#make progdocs

But   make progdocs   fails with :-

--
[EMAIL PROTECTED] asterisk]# make progdocs
doxygen asterisk-ng-doxygen
make: doxygen: Command not found
make: *** [progdocs] Error 127
[EMAIL PROTECTED] asterisk]#
-

checked and added doxygen.

Now builds successfully.



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RE: [Asterisk-Users] line battery check

[Jared Smith]

There's got to be a way... I think zttool shows a red alarm on an X100P
when there's no phone line plugged into it (and I would guess when
there's no voltage on the line.)  My guess is that it gets the info from
/proc/zap-something-or-other, but I'm just guessing.

Jared

On Tue, 2003-07-08 at 11:12, Jon Pounder wrote:
> At 05:18 PM 7/8/2003 +0100, you wrote:
> >Hi-
> >
> >Its not clear from your msg whether you want to detect the open-circuit
> >condition of  the T1/E1 directly or on one of the analog circuits connected
> >on the other side of the channel bank.
> 
> the t1 circuit is fine and the channel bank did not alarm during the 
> problem (it does have a form c alarm contact)
> 
> What I am asking is if anyone knows offhand if the alarm state can be 
> triggered on the fxo/fxs ports by reprogramming, or if it only applies to 
> the t1. OR is there any way with rbs to detect a line that lost loop power 
> ? OR is there a way to have asterisk check every few minutes when a line is 
> not in use that it can raise a dialtone ? if not then have some other 
> process get triggered.
> 
> 
> >If directly on the E1/T1, I imagine that asterisk can detect the condition
> >of the "alarm" signals for the circuit - these are separate from the
> >signalling supervision bits like on/off hook.  I'm new to asterisk, so don't
> >know how to read those alarm indicators in the software, but the ones you
> >are probably interested in are: "Red Alarm" - means loss of signal (I think
> >a cut line would cause this), "Yellow alarm" - means far end detected an
> >error, and "BPV" - bi-polar violation, means degraded signal (usually).
> >
> >Hope this helps.
> >
> >regards
> >
> >Scott M. Stingel
> >Emerging Voice Technology Inc.
> >
> >Email:  [EMAIL PROTECTED] 
> >URL:www.evtmedia.com 
> >
> >
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > Jon Pounder
> > > Sent: Tuesday, July 08, 2003 4:29 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] line battery check
> > >
> > >
> > > I know in the x100p hardware the driver can sense whether
> > > there is battery
> > > voltage on the line or not. Is this possible in the zap
> > > drivers using t1/e1
> > > interfaces ?
> > >
> > > I believe the signalling bits are only ring and hook, not
> > > loop presence, so
> > > not sure if it is possible.
> > >
> > > I had a situation last night where a main cable was cut in
> > > the area and
> > > lines were down overnight. My burglar alarm knew right away
> > > there was no
> > > line battery and started beeping. Data T1 down was a
> > > no-brainer for the
> > > monitoring software to detect. Now I am looking for a way to
> > > detect loop
> > > loss on the voice lines - any ideas how to accomplish with asterisk ?
> > >
> > > I have a zhone bank - anyone know offhand if I can program
> > > loop loss on a
> > > port to put it into an alarm state ? At least then I would
> > > have a contact
> > > closure I could monitor easily.
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
> >
> >
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> 
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Lubomir Christov]

Derek, tray this - it's working 100% with iconnect:

exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Best regards
Lubo
Derek Beaumont wrote:
Asterisk has registered with both accounts:

sip show registry
Host  Username Refresh State
213.137.73.178:5060    120 Registered
213.137.73.178:5060    120 Registered
I can make one call just fine, but when I try to make the second call,
I get an invalid extension error.  When using the following
configuration:
exten=>_91NXXNXX,1,StripMSD,1
exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] 
I get the following output

Executing Dial("Zap/4-1", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/iconnect-cd45 is making progress passing it to Zap/4-1

show channels
Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
Format

213.137.73.178   xx  6631b1e766b  00103/0  0ms  ms
4

1 active SIP channel(s)


This appears when I make the first call.  I notice that I have a 0ms 
Jitter buffer. I am now curious as to how I create a jitter buffer 
in sip.conf?  I have the following in the [general] section of sip.conf


jitterbuffer=yes
dropcount=3
maxjitterbuffer=2500
maxexccessbuffer=100


 
Below is the output when I tried to call a second long distance number

-- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 "Temporarily not available" back from
213.137.73.178
-- SIP/iconnect-fde9 is circuit-busy
  == Everyone is busy at this time
-- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
sip show channels
Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
Format
213.137.37.178   xx  7047ee1a76b  00102/0  0ms  ms
2
213.137.73.176   xx  7b782a7b3dd  00103/0  0ms  ms
4
2 active SIP channel(s)
*CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
  == No one is available to answer at this time
-- Sent into invalid extension 'xxx' in context 'outgoing'
on Zap/4-2
-- Executing Playback("Zap/4-2", "TelError") in new stack
-- Playing 'TelError'
WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)

Any help is appreciated.  Thank you for your time.

OLD MESSAGE===


Did asterisk register with both accounts ?
"sip show registry"
Can you post what happens on the console along with 'sip debug' ?

Martin


On Tue, 8 Jul 2003, Derek Beaumont wrote:


Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=>_91NXXNXX,1,StripMSD,1
exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] ;iconnect is the
first account
exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] ;iconnect2 is
the second account
But that doesn't work.  Has anybody tried this before?

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RE: [Asterisk-Users] line battery check

[Jon Pounder]

At 05:18 PM 7/8/2003 +0100, you wrote:
Hi-

Its not clear from your msg whether you want to detect the open-circuit
condition of  the T1/E1 directly or on one of the analog circuits connected
on the other side of the channel bank.
the t1 circuit is fine and the channel bank did not alarm during the 
problem (it does have a form c alarm contact)

What I am asking is if anyone knows offhand if the alarm state can be 
triggered on the fxo/fxs ports by reprogramming, or if it only applies to 
the t1. OR is there any way with rbs to detect a line that lost loop power 
? OR is there a way to have asterisk check every few minutes when a line is 
not in use that it can raise a dialtone ? if not then have some other 
process get triggered.


If directly on the E1/T1, I imagine that asterisk can detect the condition
of the "alarm" signals for the circuit - these are separate from the
signalling supervision bits like on/off hook.  I'm new to asterisk, so don't
know how to read those alarm indicators in the software, but the ones you
are probably interested in are: "Red Alarm" - means loss of signal (I think
a cut line would cause this), "Yellow alarm" - means far end detected an
error, and "BPV" - bi-polar violation, means degraded signal (usually).
Hope this helps.

regards

Scott M. Stingel
Emerging Voice Technology Inc.
Email:  [EMAIL PROTECTED] 
URL:www.evtmedia.com 


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Jon Pounder
> Sent: Tuesday, July 08, 2003 4:29 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] line battery check
>
>
> I know in the x100p hardware the driver can sense whether
> there is battery
> voltage on the line or not. Is this possible in the zap
> drivers using t1/e1
> interfaces ?
>
> I believe the signalling bits are only ring and hook, not
> loop presence, so
> not sure if it is possible.
>
> I had a situation last night where a main cable was cut in
> the area and
> lines were down overnight. My burglar alarm knew right away
> there was no
> line battery and started beeping. Data T1 down was a
> no-brainer for the
> monitoring software to detect. Now I am looking for a way to
> detect loop
> loss on the voice lines - any ideas how to accomplish with asterisk ?
>
> I have a zhone bank - anyone know offhand if I can program
> loop loss on a
> port to put it into an alarm state ? At least then I would
> have a contact
> closure I could monitor easily.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


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Re: [Asterisk-Users] Lot's of errors and warnings.

[BK [address only for mailing lists]]

On Wednesday, July 9, 2003, at 12:15 AM, marrandy wrote:

[about the kernel sources on Mandrake 9.1]

ZNot yet sure why it wasn't installed on this machine.  Will have to 
look into
that.
This would seem to be a Mandrake thing.

I had the same problem. The Mandrake installer would not install the 
kernel sources when doing a CD based installation. Although the kernel 
sources are on CD3, the package manager interface doesn't seem to 
find/show them.

I did an FTP based install instead in order to get a complete system 
with kernel sources.

rgds
bk
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[Asterisk-Users] Debug PRI!

[Cristi]

This indicate that the connection with  the local provider PTSN it is ok? :

  -- Attempting call on Zap/10 for [EMAIL PROTECTED]:1 (Retry 2)
-- Making new call for cr 32781
> Protocol Discriminator: Q.931 (8)  len=28
> Call Ref: len= 2 (reference 13/0xD) (Originator)
> Message type: SETUP (5)
> Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
>  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
>  Ext: 1  User information layer 1: A-Law (35)
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3
>   Ext: 1  Channel: 10 ]
> Display (len= 1) [ 1 ]
> Calling Number (len= 4) [ Ext: 0  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0)
>   Presentation: Unknown (67) '' ]
> Called Number (len= 5) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '10' ]
> Sending Complete (len= 0)
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 32781/0x800D) (Terminator)
< Message type: RELEASE COMPLETE (90)
< Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0 
Location: Public network serving the local user (2)
<  Ext: 1  Cause: Normal, unspecified (31), class = 
Normal Event (1) ]
-- Processing IE 8 (Cause)
-- Channel 10, span 1 got hangup
-- Hungup 'Zap/10-1'
NOTICE[28690]: File pbx_spool.c, Line 195 (attempt_thread): Call failed 
to go through, reason 1

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[Asterisk-Users] Call Accounting

[Erik Kendall]

Why doesn't the CDR show outgoing numbers?  I need a
record of outbound digits dialed to reconcile my phone bills.

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Re: [Asterisk-Users] Lot's of errors and warnings.

[marrandy]

On Tuesday 08 July 2003 11:15 am, marrandy wrote:

Spoke too soon.

Mamdrake v9.1 - hopefully, this will help other people who scan the arcvives 
first.

cd ../asterisk
# make clean ; make install

-
asterisk.c: In function `cli_complete':
asterisk.c:833: warning: assignment of read-only location
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: bison: Command not found
make: *** [ast_expr.c] Error 127


checked and installed bison.

now makes fine.

Added  make samples.

But   make progdocs   fails with :-

[EMAIL PROTECTED] asterisk]# make progdocs
doxygen asterisk-ng-doxygen
make: doxygen: Command not found
make: *** [progdocs] Error 127
[EMAIL PROTECTED] asterisk]#


-- 
The price one pays for pursuing any profession, or calling, is an intimate
knowledge of its ugly side.  -- James Baldwin

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[Asterisk-Users] Using multiple iconnecthere accounts

[Derek Beaumont]

Asterisk has registered with both accounts:

sip show registry
Host  Username Refresh State
213.137.73.178:5060    120 Registered
213.137.73.178:5060    120 Registered

I can make one call just fine, but when I try to make the second call,
I get an invalid extension error.  When using the following
configuration:
>exten=>_91NXXNXX,1,StripMSD,1
>exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
>exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]   
I get the following output

Executing Dial("Zap/4-1", "SIP/[EMAIL PROTECTED]") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/iconnect-cd45 is making progress passing it to Zap/4-1

>show channels
>Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
Format
>213.137.73.178   xx  6631b1e766b  00103/0  0ms  ms
4
>1 active SIP channel(s)

This appears when I make the first call.  I notice that I have a 0ms 
Jitter buffer. I am now curious as to how I create a jitter buffer 
in sip.conf?  I have the following in the [general] section of sip.conf

>jitterbuffer=yes
>dropcount=3
>maxjitterbuffer=2500
>maxexccessbuffer=100

 
Below is the output when I tried to call a second long distance number

-- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 "Temporarily not available" back from
213.137.73.178
-- SIP/iconnect-fde9 is circuit-busy
  == Everyone is busy at this time
-- Executing Dial("Zap/4-2", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
sip show channels
Peer UsernameCall ID  Seq (Tx/Rx)  Lag  Jitter
Format
213.137.37.178   xx  7047ee1a76b  00102/0  0ms  ms
2
213.137.73.176   xx  7b782a7b3dd  00103/0  0ms  ms
4
2 active SIP channel(s)
*CLI> WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
  == No one is available to answer at this time
-- Sent into invalid extension 'xxx' in context 'outgoing'
on Zap/4-2
-- Executing Playback("Zap/4-2", "TelError") in new stack
-- Playing 'TelError'
WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)

Any help is appreciated.  Thank you for your time.

OLD MESSAGE===

>>Did asterisk register with both accounts ?
>>"sip show registry"
>>
>>Can you post what happens on the console along with 'sip debug' ?
>>
>>Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Has anybody out there tried to use two different iconnecthere accounts
> with Asterisk?
> What I want to do is use a second account if the first is busy.
> I have tried the following:
>
> exten=>_91NXXNXX,1,StripMSD,1
> exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]  ;iconnect is the
> first account
> exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]  ;iconnect2 is
> the second account
>
> But that doesn't work.  Has anybody tried this before?
>
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>


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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[carlos del mayor]

 --- Steven Critchfield <[EMAIL PROTECTED]> escribió:
> On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:
> > Has anybody out there tried to use two different
> iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the
> first is busy.
> > I have tried the following:
> > 
> > exten=>_91NXXNXX,1,StripMSD,1
> >
> exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
> ;iconnect is the
> > first account
> >
> exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
> ;iconnect2 is
> > the second account
> > 
> > But that doesn't work.  Has anybody tried this
> before?
> 
> Isn't busy n+101 priority, or is it n+100? 

When you use the dial application, if the interface
you call to is busy, asterisk goes to priority n+101
if it's exist. If there is no response or priority
n+101 doesn't exist, then goes to priority n+1 as
usually.


>Basically
> you dial out
> similar to how you set up the busy portion of your
> voicemail.
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
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RE: [Asterisk-Users] line battery check

[Scott Stingel]

Hi-

Its not clear from your msg whether you want to detect the open-circuit
condition of  the T1/E1 directly or on one of the analog circuits connected
on the other side of the channel bank.

If directly on the E1/T1, I imagine that asterisk can detect the condition
of the "alarm" signals for the circuit - these are separate from the
signalling supervision bits like on/off hook.  I'm new to asterisk, so don't
know how to read those alarm indicators in the software, but the ones you
are probably interested in are: "Red Alarm" - means loss of signal (I think
a cut line would cause this), "Yellow alarm" - means far end detected an
error, and "BPV" - bi-polar violation, means degraded signal (usually).

Hope this helps.

regards

Scott M. Stingel 
Emerging Voice Technology Inc.

Email:  [EMAIL PROTECTED]    
URL:www.evtmedia.com    



> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jon Pounder
> Sent: Tuesday, July 08, 2003 4:29 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] line battery check
> 
> 
> I know in the x100p hardware the driver can sense whether 
> there is battery 
> voltage on the line or not. Is this possible in the zap 
> drivers using t1/e1 
> interfaces ?
> 
> I believe the signalling bits are only ring and hook, not 
> loop presence, so 
> not sure if it is possible.
> 
> I had a situation last night where a main cable was cut in 
> the area and 
> lines were down overnight. My burglar alarm knew right away 
> there was no 
> line battery and started beeping. Data T1 down was a 
> no-brainer for the 
> monitoring software to detect. Now I am looking for a way to 
> detect loop 
> loss on the voice lines - any ideas how to accomplish with asterisk ?
> 
> I have a zhone bank - anyone know offhand if I can program 
> loop loss on a 
> port to put it into an alarm state ? At least then I would 
> have a contact 
> closure I could monitor easily.
> 
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> 
> 



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Re: [Asterisk-Users] line battery check

[Steven Critchfield]

On Tue, 2003-07-08 at 10:29, Jon Pounder wrote:
> I know in the x100p hardware the driver can sense whether there is battery 
> voltage on the line or not. Is this possible in the zap drivers using t1/e1 
> interfaces ?
> 
> I believe the signalling bits are only ring and hook, not loop presence, so 
> not sure if it is possible.
> 
> I had a situation last night where a main cable was cut in the area and 
> lines were down overnight. My burglar alarm knew right away there was no 
> line battery and started beeping. Data T1 down was a no-brainer for the 
> monitoring software to detect. Now I am looking for a way to detect loop 
> loss on the voice lines - any ideas how to accomplish with asterisk ?
> 
> I have a zhone bank - anyone know offhand if I can program loop loss on a 
> port to put it into an alarm state ? At least then I would have a contact 
> closure I could monitor easily.

I don't think the zhones support external notification of alarm other
than the LEDs. The CACs do for sure, and therefore it is a good bet the
Adtran stuff does as well. I know on my CAC hardware there is an
external plug to tie into to put any number of monitors on to detect
when it has had a problem.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] three way calling and cisco ata 186

[Pavel Zheltouhov]

Thomas Dingermann wrote:
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
I just try two ATA with asterisk with that configuration files :

;
; MGCP Configuration for Asterisk
;
[general]
port = 2727
bindaddr = 0.0.0.0
allow=ulaw
inbanddtmf=yes
transfer = yes
threewaycalling=yes
[10.0.1.19]
transfer = yes
threewaycalling=yes
host = 10.0.1.19
context = default
line => aaln/1
transfer = 1
line => aaln/2
transfer = 1
line => *
[10.0.1.20]
transfer = yes
threewaycalling=yes
host = 10.0.1.20
context = default
line => aaln/1
transfer = 1
line => aaln/2
transfer = 1
line => *
and extensions.conf

---
exten => 31,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 32,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 33,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
exten => 34,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr)
-
Ordinary tasks works good. Call transfer with '#' key work too.
But three way calling not work with stange error :
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
dial to 33
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3'
-- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", 
"MGCP/aaln/[EMAIL PROTECTED]|20|tr") in new stack
-- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: 0, dnd: 0, so: 0, sno: 0
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
-- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is ringing

answer
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd'
-- MGCP/aaln/[EMAIL PROTECTED] answered MGCP/aaln/[EMAIL PROTECTED]
-- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED]
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]

Talking now
Attempt call person 3 :
 hookflash :
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
-- Swapping 1 for 0 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 1 on aaln/[EMAIL PROTECTED]
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down
now trying dial to other phone ( 600 - echo test )

-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '6'

atfter this, person1 hear 'fastbusy', short beeps !

And other output of asterisk:

-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0'
-- Executing Playback("MGCP/aaln/[EMAIL PROTECTED]", "demo-echotest") 
in new stack
-- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED]
-- Playing 'demo-echotest'
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf'
-- Swapping 0 for 1 on aaln/[EMAIL PROTECTED]
-- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 0 on aaln/[EMAIL PROTECTED]
-- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and 
MGCP/aaln/[EMAIL PROTECTED]
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): 
mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED])
WARNING[20501]: File chan_mgcp.c, Line 764 (mgcp_fixup): old channel 
wasn't 0x81065a8 but was (nil)
WARNING[20501]: File channel.c, Line 1847 (ast_do_masquerade): Fixup 
failed on channel MGCP/aaln/[EMAIL PROTECTED], strange things may happen.
NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): 
mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED])
-- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu'
  == Spawn extension (default, 600, 1) exited non-zero on 
'MGCP/aaln/[EMAIL PROTECTED]'

--

Any ideas ?

--
Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http://www.comlink.ru
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

Busy is n+1 if n+101 doesn't exist.

Martin

On 8 Jul 2003, Steven Critchfield wrote:

> On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:
> > Has anybody out there tried to use two different iconnecthere accounts
> > with Asterisk?
> > What I want to do is use a second account if the first is busy.
> > I have tried the following:
> >
> > exten=>_91NXXNXX,1,StripMSD,1
> > exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
> > first account
> > exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
> > the second account
> >
> > But that doesn't work.  Has anybody tried this before?
>
> Isn't busy n+101 priority, or is it n+100? Basically you dial out
> similar to how you set up the busy portion of your voicemail.
> --
> Steven Critchfield  <[EMAIL PROTECTED]>
>
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RE: [Asterisk-Users] Asterisk and VMWare

[John Laur]

> I use a similar asterisk setup when my T100P card is being used
outside
> the home. I ran it on a 1ghz athlon that was shared as my workstation,
> but all linux. When my screen saver started, the audio quality dropped
> below usable. So obviously my system was idle if the screen saver
> started. SO as you can see you most likely will not get good results
> under vmware even if you are on a pretty hefty machine.

VMWare GSX server would give you better performance (probably enough to
run asterisk in the VM); however, it does not meet your criteria for a
couple reasons: 1) Windows cannot be used as the host environment 2) It
costs as much as a machine sufficient to run asterisk for your purposes.

If you are not interfacing to any hardware devices you might be able to
get asterisk compiling with MinGW or Cygwin and running natively in
Win32. You would not have zaptel timing, so MusicOnHold, MeetMe, etc.
are all out unless you somehow emulate the functionality of zaprtc or
ztdummy under windows. I am sure there are a lot of other issues
compiling under Win32 that will come up as well, but still it may be
possible.

John

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Re: [Asterisk-Users] Lot's of errors and warnings.

[Steven Critchfield]

On Tue, 2003-07-08 at 10:15, marrandy wrote:
> On Tuesday 08 July 2003 10:47 am, Steven Critchfield wrote:
> 
> Yes Steven, I realized after I posted the last mail that I had only checked 
> the first two parts of the MUST requirement and had assumed the kernel src 
> was installed.
> 
> mea culpa
> 
> ZNot yet sure why it wasn't installed on this machine.  Will have to look into 
> that.

Because Mandrake is not a server distribution. Then again, I'm not sure
of any of the package oriented system placing the kernel package on a
machine by default. I guess most distributions assume if you are going
to go through the effort of recompiling the kernel, then you can go get
your kernel source.


-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] line battery check

[Jon Pounder]

I know in the x100p hardware the driver can sense whether there is battery 
voltage on the line or not. Is this possible in the zap drivers using t1/e1 
interfaces ?

I believe the signalling bits are only ring and hook, not loop presence, so 
not sure if it is possible.

I had a situation last night where a main cable was cut in the area and 
lines were down overnight. My burglar alarm knew right away there was no 
line battery and started beeping. Data T1 down was a no-brainer for the 
monitoring software to detect. Now I am looking for a way to detect loop 
loss on the voice lines - any ideas how to accomplish with asterisk ?

I have a zhone bank - anyone know offhand if I can program loop loss on a 
port to put it into an alarm state ? At least then I would have a contact 
closure I could monitor easily.

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Re: [Asterisk-Users] Answering on an zap device

[Steven Critchfield]

On Tue, 2003-07-08 at 10:06, Cristi wrote:
> Steven Critchfield wrote:
> 
> >Please trim unnecessary lines from the post. 
> >
> >On Tue, 2003-07-08 at 09:23, Cristi wrote:
> >  
> >
> >>What is the need for the callerid in the following configuration:
> >>(default config page)
> >>;callerid="Green Phone"<(256) 428-6121>
> >>;channel => 1
> >>;callerid="Black Phone"<(256) 428-6122>
> >>;channel => 2
> >>;callerid="CallerID Phone" <(256) 428-6123>
> >>;callerid="CallerID Phone" <(630) 372-1564>
> >>;callerid="CallerID Phone" <(256) 704-4666>
> >>;channel => 3
> >>;callerid="Pac Tel Phone" <(256) 428-6124>
> >>;channel => 4
> >>;callerid="Uniden Dead" <(256) 428-6125>
> >>;channel => 5
> >>;callerid="Cortelco 2500" <(256) 428-6126>
> >>;channel => 6
> >>;callerid="Main TA 750" <(256) 428-6127>
> >>;channel => 44
> >>It is the mapping between PTSN nr and channel numbers?
> >>
> >>
> >
> >This is to make internal phones show callerID. Notice the "Green phone",
> >"Black Phone", These are to specify what callerid to use when the
> >channel is used to place a call.
> >
> >In my office and at home I have set each channels caller ID to specify
> >the user at the end of the phone. This way you know who it is inside
> >your office calling you when you see the callerid. Also, when used
> >appropriately, you can specify either your main trunk number or specific
> >DID numbers for the callerID display on a PRI line. This way you could
> >set up a 9+ dialing for DID display, and 8+ for main line display. This
> >is helpful when you are doing generic calling or specific calling. 
> >  
> >
> First let me thank you for all the answers that I got from the asterisk 
> people!
> 
> Another question : Let say that now I have a E1 ISDN  into a 4E1 card 
> and I don't have any errors :. What configuration I have to do get a 
> call ? Tring a nr associated with the channel I got only a busy 
> line!From  where the asterisk know what nr to answer ? It is specified 
> into the extension config file for  the context associated with the 
> channel group?

Turn on debug for your pri spans so you can see how many digits are
being sent to you in the called number part of the q.931 packets. From
there you will need to specify in extensions.conf a extension for each
of the phone numbers associated with the PRIs to the precision of what
you observed before. Good chance is that it is the full number.

As for the context, you should have specified that in your zapata.conf
file as to where these calls would be dropped into. I have mine defined
as PRI and then reroute appropriately from there.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Martin Pycko]

Did asterisk register with both accounts ?
"sip show registry"

Can you post what happens on the console along with 'sip debug' ?

Martin

On Tue, 8 Jul 2003, Derek Beaumont wrote:

> Has anybody out there tried to use two different iconnecthere accounts
> with Asterisk?
> What I want to do is use a second account if the first is busy.
> I have tried the following:
>
> exten=>_91NXXNXX,1,StripMSD,1
> exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]  ;iconnect is the
> first account
> exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]  ;iconnect2 is
> the second account
>
> But that doesn't work.  Has anybody tried this before?
>
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Re: [Asterisk-Users] Using multiple iconnecthere accounts

[Steven Critchfield]

On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:
> Has anybody out there tried to use two different iconnecthere accounts
> with Asterisk?
> What I want to do is use a second account if the first is busy.
> I have tried the following:
> 
> exten=>_91NXXNXX,1,StripMSD,1
> exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]  ;iconnect is the
> first account
> exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]  ;iconnect2 is
> the second account
> 
> But that doesn't work.  Has anybody tried this before?

Isn't busy n+101 priority, or is it n+100? Basically you dial out
similar to how you set up the busy portion of your voicemail.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Transfert call

[carlos del mayor]

Sorry!
Didn't know it got implemented!!Last notice I had is
that it would be implemented soon, but didn't think it
was SO soon...Great then!!
cmayor

 --- Martin Pycko <[EMAIL PROTECTED]> escribió: >
That got implemented recently ...
> 
> Martin
> 
> On Tue, 8 Jul 2003, carlos del mayor wrote:
> 
> > Hi Rattana,
> >
> > That kind of transfer is not yet implemented in *.
> The
> > way it will be indicated is:
> > exten =>111,dial,Zap/1,20,T
> >
> > The T indicate that transfer is permitted for
> calling
> > party, but as I've said, that's not implemented at
> the
> > moment.
> >
> > Regards
> > cmayor
> >
> >  --- Rattana BIV <[EMAIL PROTECTED]> escribió: >
> Hi,
> > >
> > >
> > > A question about transfert.
> > >
> > > How can I make transfert with the the person who
> > > call.
> > > X call Z and X transfert Z to Y.
> > > I only succeed to do X call Z and Z transfert to
> Y.
> > >
> > > If someone have a solution it will be very good
> =)
> > >
> > >
> > > regards
> > > Rattana
> >
> >
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> >
>
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> >
> 
> 
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Re: [Asterisk-Users] Lot's of errors and warnings.

[marrandy]

On Tuesday 08 July 2003 10:47 am, Steven Critchfield wrote:

Yes Steven, I realized after I posted the last mail that I had only checked 
the first two parts of the MUST requirement and had assumed the kernel src 
was installed.

mea culpa

ZNot yet sure why it wasn't installed on this machine.  Will have to look into 
that.

-- 
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half his wife's fault, and half her mother's.

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Re: [Asterisk-Users] RTP.C codec error 19

[Lord Stroud]

Hey JJ,

  You are correct, as always. However, please remember that some carriers will 
simply say: "Sorry, if you don't support VAD, we are not going to change our 
setup for a single client".

Regards,
  Nir S

On Tuesday 08 July 2003 05:44 pm, Jeremy McNamara wrote:
> Um no.   Turn off Silence suppression (VAD) on your endpoint.
>
>
> Jeremy McNamara
>
> Lord Stroud wrote:
> >Hi Dave,
> >
> >  The RTP codec 19 error that you are getting indicates that your endpoint
> > is most probably activating silence supression, and that you are using a
> > codec such as g.729, at least, that is what I get on my platform here.
> >
> >  You can go into the the rtc.c file, and simply comment out the message.
> >Edit the rtp.c file at line 330, as the following:
> >
> >ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
> >
> >  simply edit it to be:
> >
> >//ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
> >
> >  and simply re-compile.
> >
> >Nir S
> >
> >On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote:
> >>hi ..
> >>when placing a SIP call to a sip host in the states
> >>every few seconds I get an RTP codec 19 error.
> >>I know this is related to comfort noise, and the
> >>call goes through OK ... how can I suppress
> >>the error message ?
> >>
> >>Also, many times I get "Invalid CSeq Number"
> >>back from 216.52.153.207 (which is the host
> >>i'm calling) and the call drops.. is there a solution
> >>for this ?
> >>
> >>cheers
> >>Dave
> >>
> >>(I mistakenly put an "re" in the title of this email
> >> and I think it's been ignored .. reposted)
> >>
> >>
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Re: [Asterisk-Users] oh323 problem (small one)

[Michael Manousos]

Hi,

Dave Alan Caruana wrote:
I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.
When I try to dial an outbound oh323 call I get the following error :

-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's' on channel 1, span 1
-- Executing Dial("Zap/1-1", "OH323/[EMAIL PROTECTED]") in new stack
  1:04.782  ThreadID=0x4958a540 H323Attempt to use invalid URL
"[EMAIL PROTECTED]:1720"
-- Couldn't call [EMAIL PROTECTED]
-- Hungup 'H323:0'
  == Everyone is busy at this time
-- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
There are some changes in the OpenH323 library regarding the
calling scheme used. The new style is h323:[EMAIL PROTECTED]
I thought that it would be better not to make changes
on the called address before making the call (adding this
"h323:" prefix), in order to be more flexible. Of course
this breaks older extension files (only in the case of a
direct call, something like [EMAIL PROTECTED]).
my extensions.conf is :

[incoming]
exten => s,1,Wait,1
exten => s,2,Dial(OH323/[EMAIL PROTECTED])
So, just change it into OH323/h323:[EMAIL PROTECTED]
and you should be just fine.
exten => s,3,Hangup

I can't see anything obviously wrong, and dialling that h323 address from
SJphone
works fine (and used to work fine from asterisk too before).
help please :)

Dave



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[Asterisk-Users] Using multiple iconnecthere accounts

[Derek Beaumont]

Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:

exten=>_91NXXNXX,1,StripMSD,1
exten=>_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
first account
exten=>_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
the second account

But that doesn't work.  Has anybody tried this before?

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Re: [Asterisk-Users] Answering on an zap device

[Cristi]

Steven Critchfield wrote:

Please trim unnecessary lines from the post. 

On Tue, 2003-07-08 at 09:23, Cristi wrote:
 

What is the need for the callerid in the following configuration:
(default config page)
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
It is the mapping between PTSN nr and channel numbers?
   

This is to make internal phones show callerID. Notice the "Green phone",
"Black Phone", These are to specify what callerid to use when the
channel is used to place a call.
In my office and at home I have set each channels caller ID to specify
the user at the end of the phone. This way you know who it is inside
your office calling you when you see the callerid. Also, when used
appropriately, you can specify either your main trunk number or specific
DID numbers for the callerID display on a PRI line. This way you could
set up a 9+ dialing for DID display, and 8+ for main line display. This
is helpful when you are doing generic calling or specific calling. 
 

First let me thank you for all the answers that I got from the asterisk 
people!

Another question : Let say that now I have a E1 ISDN  into a 4E1 card 
and I don't have any errors :. What configuration I have to do get a 
call ? Tring a nr associated with the channel I got only a busy 
line!From  where the asterisk know what nr to answer ? It is specified 
into the extension config file for  the context associated with the 
channel group?

 == D-Channel on span 1 up
 == D-Channel on span 3 up
   -- B-channel 1 successfully restarted on span 3
   -- B-channel 2 successfully restarted on span 3
   -- B-channel 3 successfully restarted on span 3
   -- B-channel 4 successfully restarted on span 3
   -- B-channel 5 successfully restarted on span 3
   -- B-channel 6 successfully restarted on span 3
   -- B-channel 7 successfully restarted on span 3
   -- B-channel 8 successfully restarted on span 3
   -- B-channel 9 successfully restarted on span 3
   -- B-channel 10 successfully restarted on span 3
   -- B-channel 11 successfully restarted on span 3
   -- B-channel 12 successfully restarted on span 3
   -- B-channel 13 successfully restarted on span 3
   -- B-channel 14 successfully restarted on span 3
   -- B-channel 15 successfully restarted on span 3
   -- B-channel 17 successfully restarted on span 3
   -- B-channel 18 successfully restarted on span 3
   -- B-channel 19 successfully restarted on span 3
   -- B-channel 20 successfully restarted on span 3
   -- B-channel 21 successfully restarted on span 3
   -- B-channel 22 successfully restarted on span 3
   -- B-channel 23 successfully restarted on span 3
   -- B-channel 24 successfully restarted on span 3
   -- B-channel 25 successfully restarted on span 3
   -- B-channel 26 successfully restarted on span 3
   -- B-channel 27 successfully restarted on span 3
   -- B-channel 28 successfully restarted on span 3
   -- B-channel 29 successfully restarted on span 3
   -- B-channel 30 successfully restarted on span 3
   -- B-channel 31 successfully restarted on span 3
   -- B-channel 1 successfully restarted on span 1
   -- B-channel 2 successfully restarted on span 1
   -- B-channel 3 successfully restarted on span 1
   -- B-channel 4 successfully restarted on span 1
   -- B-channel 5 successfully restarted on span 1
   -- B-channel 6 successfully restarted on span 1
   -- B-channel 7 successfully restarted on span 1
   -- B-channel 8 successfully restarted on span 1
   -- B-channel 9 successfully restarted on span 1
   -- B-channel 10 successfully restarted on span 1
   -- B-channel 11 successfully restarted on span 1
   -- B-channel 12 successfully restarted on span 1
   -- B-channel 13 successfully restarted on span 1
   -- B-channel 14 successfully restarted on span 1
   -- B-channel 15 successfully restarted on span 1
   -- B-channel 17 successfully restarted on span 1
   -- B-channel 18 successfully restarted on span 1
   -- B-channel 19 successfully restarted on span 1
   -- B-channel 20 successfully restarted on span 1
   -- B-channel 21 successfully restarted on span 1
   -- B-channel 22 successfully restarted on span 1
   -- B-channel 23 successfully restarted on span 1
   -- B-channel 24 successfully restarted on span 1
   -- B-channel 25 successfully restarted on span 1
   -- B-channel 26 successfully restarted on span 1
   -- B-channel 27 successfully restarted on span 1
   -- B-channel 28 successfully restarted on span 1
   -- B-channel 29 successfully restarted on span 1
   -- B-channel 30 successfully restarted on span 1
   -- B-channel 31 successfully resta

Re: [Asterisk-Users] overlap dialing on a pri span

[Martin Pycko]

Well first of all if you set up DigitTimeout to 5 seconds so asterisk is
going to wait up to 5 seconds to retrieve the digits specially when you
have a match of _X. that is at least to digits but with the timeout of 5
you could imagine that asterisk will intercept all digits.

How about having a pattern _X (without a dot). The amount of digits that
asterisk is waiting for is set by you. _X is one digit, _X is 5 digits

Martin

On Tue, 8 Jul 2003 [EMAIL PROTECTED] wrote:

> Martin,
>
> I probably should have mentioned that: overlapdial=yes was set in
> zapata.conf (I take it this option is inherited through all the
> channels I configure in zapata.conf). I also did a fresh checkout today.
>
> My guess is that the main problem for now lies in the fact that asterisk
> won't execute a dial application once it received the first digit.
> Apparently, the extension _X. won't spawn dial before asterisk hits
> the timeout:
>
> exten => s,1,Wait,1 ; Wait a second, just for fun
> exten => s,2,Answer ; Answer the line
> exten => s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds
> exten => s,4,ResponseTimeout,2  ; Set Response Timeout to 10 seconds
> exten => _X.,1,Dial,Zap/g8/BYEXTENSION
>
> I can see asterisk pick up:
>
> -- Executing Answer("Zap/159-1", "") in new stack
>
> the receive some digits
>
> DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 7 on Zap/159-1
> DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 8 on Zap/159-1
> <...>
>
> and seconds (!) later asterisk dials out
>
> -- Executing Dial("Zap/159-1", "Zap/g8/BYEXTENSION") in new stack
> -- Called g8/78997899
> -- Channel 1, span 8 got hangup
>
> Do you know why? Is there a minimum number of digits asterisk need for an
> inital setup message?
>
> Thilo
>
> > overlapdial=yes in zapata.conf
> > for those channels that you want the overlapdialing be activated.
> >
> > By default only incoming overlap dialing is enabled.
>
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Re: [Asterisk-Users] Lot's of errors and warnings.

[marrandy]

On Tuesday 08 July 2003 10:34 am, marrandy wrote:
> On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote:
> > Do you have the source that your kernel was compiled from? Do you at
> > least have the appropriate headers for you kernel and the config file
> > that was used?
> > 
> 
> Haven't a clue.
> 
> I was just following the install instructions as per :-
> 
> http://www.asterisk.org/index.php?menu=download
> 


Aahh...looks like the linux kernel source wasn't installed on this machine 
(Mandrake Linux v9.1).

Done...

Builds fine now.

Thread end.

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rather than surrender any material part of their advantage.
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Re: [Asterisk-Users] Lot's of errors and warnings.

[Steven Critchfield]

On Tue, 2003-07-08 at 09:34, marrandy wrote:
> On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote:
> > Do you have the source that your kernel was compiled from? Do you at
> > least have the appropriate headers for you kernel and the config file
> > that was used?
> > 
> 
> Haven't a clue.
> 
> I was just following the install instructions as per :-
> 
> http://www.asterisk.org/index.php?menu=download

Look at the last section. I'll reproduce it here so you know what I'm
talking about.

Note that your system MUST meet these requirements:

You must have readline and openssl and their respective development packages.
You must be running Linux 2.4.x
You must have the Linux Kernel Sources package installed on your system.


Look closely at the last line. If you followed instructions, then you
would have had a clue.

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] RTP.C codec error 19

[Jeremy McNamara]

Um no.   Turn off Silence suppression (VAD) on your endpoint.

Jeremy McNamara



Lord Stroud wrote:

Hi Dave,

 The RTP codec 19 error that you are getting indicates that your endpoint is
most probably activating silence supression, and that you are using a codec
such as g.729, at least, that is what I get on my platform here.
 You can go into the the rtc.c file, and simply comment out the message.
Edit the rtp.c file at line 330, as the following:
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);

 simply edit it to be:

//ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);

 and simply re-compile.

Nir S

On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote:
 

hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?
Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is the host
i'm calling) and the call drops.. is there a solution
for this ?
cheers
Dave
(I mistakenly put an "re" in the title of this email
and I think it's been ignored .. reposted)
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Re: [Asterisk-Users] Lot's of errors and warnings.

[marrandy]

On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote:
> Do you have the source that your kernel was compiled from? Do you at
> least have the appropriate headers for you kernel and the config file
> that was used?
> 

Haven't a clue.

I was just following the install instructions as per :-

http://www.asterisk.org/index.php?menu=download

-- 
"Neighbors!!  We got neighbors!  We ain't supposed to have any neighbors, and
I just had to shoot one."
-- Post Bros. Comics

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Re: [Asterisk-Users] Transfert call

[Martin Pycko]

That got implemented recently ...

Martin

On Tue, 8 Jul 2003, carlos del mayor wrote:

> Hi Rattana,
>
> That kind of transfer is not yet implemented in *. The
> way it will be indicated is:
> exten =>111,dial,Zap/1,20,T
>
> The T indicate that transfer is permitted for calling
> party, but as I've said, that's not implemented at the
> moment.
>
> Regards
> cmayor
>
>  --- Rattana BIV <[EMAIL PROTECTED]> escribió: > Hi,
> >
> >
> > A question about transfert.
> >
> > How can I make transfert with the the person who
> > call.
> > X call Z and X transfert Z to Y.
> > I only succeed to do X call Z and Z transfert to Y.
> >
> > If someone have a solution it will be very good =)
> >
> >
> > regards
> > Rattana
>
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Re: [Asterisk-Users] Answering on an zap device

[Steven Critchfield]

Please trim unnecessary lines from the post. 

On Tue, 2003-07-08 at 09:23, Cristi wrote:
> What is the need for the callerid in the following configuration:
> (default config page)
> ;callerid="Green Phone"<(256) 428-6121>
> ;channel => 1
> ;callerid="Black Phone"<(256) 428-6122>
> ;channel => 2
> ;callerid="CallerID Phone" <(256) 428-6123>
> ;callerid="CallerID Phone" <(630) 372-1564>
> ;callerid="CallerID Phone" <(256) 704-4666>
> ;channel => 3
> ;callerid="Pac Tel Phone" <(256) 428-6124>
> ;channel => 4
> ;callerid="Uniden Dead" <(256) 428-6125>
> ;channel => 5
> ;callerid="Cortelco 2500" <(256) 428-6126>
> ;channel => 6
> ;callerid="Main TA 750" <(256) 428-6127>
> ;channel => 44
> It is the mapping between PTSN nr and channel numbers?

This is to make internal phones show callerID. Notice the "Green phone",
"Black Phone", These are to specify what callerid to use when the
channel is used to place a call.

In my office and at home I have set each channels caller ID to specify
the user at the end of the phone. This way you know who it is inside
your office calling you when you see the callerid. Also, when used
appropriately, you can specify either your main trunk number or specific
DID numbers for the callerID display on a PRI line. This way you could
set up a 9+ dialing for DID display, and 8+ for main line display. This
is helpful when you are doing generic calling or specific calling. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] ENUM lookups

[dlist]

enum.diff
Description: Binary data


For those interested, this improves the ways lookups for NAPTR records 
are done. Maybe it can be patched on the main source tree by the 
authors. Other thing I bumped into was that compiling with gcc 2.95 
cause asterisk to segfault when doing ENUM lookups. Compiling with gcc 
3.3 solved this (on debian testing).

Adrian Georgescu
AG Projects
[EMAIL PROTECTED]
www.ag-projects.com
IP phone:  sip:[EMAIL PROTECTED]

Managed DNS for IP telephony http://managed-dns.org/

Re: [Asterisk-Users] Answering on an zap device

[Cristi]

Steven Critchfield wrote:

First off, a E1 circuit is 32 channels, 1-32, 33-64, 65-96, 97-128. Also
there is a D channel on 16 and 32 of each span. Then you need to add the
D channel definition to your zapata.conf files. 

I believe all this was covered in the examples.



On Tue, 2003-07-08 at 08:20, Cristi wrote:
 

How can I accept calls on a Wildcard E400P  . Please include the 
zaptel.conf , zapata.conf and extension.conf to fully understand 
everything.

Take a look firts to my configuration file. Where I did wrong?

zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
# Span 1
bchan=1-15,17-31
dchan=16
# Span 2
bchan=32-46,48-62
dchan=47
# Span 3
bchan=63-77,79-93
dchan=78
# Span 4
bchan=94-108,110-124
dchan=109
alaw=1-124

loadzone = nl
defaultzone=nl


zapata.conf:
[channels]
context=inbound
switchtype=euroisdn
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=no
rxgain=0.0
txgain=0.0
immediate=yes
; Span 1
group=1
context=inbound
signalling=pri_cpe
channel => 1-15
channel => 17-31
; Span 2
group=2
context=inbound
signalling=pri_cpe
channel => 32-46
channel => 48-62
; Span 3
group=3
context=outbound
signalling=pri_cpe
channel => 63-77
channel => 79-93
; Span 4
group=4
context=outbound
signalling=pri_cpe
channel => 94-108
channel => 110-124
and
extensions.conf:
[general]
static=yes
writeprotect=yes
...
[inbound]
exten => 1,1,Ringing
exten => 1,2,Wait,2
exten => 1,3,Playback(beep)
exten => 1,4,Playback(agent-alreadyon)
exten => _XXX,1,Ringing
exten => _XXX,2,Wait,2
exten => _XXX,3,Playback(beep)
exten => #,1,Hangup

...

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What is the need for the callerid in the following configuration:
(default config page)
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
;callerid="Uniden Dead" <(256) 428-6125>
;channel => 5
;callerid="Cortelco 2500" <(256) 428-6126>
;channel => 6
;callerid="Main TA 750" <(256) 428-6127>
;channel => 44
It is the mapping between PTSN nr and channel numbers?
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Re: [Asterisk-Users] oh323 problem (small one)

[carlos del mayor]

can you please send your oh323.conf and your
zapata.conf?

cmayor

 --- Dave Alan Caruana <[EMAIL PROTECTED]> escribió: >
I have just compiled & installed the latest oh323,
> on a fresh asterisk
> installation
> however using a previously working oh323.conf file.
> 
> When I try to dial an outbound oh323 call I get the
> following error :
> 
> -- Going to extension s|1 because of
> immediate=yes
> -- Executing Wait("Zap/1-1", "1") in new stack
> -- Accepting call from '21382890' to 's' on
> channel 1, span 1
> -- Executing Dial("Zap/1-1",
> "OH323/[EMAIL PROTECTED]") in new stack
>   1:04.782  ThreadID=0x4958a540 H323   
> Attempt to use invalid URL
> "[EMAIL PROTECTED]:1720"
> -- Couldn't call [EMAIL PROTECTED]
> -- Hungup 'H323:0'
>   == Everyone is busy at this time
> -- Executing Hangup("Zap/1-1", "") in new stack
>   == Spawn extension (incoming, s, 3) exited
> non-zero on 'Zap/1-1'
> -- Hungup 'Zap/1-1'
> 
> my extensions.conf is :
> 
> [incoming]
> exten => s,1,Wait,1
> exten => s,2,Dial(OH323/[EMAIL PROTECTED])
> exten => s,3,Hangup
> 
> I can't see anything obviously wrong, and dialling
> that h323 address from
> SJphone
> works fine (and used to work fine from asterisk too
> before).
> 
> help please :)
> 
> Dave
> 
> 
> 
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Re: [Asterisk-Users] RTP.C codec error 19

[Lord Stroud]

Hi Dave,

  The RTP codec 19 error that you are getting indicates that your endpoint is
most probably activating silence supression, and that you are using a codec
such as g.729, at least, that is what I get on my platform here.

  You can go into the the rtc.c file, and simply comment out the message.
Edit the rtp.c file at line 330, as the following:

ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);

  simply edit it to be:

//ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);

  and simply re-compile.

Nir S

On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote:
> hi ..
> when placing a SIP call to a sip host in the states
> every few seconds I get an RTP codec 19 error.
> I know this is related to comfort noise, and the
> call goes through OK ... how can I suppress
> the error message ?
>
> Also, many times I get "Invalid CSeq Number"
> back from 216.52.153.207 (which is the host
> i'm calling) and the call drops.. is there a solution
> for this ?
>
> cheers
> Dave
>
> (I mistakenly put an "re" in the title of this email
>  and I think it's been ignored .. reposted)
>
>
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Re: [Asterisk-Users] Agent in new CVS

[Mark Spencer]

Not that I know of, but I'd be interested in hearing more.  I've just
fixed a potential seg in the queue app this morning (thanks to TClark for
providing me a machine to track it down on), although I don't know if
that's related to your problem.

can you contact me off-list with more details?

Mark

On Tue, 8 Jul 2003, John Congdon wrote:

> I installed the 7/7/03 CVS today, and my customer service reps
> said that there were problems.  So I went back to an earlier version.
>
> They could log in, but when they received a call, they hear
> the beep, but not the announcement or call.
>
> I am using macros, if night is on, I dial the 800# that is passed.
> Otherwise I play the Thank You (ty_pn), and I also pass the
> announcement that the agent hears (pillnetwork).
>
> Is this something in the latest CVS that may be a problem?
>
> ===
>
> exten => 1086,1,Macro(enqueue,ty_pn,pillnetwork,18009159222)
>
> [macro-enqueue]
> exten => s,1,Wait,1
> exten => s,2,DBGet($Night=GlobalSettings/Night)
> exten => s,3,GotoIf($[${ARG3}]?4:5)
> exten => s,4,Dial(Zap/g1/${ARG3}|120|t)
> exten => s,5,VoiceMail,u300
> exten => s,103,PlayBack,${ARG1}
> exten => s,104,PlayBack,please_hold
> exten => s,105,Queue(PillNetwork|t||${ARG2})
>
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[Asterisk-Users] oh323 problem (small one)

[Dave Alan Caruana]

I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.

When I try to dial an outbound oh323 call I get the following error :

-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's' on channel 1, span 1
-- Executing Dial("Zap/1-1", "OH323/[EMAIL PROTECTED]") in new stack
  1:04.782  ThreadID=0x4958a540 H323Attempt to use invalid URL
"[EMAIL PROTECTED]:1720"
-- Couldn't call [EMAIL PROTECTED]
-- Hungup 'H323:0'
  == Everyone is busy at this time
-- Executing Hangup("Zap/1-1", "") in new stack
  == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'

my extensions.conf is :

[incoming]
exten => s,1,Wait,1
exten => s,2,Dial(OH323/[EMAIL PROTECTED])
exten => s,3,Hangup

I can't see anything obviously wrong, and dialling that h323 address from
SJphone
works fine (and used to work fine from asterisk too before).

help please :)

Dave



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[Asterisk-Users] RTP.C codec error 19

[Dave Alan Caruana]

hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?

Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is the host
i'm calling) and the call drops.. is there a solution
for this ?

cheers
Dave

(I mistakenly put an "re" in the title of this email
 and I think it's been ignored .. reposted)


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Re: [Asterisk-Users] Answering on an zap device

[Steven Critchfield]

First off, a E1 circuit is 32 channels, 1-32, 33-64, 65-96, 97-128. Also
there is a D channel on 16 and 32 of each span. Then you need to add the
D channel definition to your zapata.conf files. 

I believe all this was covered in the examples.



On Tue, 2003-07-08 at 08:20, Cristi wrote:
> How can I accept calls on a Wildcard E400P  . Please include the 
> zaptel.conf , zapata.conf and extension.conf to fully understand 
> everything.
> 
> Take a look firts to my configuration file. Where I did wrong?
> 
> zaptel.conf:
> span=1,1,0,ccs,hdb3,crc4
> span=2,0,0,ccs,hdb3
> span=3,0,0,ccs,hdb3
> span=4,0,0,ccs,hdb3
> 
> # Span 1
> bchan=1-15,17-31
> dchan=16
> 
> # Span 2
> bchan=32-46,48-62
> dchan=47
> 
> # Span 3
> bchan=63-77,79-93
> dchan=78
> 
> # Span 4
> bchan=94-108,110-124
> dchan=109
> 
> alaw=1-124
> 
> loadzone = nl
> defaultzone=nl
> 
> 
> 
> 
> zapata.conf:
> [channels]
> context=inbound
> switchtype=euroisdn
> signalling=pri_cpe
> ;rxwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> echocancel=no
> rxgain=0.0
> txgain=0.0
> immediate=yes
> 
> ; Span 1
> group=1
> context=inbound
> signalling=pri_cpe
> channel => 1-15
> channel => 17-31
> 
> ; Span 2
> group=2
> context=inbound
> signalling=pri_cpe
> channel => 32-46
> channel => 48-62
> 
> ; Span 3
> group=3
> context=outbound
> signalling=pri_cpe
> channel => 63-77
> channel => 79-93
> 
> ; Span 4
> group=4
> context=outbound
> signalling=pri_cpe
> channel => 94-108
> channel => 110-124
> 
> 
> and
> extensions.conf:
> [general]
> static=yes
> writeprotect=yes
> 
> ...
> [inbound]
> 
> exten => 1,1,Ringing
> exten => 1,2,Wait,2
> exten => 1,3,Playback(beep)
> exten => 1,4,Playback(agent-alreadyon)
> 
> exten => _XXX,1,Ringing
> exten => _XXX,2,Wait,2
> exten => _XXX,3,Playback(beep)
> 
> 
> exten => #,1,Hangup
> 
> ...
> 
> 
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[Asterisk-Users] Agent in new CVS

[John Congdon]

I installed the 7/7/03 CVS today, and my customer service reps
said that there were problems.  So I went back to an earlier version.
They could log in, but when they received a call, they hear
the beep, but not the announcement or call.
I am using macros, if night is on, I dial the 800# that is passed.
Otherwise I play the Thank You (ty_pn), and I also pass the
announcement that the agent hears (pillnetwork).
Is this something in the latest CVS that may be a problem?

===

exten => 1086,1,Macro(enqueue,ty_pn,pillnetwork,18009159222)

[macro-enqueue]
exten => s,1,Wait,1
exten => s,2,DBGet($Night=GlobalSettings/Night)
exten => s,3,GotoIf($[${ARG3}]?4:5)
exten => s,4,Dial(Zap/g1/${ARG3}|120|t)
exten => s,5,VoiceMail,u300
exten => s,103,PlayBack,${ARG1}
exten => s,104,PlayBack,please_hold
exten => s,105,Queue(PillNetwork|t||${ARG2})
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[Asterisk-Users] Answering on an zap device

[Cristi]

How can I accept calls on a Wildcard E400P  . Please include the 
zaptel.conf , zapata.conf and extension.conf to fully understand 
everything.

Take a look firts to my configuration file. Where I did wrong?

zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3
# Span 1
bchan=1-15,17-31
dchan=16
# Span 2
bchan=32-46,48-62
dchan=47
# Span 3
bchan=63-77,79-93
dchan=78
# Span 4
bchan=94-108,110-124
dchan=109
alaw=1-124

loadzone = nl
defaultzone=nl


zapata.conf:
[channels]
context=inbound
switchtype=euroisdn
signalling=pri_cpe
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=no
rxgain=0.0
txgain=0.0
immediate=yes
; Span 1
group=1
context=inbound
signalling=pri_cpe
channel => 1-15
channel => 17-31
; Span 2
group=2
context=inbound
signalling=pri_cpe
channel => 32-46
channel => 48-62
; Span 3
group=3
context=outbound
signalling=pri_cpe
channel => 63-77
channel => 79-93
; Span 4
group=4
context=outbound
signalling=pri_cpe
channel => 94-108
channel => 110-124
and
extensions.conf:
[general]
static=yes
writeprotect=yes
...
[inbound]
exten => 1,1,Ringing
exten => 1,2,Wait,2
exten => 1,3,Playback(beep)
exten => 1,4,Playback(agent-alreadyon)
exten => _XXX,1,Ringing
exten => _XXX,2,Wait,2
exten => _XXX,3,Playback(beep)
exten => #,1,Hangup

...

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Re: [Asterisk-Users] Asterisk and VMWare

[Steven Critchfield]

I use a similar asterisk setup when my T100P card is being used outside
the home. I ran it on a 1ghz athlon that was shared as my workstation,
but all linux. When my screen saver started, the audio quality dropped
below usable. So obviously my system was idle if the screen saver
started. SO as you can see you most likely will not get good results
under vmware even if you are on a pretty hefty machine.
 

On Tue, 2003-07-08 at 01:43, Dan wrote:
> Hi Gary,
> 
> This possibility is excluded because the Home Automation framework runing on
> WinXP now needs to have direct access to a couple of proprietary hardware
> devices.
> I think to have a home Asterisk box on the same computer without any
> specific hardware, just to be able to use a couple of SIP phones (Cisco 7960
> and ATA 186) without GSM support.
> The PSTN connection is made at the office, through another Asterisk box
> connected to the home one using IAX and GSM as codec.
> 
> BR,
> Dan
> 
> - Original Message - 
> From: "Gary Gapinski" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, July 07, 2003 10:55 PM
> Subject: Re: [Asterisk-Users] Asterisk and VMWare
> 
> 
> > On Monday 07 July 2003 15:26, Dan wrote:
> > > The reason I ask this is because I have a Win2K PC running 24/7 which
> > > has enough power left, but if I cannot use any of the Digium hardware
> > > from inside VMWare then is useless.
> >
> > If you have not yet purchased the VMware license, run Linux, Asterisk,
> > VMware _for Linux_, and W2K within VMware (with the caveat that special
> > hardware will likely not be supported within VMs).
> >
> > Even if you have already purchased VMware for Microsoft Windows (the MS
> > and Linux licenses are not interchangeable), contact VMware regarding a
> > possible exchange.
> >
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> >
> >
> 
> 
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Re: [Asterisk-Users] Integratting * With Database(Newbie)

[Steven Critchfield]

This is where you need to read the docs.

Also use you need to check the archive. look at this URL.

http://tinyurl.com/gb7a



On Tue, 2003-07-08 at 00:19, God Knows Well wrote:
> Hi Steven
> 
> 
> Thanx for your reply . U said i can store data in Postgres would u plz me 
> tell the steps to Configure it ,as i am newbie i didnt understand properly 
> ur reply if u want i can send u the whole scenario.

> >On Sat, 2003-07-05 at 08:42, God Knows Well wrote:
> > > Hi
> > >
> > > I think * supports database integration i would be thankful if anyone 
> >help
> > > me to configure my asterisk box with database support. One more think 
> >can i
> > > store Log in database.
> >
> >asterisk supports a couple of things being integrated with databases,
> >but probably not what you are thinking about. Currently you can store
> >data and get to it via the PSQL application for integration with
> >postgres. Or you can use the Berkly DB code and the various DB commands.
> >CDR can be pushed out to a mysql database. From AGI you can connect to
> >any database you have linux code to support.
> >
> >The config files are flat files and there is no integration with a
> >database to make them available that way.
> >--
> >Steven Critchfield <[EMAIL PROTECTED]>
> >
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Re: [Asterisk-Users] three way calling and cisco ata 186

[Thomas Dingermann]

Pavel Zheltouhov wrote:

Ok, if this is not working with sip or h.323, maybe it does with mgcp ?
I tried to get ATA and Asterisk working with MGCP, but nothing worked!
Any Howtos available about MGCP/ATA186/Asterisk?
Thomas

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[Asterisk-Users] asterisk-oh323 v0.5.3

[Michael Manousos]

Hello all,

I have updated the asterisk-oh323 package. The new version
has several improvements (fixes in audio/RTP stream generation,
music-on-hold working, flash hook detection, more config options).
You can download it from:
http://www.inaccessnetworks.com/projects/asterisk-oh323

Feedback is always welcome.

Regards,
Michael.
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[Asterisk-Users] Patch to fix some segfaults in Asterisk

[Michael Manousos]

Hi,

This patch fixes a couple of segfaults in music-on-hold,
frame smoother routines and channel allocation in Asterisk.
Mark, feel free to apply it in CVS (if approved).
Regards,
Michael.

Index: channel.c
===
RCS file: /usr/cvsroot/asterisk/channel.c,v
retrieving revision 1.25
diff -u -r1.25 channel.c
--- channel.c   4 Jul 2003 16:49:11 -   1.25
+++ channel.c   8 Jul 2003 10:48:13 -
@@ -279,8 +279,8 @@
return NULL;
PTHREAD_MUTEX_LOCK(&chlock);
tmp = malloc(sizeof(struct ast_channel));
-   memset(tmp, 0, sizeof(struct ast_channel));
if (tmp) {
+   memset(tmp, 0, sizeof(struct ast_channel));
pvt = malloc(sizeof(struct ast_channel_pvt));
if (pvt) {
memset(pvt, 0, sizeof(struct ast_channel_pvt));
Index: frame.c
===
RCS file: /usr/cvsroot/asterisk/frame.c,v
retrieving revision 1.3
diff -u -r1.3 frame.c
--- frame.c 28 Jun 2003 16:40:02 -  1.3
+++ frame.c 8 Jul 2003 10:48:15 -
@@ -125,8 +125,10 @@
/* Make frame */
s->f.frametype = AST_FRAME_VOICE;
s->f.subclass = s->format;
-   s->f.data = s->framedata + AST_FRIENDLY_OFFSET;
-   s->f.offset = AST_FRIENDLY_OFFSET;
+   /*s->f.data = s->framedata + AST_FRIENDLY_OFFSET;*/
+   s->f.data = s->framedata;
+   /*s->f.offset = AST_FRIENDLY_OFFSET;*/
+   s->f.offset = 0;
s->f.datalen = s->size;
s->f.samples = s->size * s->samplesperbyte;
/* Fill Data */
@@ -141,7 +143,8 @@
 
 void ast_smoother_free(struct ast_smoother *s)
 {
-   free(s);
+   if (s != NULL)
+   free(s);
 }
 
 static struct ast_frame *ast_frame_header_new(void)
Index: apps/Makefile
===
RCS file: /usr/cvsroot/asterisk/apps/Makefile,v
retrieving revision 1.8
diff -u -r1.8 Makefile
--- apps/Makefile   16 May 2003 20:57:12 -  1.8
+++ apps/Makefile   8 Jul 2003 10:48:15 -
@@ -35,7 +35,7 @@
 all: $(APPS)
 
 clean:
-   rm -f *.so *.o look
+   rm -f *.so *.o look .depend
 
 %.so : %.o
$(CC) -shared -Xlinker -x -o $@ $<
Index: res/res_musiconhold.c
===
RCS file: /usr/cvsroot/asterisk/res/res_musiconhold.c,v
retrieving revision 1.1.1.5
diff -u -r1.1.1.5 res_musiconhold.c
--- res/res_musiconhold.c   18 Mar 2003 06:00:18 -  1.1.1.5
+++ res/res_musiconhold.c   8 Jul 2003 10:48:30 -
@@ -400,7 +400,8 @@
ast_log(LOG_WARNING, "Only doing %d of %d requested bytes on %s\n", 
sizeof(buf), len, chan->name);
len = sizeof(buf);
}
-   res = read(moh->pipe[0], buf + AST_FRIENDLY_OFFSET/2, len);
+   /* XXX SEGFAULT res = read(moh->pipe[0], buf + AST_FRIENDLY_OFFSET/2, len);*/
+   res = read(moh->pipe[0], buf, len);
 #if 0
if (res != len) {
ast_log(LOG_WARNING, "Read only %d of %d bytes: %s\n", res, len, 
strerror(errno));
@@ -413,7 +414,8 @@
f.mallocd = 0;
f.datalen = res;
f.samples = res / 2;
-   f.data = buf + AST_FRIENDLY_OFFSET / 2;
+   /* XXX SEGFAULT f.data = buf + AST_FRIENDLY_OFFSET / 2;*/
+   f.data = buf;
f.offset = AST_FRIENDLY_OFFSET;
if (ast_write(chan, &f)< 0) {
ast_log(LOG_WARNING, "Failed to write frame to '%s': %s\n", 
chan->name, strerror(errno));

Re: [Asterisk-Users] chanh323 dialling

[carlos del mayor]

 --- Dave Alan Caruana <[EMAIL PROTECTED]> escribió: > 

>what is the format for an h323 entry in the
> dialplan?

[o]H323/@

for example:
exten=>21,1,dial(H323/[EMAIL PROTECTED])

when you dial this extension Asterisk call to the
extension 11 on the host 192.192.192.192.

> can I use chan_h323 without compiling anything else

Yes, you can, you don't need anything else

> or should I compile oh323?

Only use ONE of them, h323 or oh323. If you want to
use oh323 you must compile it.

> 
> basically what's the best way :)

Both goes ok, it's your decision.

> 
> cheers
> Dave
>
regards
cmayor
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Re: [Asterisk-Users] three way calling and cisco ata 186

[Pavel Zheltouhov]

Thomas Dingermann wrote:
I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and
asterisk  as pbx. I need feature called as 'three way calling' or 
'transfer with consultation'. Registering,calling and 'blind transfer' 
work fine.
Same here - and if you hang up, the call is not transferred...
Something goes wrong when using the Hook-Flash with ATA/Asterisk.
Any solutions out there?
Ok, if this is not working with sip or h.323, maybe it does with mgcp ?

--
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[Asterisk-Users] re. rtp.c RTP codec 19

[Dave Alan Caruana]

hi ..
when placing a SIP call to a sip host in the states
every few seconds I get an RTP codec 19 error.
I know this is related to comfort noise, and the
call goes through OK ... how can I suppress
the error message ?

Also, many times I get "Invalid CSeq Number"
back from 216.52.153.207 (which is the host
i'm calling) and the call drops.. is there a solution
for this ?

cheers
Dave


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[Asterisk-Users] ECHO on sip- call

[Ing. Angel Gomez Garcia]

   Hi all.

   Just got my 'Developer Kit Lite', installed it, and made the changes 
to load the modules in kernel and in the configuration files. Call 
thru-from fxo and the fxs sound great. Even fxs-iax-sip sound ok.

   When a answer a call coming into asterisk from the PSTN thru the fxo 
i have a loud echo. I have  echocancel=yes.

   Is there another parameter I can change in the configuration files ? 
or do I have to change in... don't remember the name of the source 
program, the 'echo cancellation' to strong and recompile asterisk again?

   Thank's.

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[Asterisk-Users] chanh323 dialling

[Dave Alan Caruana]

what is the format for an h323 entry in the dialplan?
can I use chan_h323 without compiling anything else
or should I compile oh323?

basically what's the best way :)

cheers
Dave


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Re: [Asterisk-Users] Transfert call

[carlos del mayor]

Sorry

I did a mistake!!
the correct way for the extension line is:
exten =>111,1,dial,Zap/1|20|T
or
exten =>111,1,dial(Zap/1,20,T)

I did a mix...

cmayor


 --- Rattana BIV <[EMAIL PROTECTED]> escribió: > Hi,
> 
> 
> A question about transfert.
> 
> How can I make transfert with the the person who
> call.
> X call Z and X transfert Z to Y.
> I only succeed to do X call Z and Z transfert to Y.
> 
> If someone have a solution it will be very good =)
> 
> 
> regards
> Rattana 

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Re: [Asterisk-Users] asterisk and uclinux

[Marian Danisek]

> Hello,every one! 
> I would like to know if asterisk could run under uclinux.
look at the archives in jun-2003, we spoke about how to compile in under
uclibc - there is a patch.

but i personaly had problem to run it, because of i cannot run asterisk
a as a daemon ( fork ).. i thing beacuse uClinux can only do vfork().  

cheers

Marian


> Regards.
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Re: [Asterisk-Users] Transfert call

[carlos del mayor]

Hi Rattana,

That kind of transfer is not yet implemented in *. The
way it will be indicated is:
exten =>111,dial,Zap/1,20,T

The T indicate that transfer is permitted for calling
party, but as I've said, that's not implemented at the
moment.

Regards
cmayor

 --- Rattana BIV <[EMAIL PROTECTED]> escribió: > Hi,
> 
> 
> A question about transfert.
> 
> How can I make transfert with the the person who
> call.
> X call Z and X transfert Z to Y.
> I only succeed to do X call Z and Z transfert to Y.
> 
> If someone have a solution it will be very good =)
> 
> 
> regards
> Rattana 

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[Asterisk-Users] Transfert call

[Rattana BIV]

Hi,
 
 
A question about transfert.
 
How can I make transfert with the the person who 
call.
X call Z and X transfert Z to Y.
I only succeed to do X call Z and Z transfert to 
Y.
 
If someone have a solution it will be very good 
=)
 
 
regards
Rattana

Re: [Asterisk-Users] msn

[Masakazu Nakano]

On Tue, 8 Jul 2003 14:23:51 +0800
"Kelvin Chua" <[EMAIL PROTECTED]> wrote:

>hi guys,
>
>have any of you guys managed to use msn messenger to authenticate with asterisk using 
>its DNS name? based on my experience with other sip proxies, msn will not 
>authenticate if it sees a different realm than the realm of the client. one 
>workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also 
>edit the Contact: field. after this, asterisk would send a 401 to the register 
>message, at this point, i'm quite stuck. i also noticed that the nonce field is 
>randdata? compared to iptel.org's ser and vovida's vocal. i notice a lot of 
>difference on the sip messages composition. i'm running 0.4.0. 

I think maybe msn has one of heart-beat feature.

When it cannot get result of icmp, send bye to sip server by itself.

require server_addr and asterisk (realm) with your hosts file in your windows.

---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan.
http://www.dairiten.com/modules/news/
powered by xoops at http://www.xoops.org

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RE: [Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)

[Adam Goryachev]

> The problems I experienced is that when the link between the two
> servers is down the local server becomes highly unreliable..
>
> Firstly when dialing an outside line it attempts to search the
> remote (at this time unavailable) dialplan before it has
> processed all the local options, This causes a massive delay
> between the time the call is dialed and the time the call
> actually starts the connection.. I have tried qualify=yes in the
> iax.conf but this did not alter the call processing routine in
> any way or reduce the time it takes from the time the number is
> dailed to the time it finds the extention mapping it needs to use..

Perhaps something similar to the qualify for SIP clients could be
implemented with IAX and IAX2 connections, ie, if ping time is greater than
qualify time, then don't use it.

It would also help if periodically asterisk would collect the extensions
list for each switch statement included. This might cause minor problems if
your extensions change frequantly (perhaps rather than just a periodic check
include support to 'notify' remote servers when the local dialplan is
changed like the BIND DNS support for zone files).

I have only just started using IAX connections over a 64k ISDN link, and it
is bearable as long as there is no other traffic using the link at the same
time... I am yet to really play with it and make it work fantastically.


Regards,
Adam

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FW: [Asterisk-Users] ATA 186 in Australia

[Adam Goryachev]

The details for the Australian cisco ATA186 are below:

> -Original Message-
> From: Tony Du [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, 8 July 2003 4:31 PM
> To: 'Adam Goryachev'
> Subject: RE: [Asterisk-Users] ATA 186 in Australia
>
> Hi Adam,
>
> I sold a Cisco ATA186 I1 2 port adaptor (Cisco code:
> SW-SMH-UL-ATA-2P)to you on 16/10/02)
>
> Cheers
> Tony Du
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Adam
> > Goryachev
> > Sent: Tuesday, 8 July 2003 1:22 PM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] ATA 186 in Australia
> >
> >
> > Talk to Tony from Action Computers in Sydney (02) 9281 3988 and
> tell him I
> > sent you, or else LAN Systems (Denis Valente) or any Tech Pacific
> > reseller...
> >
> > I bought one from Tony, but I don't remember which one I ended up
> > with. The
> > easy way to tell is that one model is 'in stock' while the other was a 8
> > week lead time for me...
> >
> > Regards,
> > Adam
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] Behalf Of
> Steven Honson
> > > Sent: Tuesday, 8 July 2003 9:40 AM
> > > To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] ATA 186 in Australia
> > >
> > >
> > > Hi All,
> > >
> > > I'm looking at setting up a Asterisk system, and hope to use ATA 186's
> > > with it.
> > > Im in Australia, and am getting mixed answers to if its the I1 or I2 i
> > > need, does anyone have any experience with using ATA 186's in
> Australia
> > > Also, can anyone recommend a good place to obtain these locally?

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[Asterisk-Users] Conferences with CAPI and H323

[Rattana BIV]

Hi;
 
I try to work with conferences room by MeetMe and 
ztdummy modules.
 
I notice that Conference with phones work well. But 
conference with H323 terminals work but there no sound. And conferences between 
H323 and capi only CAPI can talk and hear something.
 
Does anyone has successfully make this 
?
 
 
 
Regards
Rattana

[Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)

[WipeOut .]

Hi,

NOTE: This mail is sent for general information based on my experience to the Asterisk 
community not as an attack on Asterisk which I think is an amazing system..

Some of you may have seen my posting a few days ago where I was having problems with 
using the "switch" command in my dialplan..

It was suggested to put the switch command into its own [context] and then to include 
it into the contexts where it was required.. While using switch in the context that 
contains all my extensions or using it in a seperate context and then including it 
both worked fine on my setup this had not solved the issue I was having. After more 
testing I would simply advise that anyone who does not have a dedicated (and somewhat 
reliable) link between their two systems simply don't use the switch command and 
rather configure a wildcard extension mapping to the remote Asterisk system..

The problems I experienced is that when the link between the two servers is down the 
local server becomes highly unreliable.. 

Firstly when dialing an outside line it attempts to search the remote (at this time 
unavailable) dialplan before it has processed all the local options, This causes a 
massive delay between the time the call is dialed and the time the call actually 
starts the connection.. I have tried qualify=yes in the iax.conf but this did not 
alter the call processing routine in any way or reduce the time it takes from the time 
the number is dailed to the time it finds the extention mapping it needs to use..

The second issue is that when the line is down and an outbound call is placed (on my 
local system it is using an X100P) the call will terminate after approx 1-2 mins for 
no apparent reason.. I can only assume this has somthing to do with Asterisk trying to 
test connectivity with the remote system or somthing else to do with the switch 
command becasue when I comment out the switch command and use wildcard extension 
mappings the problem is gone..

so effectively what I have done is replaced this..
switch => IAX2/...

with this..
exten => _2xxx,1,IAX2/...
etc..

Now I am sure many of you are going to respond and tell me that my dialplan is wrong 
or that you are using switch and its perfect.. Thats all fine and I agree that switch 
is perfect when the link between the two systems is up.. All I am saying is that one 
day when you link is down for whatever reason and suddenly your local pbx starts to 
freak out and has long delay when placing a call or disconnects you constantly then 
rememebr my experience.. It will save you many hours of stress..

Later..


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