RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

>On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote:
>> I have configured the voicemail.conf file as per the wiki to email
>> voicemails as an attachment. I cannot find any instructions/locations to
>> set the outgoing server login information. Furthermore, I can get no
>> emails from asterisk. Can anyone point me to the next step to setup the
>> attachment of voicemail messages to an email?
>
>Set up a "sendmail". Or basically: an MTA. Any linux distro comes with
>at least one (postfix seems to be the preffered choice nowadays). Which
>one do you use?
>
>There are a bunch of programs that provide /usr/sbin/sendmail but don't
>spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are
>probably others.
>
>The downside is that messages that have, for some reason, not been
>delivered in the first shot (e.g: due to some transient network error)
>will be dropped rather than queued.


I was playing with mta, but this is so complicated, specially if you are on
dynamic ip address, so it is much easier to use smtp for sending mails..

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[Asterisk-Users] Re: SNOM 360 Unknown SIP command 'PUBLISH'

[Shanon Swafford]

Without seeing the actual SIP Message.  I'm guessing it is "Number Guessing".  
It is on default on Snom phones.

Regards,
Shanon

<[EMAIL PROTECTED]> wrote in message news:<[EMAIL PROTECTED]>...
> Hi List
> 
>  
> 
> I'm getting this notification from my one and only SNOM 360 every time 
> a number button is pushed.
> 
> I know that it's only a notification, but it really irritates me. Is 
> it anything I can/should do anything about ??
> 
>  
> 
> Oct 12 10:34:33 NOTICE[3566]: chan_sip.c:10530 handle_request: Unknown 
> SIP command 'PUBLISH' from '192.168.100.100'
> 
>  
> 
>  
> 
> By the way I'm using * 1.0.9 CVS-HEAD September 15. 2005
> 
>  
> 
> Best regards
> 
>  
> 
> BennyBad
> 
> 


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RE: [Asterisk-Users] Voicemail as an email attachement

[Goran Skular]

Small.. just app_voicemail.c and a sendEmail script...

You can download it from here:

app_voicemail.c
http://www.migo-systems.com/index.php?option=com_remository&Itemid=11&func=f
ileinfo&id=9
and


sendEmail
http://www.migo-systems.com/index.php?option=com_remository&Itemid=11&func=f
ileinfo&id=10




sendEmail is most important.. code change is really small in app_voicemail..
but here it is..


1. install sendEmail

2. Edit app_voicemail.c :


You will need to change app_voicemail.c to suit your needs.. Go to line 1035
(or find goran.skular) and:

Change [EMAIL PROTECTED] to from address you want to show up

Mail.slsolucije.hr:25 change to your mail.server.xxx:smtp 

Password_here is place for your password..


Go to line 1130 also (or find next appereance of goran.skular) and to the
same again.


That's all in short.

Have a nice day.
 

>-Original Message-
>From: [EMAIL PROTECTED] [mailto:asterisk-users-
>[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
>Sent: Wednesday, October 19, 2005 4:27 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Voicemail as an email attachement
>
>Yes. I am interested. I will make provisions for the upload. How big are
>the files?
>
>Thanks
>
>BEN
>
>Goran Skular wrote:
>> I changed my app_voicemail.c to work not with sendmail but with sendEmail
>> that connects to any SMTP and sends email with attachment...
>>
>> It's dirty, but it works.
>>
>> If you are interested I can upload app_voicemail.c and sendEmail package
>> somewhere..
>>
>>
>>
>>>I have configured the voicemail.conf file as per the wiki to email
>>>voicemails as an attachment. I cannot find any instructions/locations to
>>>set the outgoing server login information. Furthermore, I can get no
>>>emails from asterisk. Can anyone point me to the next step to setup the
>>>attachment of voicemail messages to an email?
>>>
>>>Thanks
>>>
>>>BEN
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[Asterisk-Users] Dial Limit Call Options

[Alejandro G]

Hi,

Is there a way to know if after using the Dial command and specifying
L(X:Y:Z) option for limiting the duration of the call and if the calls
reachs that limit have an indication that the caller reachs the limit? (i.e.
DIALSTATUS)

Thanks


Alejandro Ghergherian

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Re: [Asterisk-Users] E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)

[steve]

On Wed, 19 Oct 2005, Dinesh Nair wrote:

> hey * folk,
> 
> i've got a TE410P (generation 1 firmware) stuck in a box with a single xeon
> 2.8Ghz and 1GB RAM. there's a loopback E1 cable connecting span 1 to span 4
> (zaptel.conf and zapata.conf below). upon starting up asterisk, i see the
> following errors consistently on the screen,
> 
> !! Got I-frame while link state 2
> !! Got a UA, but i'm in state 1
> 

This boils down to "I'm trying to start up the link, but the other side 
seems to think that it IS up".

In an earlier post you showed that you had some spans set for internal 
clocking, some for external.  If you are using loopback cables, I'd 
suggest setting all the spans for internal (X,0,0,ccs,hdb3[,crc4])

Make sure nothing else is connected on the PRI, just the loopback cable.  
And the loopback wiring is the pair on 1/2 crossed over to the pair on 
4/5.

> a snapshot of pri debug span 1, shows:

The part you posted is just where Asterisk is restarting each B-channel.  
More useful would be the part corresponding to the debug messages logged 
above.


> could it be due to a buggy card ? if this is the case, i really wont be
> able to tell as i dont have any spare cards to test with. i've read that
> the newer versions of the drivers may cause similar problems to old cards,
> but since we're on freebsd, we're unable to revert to an old version of the
> driver.

I'm not sure how old your card is, but I routinely run the CVS-HEAD 
zaptel/libpri/asterisk on generation 1, "version 10" cards.

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

Unusual order here.  I've always followed the sample and put the span 
definitions at the top and the bchan/dchan following.

Also, see the comment about the "1" for taking clock - rather make it 0 
for loopback tests.

(You've also asked for two primary sync sources, which isn't right).



> 
> --- zapata.conf ---
> [channels]
> signalling=pri_net
> context=default
> group=1
> callgroup=1
> pickupgroup=1
> priindication=outofband
> switchtype=euroisdn
> context=default
> amaflags=default
> busycount=4
> callwaiting=no
> transfer=yes
> useincomingcalleridonzaptransfer=yes
> threewaycalling=yes
> callreturn=yes
> relaxdtmf=no
> busydetect=no
> usecallerid=yes
> hidecallerid=no
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=no
> immediate=no
> channel => 1-15
> channel => 17-31
> 
> signalling=pri_cpe
> context=default
> group=1
> callgroup=1
> pickupgroup=1
> priindication=outofband
> switchtype=euroisdn
> context=default
> amaflags=default
> busycount=4
> callwaiting=no
> transfer=yes
> useincomingcalleridonzaptransfer=yes
> threewaycalling=yes
> callreturn=yes
> relaxdtmf=no
> busydetect=no
> usecallerid=yes
> hidecallerid=no
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=no
> immediate=yes
> channel => 94-108
> channel => 110-124
> --- zapata.conf ---


Did you leave out the other two spans in your zapata.conf?

Its a bit unusual to put both spans in the same group - calls to Zap/g1 
may use any channel on either of the spans.  Is that what you want?
And immediate=yes is probably not what you want for an ISDN link.

Steve

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Re: [Asterisk-Users] E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)

[Dinesh Nair]

On 10/20/05 08:30 Matthew Fredrickson said the following:
Have you tried this in a Linux machine or are you still trying this in  
your FreeBSD box?


we knew that'd be the first question someone would ask, hence we've already 
done the following:


1. swapped operating systems
2. swapped servers

the problem persists. this is something we've been grappling with over the 
last week.


but even so, what has the OS got to do with it ? there're many people using 
the TE4XXP cards on freebsd without a problem, so the quality of the 
drivers are not in question.


--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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+=+
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Re: [Asterisk-Users] Re: Asterisk Evening in Melbourne Australia!

[Howard Lowndes]

Is there any chance anyone could discuss my post under "Can someone 
please explain caller line identification".  I live in Albury so I have 
no chance of getting to ML tonight.



jurgen wrote:

Just a quick reminder - this is happening *TONIGHT*. Hope to see all
local Asteriskers come out (except PaulH, who went to great lengths to
avoid us this time).

jurgen

On 14/10/05, jurgen <[EMAIL PROTECTED]> wrote:


Hi all,

Come out come out! If you're involved in Asterisk and live around the
Melbourne area, please come out and join us for an evening of geeking
out with Asterisk, socialising and generally having fun.

Please note, people who have before, the venue has changed from last
time because it was invaded by an annoying DJ.

Date and time: Thursday October 20th at 7pm.
Location: Mitre Tavern: http://www.melbournepubs.com/v/487/

If it's a warm evening, we'll be outside in the courtyard, but if it's
not so warm, look for us inside. I'll bring along an old skool Telecom
9600 PABX phone and put it on the table. If anyone else has some
classic technology, bring it along for a laugh. We've been thinking
about doing a more geeky, less social evening as well, so we'll be
talking about that - plus whatever else everyone has been up lately.

Questions? Send them to [EMAIL PROTECTED], or give me a ring on 0415 276 127.

See you there!

...jurgen


--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.





--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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--
Howard.
LANNet Computing Associates - Your Linux people 
--
When you just want a system that works, you choose Linux;
When you want a system that works, just, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.

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Re: [Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

[John Daragon]

Voicomm User wrote:

Hello

Hardware: Eicon Diva 4BRI ISDN Card
Software : Asterisk : Asterisk CVS-v1-0-08/13/05-19:51:52
   Chan Capi: chan_capi-0.6

We are using an Eicon 4BRI ISDN Card here in Australia with Asterisk,
connected to 4 OnRamp services with Telstra.

There are 8 available channels, but after upgrading to latest capi
driver we notice that the box is not able to handle more than 2 calls
at the same time. An engaged signal is heard at the other end. After
this happens once, some calls fail even when all channels are free.
I don't see any messages on console for failes calls. Even when I turn
on 'capi debug' and 'set verbose 20'.

The telstra personnel have confirmed busy signal is sent out by the
PABX. But its bizarre not to see any messages. No error messages are
logged as well.

capi info :
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Contr3: 2 B channels total, 2 B channels free.
Contr4: 2 B channels total, 2 B channels free.

capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[isdn]
isdnmode=ptp ; Is this correct for Point to Point Mode?
msn=<8 digit local number>
group=1
incomingmsn=*
controller=1,2,3,4 ; there are 4 controllers
devices=2   ; should this be 8?
softdtmf=on
relaxdtmf=on
accountcode=
context=main-menu
echocancelold=yes
;echocancel=yes  ; Turning this on gives a error message each time a
call is terminated.
usecallerid=yes
callerid=asreceived
;echosquelch=1
;echotail=64
;callgroup=1
;pickupgroup=1


The syntax has changed a bit. Time was when the "devices=" line 
basically said "OK, that's this controller done with, let's commit that 
and start on the next one..."  With 0.6 (if I read it correctly) it goes :


[general]
.
.

[some_string]

group=1
isdnmode=did   <-- note this has changed  [DID/MSN]
incomingmsn=*
rxgain=1.0
txgain=0.8
controller=1
softdtmf=0
accountcode=
context=from-pstn
echosquelch=0
echocancel=yes
echotail=64
devices=2

[some_other_string]

group=1
isdnmode=did
incomingmsn=*
rxgain=1.0
txgain=0.8
controller=2
softdtmf=0
accountcode=
context=from-pstn
echosquelch=0
echocancel=yes
echotail=64
devices=2

Hope this helps...

jd


--

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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[Asterisk-Users] how to edit or delete calleridname in From URI

[Raymond Chen]

Dear all,

 

I would like to delete the
calleridname in the FROM URI so it will not forward to the gateway in SIP.
  I’ve tried everything available in SIP.conf but not able to do it.
 Please help.

 

[test]

type=friend

context=sip-in

setvar(CALLERIDNAME =
"")

callerid="123123123"
<123123123>

username=123123123

fromuser=123123123

fromdomain=xxx.xxx.xxx.xxx

secret=xx

host=xxx.xxx.xx.xxx

port=5060

nat=yes

canreinvite=no

 

Ray

 






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[Asterisk-Users] chan_capi-0.6 configuration Query with Eicon Diva 4BRI

[Voicomm User]

Hello

Hardware: Eicon Diva 4BRI ISDN Card
Software : Asterisk : Asterisk CVS-v1-0-08/13/05-19:51:52
   Chan Capi: chan_capi-0.6

We are using an Eicon 4BRI ISDN Card here in Australia with Asterisk,
connected to 4 OnRamp services with Telstra.

There are 8 available channels, but after upgrading to latest capi
driver we notice that the box is not able to handle more than 2 calls
at the same time. An engaged signal is heard at the other end. After
this happens once, some calls fail even when all channels are free.
I don't see any messages on console for failes calls. Even when I turn
on 'capi debug' and 'set verbose 20'.

The telstra personnel have confirmed busy signal is sent out by the
PABX. But its bizarre not to see any messages. No error messages are
logged as well.

capi info :
Contr1: 2 B channels total, 2 B channels free.
Contr2: 2 B channels total, 2 B channels free.
Contr3: 2 B channels total, 2 B channels free.
Contr4: 2 B channels total, 2 B channels free.

capi.conf

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[isdn]
isdnmode=ptp ; Is this correct for Point to Point Mode?
msn=<8 digit local number>
group=1
incomingmsn=*
controller=1,2,3,4 ; there are 4 controllers
devices=2   ; should this be 8?
softdtmf=on
relaxdtmf=on
accountcode=
context=main-menu
echocancelold=yes
;echocancel=yes  ; Turning this on gives a error message each time a
call is terminated.
usecallerid=yes
callerid=asreceived
;echosquelch=1
;echotail=64
;callgroup=1
;pickupgroup=1

Should I define config for each controller seperately? The card is
currently configured on a
TE mode with Point to Point. Any help appreciated.

cheers
-r
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Re: [Asterisk-Users] Can someone please explain caller line identification

[Brian May]

On Thu, Oct 20, 2005 at 10:09:01AM +1000, Howard Lowndes wrote:
> There is/was a patch to * that suggested that in /channels/chan_zap.c 
> the variable DEFAULT_CIDRINGS should be changed from 1 to 2 to suit 
> Australian conditions and I had this done and everything worked.

Sorry, this probably won't help you...

However, I don't think you need to change the source code, I think
you can do the same thing by setting:

sendcalleridafter=2

in /etc/asterisk/zapata.conf.

At least this works for me.

> Since I recompiled * I have lost inbound CLID recognition but have 
> gained the distinctive ring recognition ability which I previously 
> didn't have.

I would speculate that distinctive ring recognition broke CLID.

> I also have a Wait(2) at the start of the relevant amswering dial plan 
> as also recommended.

I have found I don't need this. The extension file does not appear to be
processed until caller-id is established. This would just add extra
delay the caller must wait until the phones start ringing.

In fact,if you enable callerid without callerid support on the telephone
line, I noticed the phones will ring 2+ times before asterisk will
detect it - as it is looking for callerid information that isn't there.

So I am curious - how long is it from when the phone line starts ringing
to when asterisk detects the call?

I have only had some minor issues with callerid and related stuff:

* on one phone it now indicates all phone calls are "OUT OF AREA".
The phone number appears OK. I can't work out where this message comes
from.

* one another cordless phone system, it registers when there is a
message waiting, but it never resets back to no message waiting, even
after the message has been deleted. (Note this is using a Sipura POTS
adaptor).

I have:

└─(13:21!505:%)── dpkg -l asterisk
──(Thu,Oct20)─┘
Desired=Unknown/Install/Remove/Purge/Hold
|
Status=Not/Installed/Config-files/Unpacked/Failed-config/Half-installed
|/ Err?=(none)/Hold/Reinst-required/X=both-problems (Status,Err:
uppercase=bad)
||/ Name   VersionDescription
+++-==-==-
ii  asterisk   1.0.9.dfsg.1-3 open source Private Branch Exchange (PBX)
-- 
Brian May <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] OT: Samsung DCS 70

[Tad Heckaman]

If its like a samsung DCS 400 system, its all programmed from the
phones. Perhaps the advanced dial planning is done through the serial
cable. I have a samsung DCS 400, and it has really wierd analog lines
(voicemail lines, so it sends CID as DTMF tones, and never hangs up,
but sends a 'c' DTMF digit to signal a hangup). I gave up trying to
integrate it in, now I just have a T1 and we dial DID's straight into
the asterisk box. 

My DCS is all analog, no T1 attached to it, nor does it have a T1 card.
So no CID or DID numbers, and we will get a 'bad' line every once in
awhile when one goes bad.

Good luck! 

HOn 10/19/05, Rod Bacon <[EMAIL PROTECTED]> wrote:
This may be a little off topic, but I'm hoping to find someone who knowssomething about integration with "legacy" phone systems, specifically a SamsungDCS 70.Our current service provider charges us a packet each time we want to make a
small change, so I want to avoid using them to completely reprogram the entiresystem when I front-end it with an Asterisk box over the Christmas break.Does anyone out there have any experience with this model system? I (think) I
have the correct software to re-configure it. I can get the password (anddial-in number) or access to the local RS-232 port.--==Rod BaconEmpowered Communications
Ground Floor, 102 York St. South MelbourneVictoria, Australia. 3205Phone: +613 99401600Fax: +613 99401650FWD:
512237  
ICQ: 5662270==___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
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Tad Heckaman
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RE: [Asterisk-Users] Digium TDM400P (11B) problems

[Jason Walker]

As an FYI - here is the output of my TDM400P:

Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)

 
I do not have newt installed on this machine, so zttool bombs. Just sending
this out as an example.

Here are my zap[tel|ata] conf files:

Zaptel.conf:
fxols=1
fxsls=3-4


[channels]
context=fxo1
signalling=fxs_ls
...
channel => 3

context=fxo1
signalling=fxs_ls
...
channel => 4

context=fxs1
signalling=fxo_ls
...
channel => 1


The ... Represents more information that may or may not be useful...I didn't
think it was necessary for this thread. And frankly, I have rambled on
enough ;)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip Murray
Sent: Wednesday, October 19, 2005 6:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Digium TDM400P (11B) problems

Hi Rich,

On 20/10/2005, at 2:42 PM, Rich Adamson wrote:
>> 
>>
>>
>> dmesg:
>> PCI: Found IRQ 12 for device :00:0a.0 Freshmaker version: 71 
>> Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO 
>> Module 1: Not installed Module 2: Not installed Adjusting gain Module 
>> 3: Installed -- AUTO FXO (NEWZEALAND mode) Found a Wildcard TDM: 
>> Wildcard TDM400P REV E/F (2 modules)
>>
>
> Assuming you copy/pasted the above accurately, the output from dmesg 
> is missing the stuff related to Module 4. You might also try zttool to 
> see what it thinks is going on.

That is all the output in dmesg, is there any other output in particular you
were expecting? It's zero-based so Module 3 is the 4th module on the card.

zttool doesn't report any alarms or anything of note. What does the loop
button do?

Cheers

Phil Murray
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RE: [Asterisk-Users] Dial 2 channels at onece: Not working anymore atCVS?

[Jason Walker]

What if you force a hangup between the two steps?

I have multiple destinations specified when my internal number is called at
work using similar syntax. All of the SIP and SCCP extensions dial based on
my setup - which again, is very similar to yours.

I do not use CVS HEAD on the production boxes...I am somewhat stuck with
1.0.9 at work. I don't know if that is the difference or not. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres Tello
Abrego
Sent: Wednesday, October 19, 2005 1:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Dial 2 channels at onece: Not working anymore
atCVS?

Version
smbserver*CLI> show  version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-20 01:13:39 UTC


exten => 1100,1,Dial(Zap/5,30,Ttr)
exten => 1100,n,Dial(Zap/8&Zap/5,30,Ttr)

Doesn´t dial zap/8 & zap/5...

* output:

 -- Nobody picked up in 3 ms
 -- Hungup 'Zap/5-1'
 -- Executing Dial("Zap/7-1", "Zap/8&Zap/5|30|Ttr") in new stack
 -- Called 8
Oct 19 20:33:55 WARNING[547]: chan_zap.c:1819 zt_call: Unable to ring 
phone: Device or resource busy
 -- Couldn't call 5


This USED to work, not so long ago...

Why is this not working anymore?
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[Asterisk-Users] Re: Asterisk Evening in Melbourne Australia!

[jurgen]

Just a quick reminder - this is happening *TONIGHT*. Hope to see all
local Asteriskers come out (except PaulH, who went to great lengths to
avoid us this time).

jurgen

On 14/10/05, jurgen <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> Come out come out! If you're involved in Asterisk and live around the
> Melbourne area, please come out and join us for an evening of geeking
> out with Asterisk, socialising and generally having fun.
>
> Please note, people who have before, the venue has changed from last
> time because it was invaded by an annoying DJ.
>
> Date and time: Thursday October 20th at 7pm.
> Location: Mitre Tavern: http://www.melbournepubs.com/v/487/
>
> If it's a warm evening, we'll be outside in the courtyard, but if it's
> not so warm, look for us inside. I'll bring along an old skool Telecom
> 9600 PABX phone and put it on the table. If anyone else has some
> classic technology, bring it along for a laugh. We've been thinking
> about doing a more geeky, less social evening as well, so we'll be
> talking about that - plus whatever else everyone has been up lately.
>
> Questions? Send them to [EMAIL PROTECTED], or give me a ring on 0415 276 127.
>
> See you there!
>
> ...jurgen
>
>
> --
> [EMAIL PROTECTED] is jurgen's gmail address.
> Visit http://jurgen.ca/ for more yummy goodness.
>


--
[EMAIL PROTECTED] is jurgen's gmail address.
Visit http://jurgen.ca/ for more yummy goodness.
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Re: [Asterisk-Users] Can IAX be used without going thre a IAX server

[Chadwick E. Labno]

Thanks to all who helped with this problem, I'm now heading in the right 
direction

Chad



Chadwick E. Labno wrote:
 


Is it possible to route a call from an Asterisk box through the
Internet to a IAX device (in this case Digium IAXy) without
using an IAX service like IAXTel? I have it working on my
local Ethernet LAN so it should be possible to use VPN to
cross the internet. Anyone using VPN or other method to
acomplish this?
   



 


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[Asterisk-Users] Dial 2 channels at onece: Not working anymore at CVS?

[Andres Tello Abrego]

Version
smbserver*CLI> show  version
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 
2005-10-20 01:13:39 UTC



exten => 1100,1,Dial(Zap/5,30,Ttr)
exten => 1100,n,Dial(Zap/8&Zap/5,30,Ttr)

Doesn´t dial zap/8 & zap/5...

* output:

-- Nobody picked up in 3 ms
-- Hungup 'Zap/5-1'
-- Executing Dial("Zap/7-1", "Zap/8&Zap/5|30|Ttr") in new stack
-- Called 8
Oct 19 20:33:55 WARNING[547]: chan_zap.c:1819 zt_call: Unable to ring 
phone: Device or resource busy

-- Couldn't call 5


This USED to work, not so long ago...

Why is this not working anymore?
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Re: [Asterisk-Users] Digium TDM400P (11B) problems

[Philip Murray]

Hi Rich,

On 20/10/2005, at 2:42 PM, Rich Adamson wrote:




dmesg:
PCI: Found IRQ 12 for device :00:0a.0
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Not installed
Adjusting gain
Module 3: Installed -- AUTO FXO (NEWZEALAND mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules)



Assuming you copy/pasted the above accurately, the output
from dmesg is missing the stuff related to Module 4. You might
also try zttool to see what it thinks is going on.


That is all the output in dmesg, is there any other output in  
particular you were expecting? It's zero-based so Module 3 is the 4th  
module on the card.


zttool doesn't report any alarms or anything of note. What does the  
loop button do?


Cheers

Phil Murray
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Re: [Asterisk-Users] zaptel.conf config for CAS signalling

[C F]

would you please share line 213 with us?

On 10/18/05, Matt Hess <[EMAIL PROTECTED]> wrote:
> I have a customer that needs to do cas signaling across a t1,esf span..
> it looks like this can be done but I'm not sure how as the documentation
> is very light in regards to cas.. it would appear that I need to use sf
> signaling but I get an error saying:
> $ ztcfg -vv
> Notice: Configuration file is /etc/zaptel.conf
> line 213: Unknown keyword 'sf'
>
> I've also tried the format suggested in zaptel.conf
>
> channel# => (etc.)
>
> but I continue to fail.. I'd love a few pointers here..
>
>
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[C F]

Darren Thanks for your reply to my problem with the same setup, I have
found the problem to be Telco related and had it fixed since. But not
before I tried a Mediatrix 1204 on that setup. It was then that I
ralized that the problem is with the telco.
The Mediatrix came in at around $600 + (for 4 ports) while the Adit
with the Digium card came in at $1100 + (for 8 ports). What make the
Adit attractive is that you can let the customers hook up fax machine
and credit card machines if they want, using FXS cards with the Adit.
This all came in as a very good expereince for me, it allowed me to
try out both setups. In general I think, that if its a small setup for
just a few POTS, and the Fax/CC machine can be connected directly to a
POTS and not Asterisk then solutions like the Adit should be avoided,
it makes the whole system much cheaper. If all the transcoding gets
done outside Asterisk (as is the case with using something like the
Mediatrix) then you should be able to hook up around 70-80 Sip phones
or ATAs with Asterisk using just a single P4, and 512 MB RAM. While
some have seen the same results with the TE110 that is only true to
the maximum of 23 channels, but not if you plan on using 70, 80
simultaneus channels.

When you have to deal with lots of POTS lines (12 to 16 range), it
still might be cheaper to use the Mediatrix than the Adit 600, since
with something like the Adit you will need a more powerfull machine,
while with external gateways you can get by with just a P4 (or
something as small as the mini ITX systems).

When comparing sound quality I got the same with both devices (actualy
as long as I had the Adit with the TE110 and the problem form the
Telco I had better quality with the Adit since Asterisk was doing a
great job with the Echo can).

In overall I would recommend using an external Sip gateway instead of
the Adit when trying to compete (like for small offices with just a
few extensions). But where more then just the FXO is needed (like
faxing/CC) then the Adit is the way to go.

On 10/18/05, Darren Wright <[EMAIL PROTECTED]> wrote:
>
> 8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running
> 1.0.9 and 1.2 (tried both)
>
>
> The echo is insurmountable.  I have tried everything, and the pots lines
> are clean.  If I go from an FXO on the Adit 600 straight to an FXS, I
> get no echo from an analog phone.
>
> I put an 128ms T1 echo canceller in between the adit and the TE110P, and
> the echo was still horrible.
>
> I finally disabled the Zapata echo cancellerand WHAMMO!  It's
> perfect now.
>
> The TE110P is on it's own IRQ.. and the machine has PLENTY of
> horsepower.
>
> Any ideas so I don't have to spend $1000 on an echo canceller?
>
> -Darren
>
>
>
>
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Re: [Asterisk-Users] Digium TDM400P (11B) problems

[Rich Adamson]

> I just got a new TDM11B card to replace an old X100P card and the FXS  
> port on it works great. However I can't get the FXO port to function.
> 
> Asterisk will answer an incoming call but I can only hear noise  
> (almost like a modem squelch) down the line, and Asterisk doesn't  
> seem to receive any DTMF tones either. I also can't dial out over the  
> FXO port, Asterisk reports it's connected to the SIP client but  
> nothing seems to happen, the card doesn't even send any DTMF tones.
> 
> Does this sound like a faulty card/module? or a configuration error?
> 
> Configuration data below:
> 
> hacienda:~# cat /etc/zaptel.conf
> fxols=1
> fxsls=4
> loadzone=nz
> defaultzone=nz
> 
> hacienda:~# cat /etc/asterisk/zapata.conf
> [channels]
> language=en
> 
> signalling=fxo_ls
> context=default
> group=1
> channel => 1
> 
> signalling=fxs_ls
> context=from-pstn
> callerid=asreceived
> group=2
> channel => 4
> 
> hacienda:~# cat /proc/zaptel/1
> Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"
> 
> 1 WCTDM/0/0 FXOLS
> 2 WCTDM/0/1
> 3 WCTDM/0/2
> 4 WCTDM/0/3 FXSLS
> 
> 
> dmesg:
> PCI: Found IRQ 12 for device :00:0a.0
> Freshmaker version: 71
> Freshmaker passed register test
> Module 0: Installed -- AUTO FXS/DPO
> Module 1: Not installed
> Module 2: Not installed
> Adjusting gain
> Module 3: Installed -- AUTO FXO (NEWZEALAND mode)
> Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules)

Assuming you copy/pasted the above accurately, the output
from dmesg is missing the stuff related to Module 4. You might
also try zttool to see what it thinks is going on.

I'd suggest calling digium support and let them lead you through
the diagnoses.

You can truly contact them through the Demo function contained
in the sample config files (over the Internet).


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Re: [Asterisk-Users] sixtel DID

[JP Carballo]

Yu Safin wrote:


has anybody tried to register with Sixtel to obtain a DID?
I signed up 9/27 and I am yet to receive my DID.
Also, how do I change my aix.conf to connect to Sixtel?
I have a userid and password but I don't have details about all the parameters.
 


Log in to your account at http://control.sixtel.net and click on DIDs
You should then see 3 sections:
Numbers assigned to you:
Telephone Numbers (DID)
and
Toll Free numbers

Click on the link under Telephone Numbers to get a DID assigned to you.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)

[Matthew Fredrickson]

Have you tried this in a Linux machine or are you still trying this in  
your FreeBSD box?


Matthew Fredrickson

On Oct 19, 2005, at 10:51 AM, Dinesh Nair wrote:
i've got a TE410P (generation 1 firmware) stuck in a box with a single  
xeon
2.8Ghz and 1GB RAM. there's a loopback E1 cable connecting span 1 to  
span 4
(zaptel.conf and zapata.conf below). upon starting up asterisk, i see  
the

following errors consistently on the screen,

!! Got I-frame while link state 2
!! Got a UA, but i'm in state 1

they seem to be coming from libpri.so.1 and the spans seem to be  
restarting
each other infinitely. i also get a number of the following messages  
from

chan_zap.so:

B-channel 0/6 restarted on span 1
B-channel 0/6 restarted on span 4
B-channel 0/7 restarted on span 1
B-channel 0/7 restarted on span 4
B-channel 0/8 restarted on span 1
B-channel 0/8 restarted on span 4
B-channel 0/9 restarted on span 1
B-channel 0/9 restarted on span 4

No D-channels available! Using Primary Channel 16 as D-channel anyway!
No D-channels available! Using Primary Channel 109 as D-channel anyway!

both spans show "Provisioned,Up, Active" in pri show span, and zttest  
shows

100% all the way.

a snapshot of pri debug span 1, shows:

> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 83]
> 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: 3 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated  
Channel (0) ]

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 0/0x0) (Terminator)
< Message type: RESTART ACKNOWLEDGE (78)
< [18 03 a9 83 83]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,  
Exclusive Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 84]
> 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: 4 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated  
Channel (0) ]

!! Got I-frame while link state 2
> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 83]
> 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: 3 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated  
Channel (0) ]

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 0/0x0) (Originator)
< Message type: RESTART (70)
< [18 03 a9 83 84]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0,  
Exclusive Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Terminator)
> Message type: RESTART ACKNOWLEDGE (78)
> [18 03 a9 83 84]
> 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: 4 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated  
Channel (0) ]

!! Got I-frame while link state 2
> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 85]
> 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: 5 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Reset

Re: [Asterisk-Users] goiax configuration help please

[Sergey Okhapkin]

Replace [goiax] with [87820]. Just replace the section name.

On Wed, 2005-10-19 at 20:21 -0400, Jim Duda wrote:


I saw the posting concerning goiax offering free DIDs.  I went ahead, 
created an account, and got myself a DID.

Who is goiax, and how can they be doing this for free?  It's nice, but 
how can they offer that?

I have outbound calling working from asterisk, to 800 numbers.

I cannot seem to get inbound calls working though.  I cannot figure out 
why.  I get a message from asterisk saying that asterisk rejected the 
call due to an authorization failure.  Asterisk reported a failure at a 
specific line of code in chan_iax2.c and it has to do with 
authentication.  I'm registered with goiax as I see the proper result in 
iax2 show registry.  So, I at least have the correct ID and SECRET.

Do these DIDs really work?

In my iax.conf, I have:

;
; GOIAX
;
register => 87820:@server1.goiax.com

[goiax]
context=home
type=friend
host=server1.goiax.com
auth=md5
username=87820
secret=
disallow=all
allow=ulaw
allow=gsm

In my extensions.conf, I have (2 entries since I wasn't sure which 
number would be used)

;
; Goiax
;
exten => ,1,AGI(MisterHouse.agi,"CallerID")
exten => ,2,Dial(${PHONES0}&${PHONES1}&${PHONES2},20,tr)
exten => ,3,Macro(voicemail,${PHONES0VM})
exten => ,4,Hangup
exten => ,103,Macro(voicemail,${PHONES0VM})
exten => ,104,Hangup

exten => 87820,1,AGI(MisterHouse.agi,"CallerID")
exten => 87820,2,Dial(${PHONES0}&${PHONES1}&${PHONES2},20,tr)
exten => 87820,3,Macro(voicemail,${PHONES0VM})
exten => 87820,4,Hangup
exten => 87820,103,Macro(voicemail,${PHONES0VM})
exten => 87820,104,Hangup

Can anyone see what I might be doing wrong?

Jim




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[Asterisk-Users] goiax configuration help please

[Jim Duda]

I saw the posting concerning goiax offering free DIDs.  I went ahead, 
created an account, and got myself a DID.


Who is goiax, and how can they be doing this for free?  It's nice, but 
how can they offer that?


I have outbound calling working from asterisk, to 800 numbers.

I cannot seem to get inbound calls working though.  I cannot figure out 
why.  I get a message from asterisk saying that asterisk rejected the 
call due to an authorization failure.  Asterisk reported a failure at a 
specific line of code in chan_iax2.c and it has to do with 
authentication.  I'm registered with goiax as I see the proper result in 
iax2 show registry.  So, I at least have the correct ID and SECRET.


Do these DIDs really work?

In my iax.conf, I have:

;
; GOIAX
;
register => 87820:@server1.goiax.com

[goiax]
context=home
type=friend
host=server1.goiax.com
auth=md5
username=87820
secret=
disallow=all
allow=ulaw
allow=gsm

In my extensions.conf, I have (2 entries since I wasn't sure which 
number would be used)


;
; Goiax
;
exten => ,1,AGI(MisterHouse.agi,"CallerID")
exten => ,2,Dial(${PHONES0}&${PHONES1}&${PHONES2},20,tr)
exten => ,3,Macro(voicemail,${PHONES0VM})
exten => ,4,Hangup
exten => ,103,Macro(voicemail,${PHONES0VM})
exten => ,104,Hangup

exten => 87820,1,AGI(MisterHouse.agi,"CallerID")
exten => 87820,2,Dial(${PHONES0}&${PHONES1}&${PHONES2},20,tr)
exten => 87820,3,Macro(voicemail,${PHONES0VM})
exten => 87820,4,Hangup
exten => 87820,103,Macro(voicemail,${PHONES0VM})
exten => 87820,104,Hangup

Can anyone see what I might be doing wrong?

Jim




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Re: [Asterisk-Users] TDMoE question

[Martin Vit]

TDMoE is useless. I've tested it on newer intel P4 machines with 2.4 and 
2.6 kernels. There is CPU peaks causing by TMDoE driver.
If you want pass modem data, try IAX u/alaw codec. In my environment it 
works great (switched lan)


trixter aka Bret McDanel wrote:

On Wed, 2005-10-19 at 10:43 +0100, Appan KH wrote:
  

You can use MPLS which takes care all the point you had mentioned.

appan kh



Not entirely, at least not as I understand MPLS.  MPLS will add a little
bit of data which is used to route the traffic, it doesnt deal with
encapsulating TDM data (say from a T1 or DS3 from a telco) and allowing
that to cross a data link.  So that still leaves the question of TDMoE
or not given that I need to optionally (and unknown beforehand) be able
to traffic modem data reliably.  


Unless you are talknig about using MPLS with TDMoE which doesnt answer
the actual question I had about has anyone tried it, does it work
reliably even at the faster modem speeds, etc.


  



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[Asterisk-Users] Digium TDM400P (11B) problems

[Philip Murray]

Hi,

I just got a new TDM11B card to replace an old X100P card and the FXS  
port on it works great. However I can't get the FXO port to function.


Asterisk will answer an incoming call but I can only hear noise  
(almost like a modem squelch) down the line, and Asterisk doesn't  
seem to receive any DTMF tones either. I also can't dial out over the  
FXO port, Asterisk reports it's connected to the SIP client but  
nothing seems to happen, the card doesn't even send any DTMF tones.


Does this sound like a faulty card/module? or a configuration error?

Configuration data below:

hacienda:~# cat /etc/zaptel.conf
fxols=1
fxsls=4
loadzone=nz
defaultzone=nz

hacienda:~# cat /etc/asterisk/zapata.conf
[channels]
language=en

signalling=fxo_ls
context=default
group=1
channel => 1

signalling=fxs_ls
context=from-pstn
callerid=asreceived
group=2
channel => 4

hacienda:~# cat /proc/zaptel/1
Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1"

   1 WCTDM/0/0 FXOLS
   2 WCTDM/0/1
   3 WCTDM/0/2
   4 WCTDM/0/3 FXSLS


dmesg:
PCI: Found IRQ 12 for device :00:0a.0
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Not installed
Module 2: Not installed
Adjusting gain
Module 3: Installed -- AUTO FXO (NEWZEALAND mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules)


Cheers
Phil Murray

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Re: [Asterisk-Users] Free DID's

[trixter aka Bret McDanel]

Because voip-info.com doesnt really list free providers as a seperate
category and I really hate wikis for some reason I decided to keep track
of these on my personal page.  If anyone has any others I would
appreciate them telling me offlist so that I can add them.  I am also
listing free software to use with this, I know I am missing a bunch, if
people feel like emailing me privately about those as well I will add
them.

I list some of the basic features (like e911 support that one offers,
locations offered for free, etc) as well, so you can choose a bit more
without going everywhere.  I would be really interested in more
european, austrialian and asian providers that give free service since I
have very little right now.

Thanks

http://www.0xdecafbad.com/Free-VoIP-Providers.html


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Can someone please explain caller line identification

[Howard Lowndes]

This is not a newbie question, and my problem may be related to 
Australia only or may be wider based.


I have a PSTN line that has Caller ID presentation enabled.

It used to work fine until recently, in as much as I could identify 
inbound CLID.


There is/was a patch to * that suggested that in /channels/chan_zap.c 
the variable DEFAULT_CIDRINGS should be changed from 1 to 2 to suit 
Australian conditions and I had this done and everything worked.


Recently I upgraded my kernel from 2.6.12 to 2.6.13 and did a clean 
recompile of * to suit, and to get the updated modules.  BTW, * is 
CVS-HEAD of about 15 Sept.


Since I recompiled * I have lost inbound CLID recognition but have 
gained the distinctive ring recognition ability which I previously 
didn't have.


I still have the Australian mod in the chan_zap.c file, but I now note 
that the documentation indicates that this variable only applies to 
outbound CLID and not inbound CLID, or that is how I am reading the comment:

/* Typically, how many rings before we should send Caller*ID */
/* #define DEFAULT_CIDRINGS 1
   this needs to be set to 2 for Australia */
#define DEFAULT_CIDRINGS 2

[the 3rd & 4th lines are my mod and are not in the original code]

I also have a Wait(2) at the start of the relevant amswering dial plan 
as also recommended.


I am not sure how CLID works technically, and the callerid.c code 
appears somewhat esoteric, so I would appreciate any assistance, esp 
from an Australian connection who has got inbound CLID working.


--
Howard.
LANNet Computing Associates - Your Linux people 
--
When you just want a system that works, you choose Linux;
When you want a system that works, just, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.

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Re: [Asterisk-Users] Free DID's

[Sergey Okhapkin]

Thanx! voip.callpacket.com didn't work before - "487 Too many Hoops" (double o:-) in response to REGISTER. That's why I used IP address. Looks like they fixed something:-)

On Wed, 2005-10-19 at 16:26 -0700, Thameem Ansari wrote:


For callpacket host try using voip.callpacket.com. This is a recent change they made and ser.callpacket.com will not work. But if you nslookup both the names pointing to same ip. They might be using some kind of virtual hosting on that name I think.

-Thameem



On 10/19/05, Sergey Okhapkin <[EMAIL PROTECTED]> wrote:

On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote:


On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote:
> 
Callpacket.com has a free plan (up to 100 mins/month outbound,
> unlimited inbound, free DID).

Do you have hints on using callpacket w/ Asterisk?





register => sipusername:[EMAIL PROTECTED]

[callpacket-out]
type=peer
username=sipusername
secret=sipsecret
fromuser=sipusername
host=ser.callpacket.com
dtmfmode=rfc2833

[callpacket-in]
type=user
host=ser.callpacket.com
dtmfmode=rfc2833 
context=from-callpacket


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[Asterisk-Users] OT: Samsung DCS 70

[Rod Bacon]

This may be a little off topic, but I'm hoping to find someone who knows 
something about integration with "legacy" phone systems, specifically a Samsung 
DCS 70.


Our current service provider charges us a packet each time we want to make a 
small change, so I want to avoid using them to completely reprogram the entire 
system when I front-end it with an Asterisk box over the Christmas break.


Does anyone out there have any experience with this model system? I (think) I 
have the correct software to re-configure it. I can get the password (and 
dial-in number) or access to the local RS-232 port.




--
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237   ICQ: 5662270
==
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Re: [Asterisk-Users] Free DID's

[Thameem Ansari]

For callpacket host try using voip.callpacket.com. This is a recent
change they made and ser.callpacket.com will not work. But if you
nslookup both the names pointing to same ip. They might be using some
kind of virtual hosting on that name I think.

-Thameem
On 10/19/05, Sergey Okhapkin <[EMAIL PROTECTED]> wrote:



  
  


On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote:

On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote:> 
Callpacket.com has a free plan (up to 100 mins/month outbound,> unlimited inbound, free DID).Do you have hints on using callpacket w/ Asterisk?



register => sipusername:[EMAIL PROTECTED]

[callpacket-out]
type=peer
username=sipusername
secret=sipsecret
fromuser=sipusername
host=ser.callpacket.com
dtmfmode=rfc2833

[callpacket-in]
type=user
host=ser.callpacket.com
dtmfmode=rfc2833 
context=from-callpacket




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RE: [Asterisk-Users] Please recommend a phone

[Christian Stredicke]

Take a look at snom.com...

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jesse Keating
> Sent: Wednesday, October 19, 2005 5:31 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Please recommend a phone
> 
> On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote:
> > 
> >I'm in need of a phone that would blink a led to let the callee 
> > know that there is an incoming call. The GXP-2000 does this 
> but I want 
> > an alternative to Grandstream. Any help is appreciated.
> 
> Polycom IP301s and 501s have a red LED that blinks when calls 
> are coming in.
> 
> --
> Jesse Keating
> GameHouse -- Systems Engineer
> 
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> 
> 
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[Asterisk-Users] Voicetronix FXS Open Loop Disconnects

[David Stude]

Hi-

Attempting to create a bridge using a Voicetronix OpenSwitch 12 between
Asterisk and Norstar MICS.  I have the Voicetronix ports configured as FXS,
and the Norstar seems to recognize open circuits on its FXO ports as a
trigger to disconnect (this doesn't happen when hanging up in asterisk, but
I was able to pull the wires to verify this happens).  This seems to be a
common way to do this, and I'm wondering if anyone has accomplished this
using Voicetronix products.

-David Stude

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[Asterisk-Users] New ISDN architecture available for asterisk

[Matteo Brancaleoni]

Hi to all,

sorry for crossposting the -dev and -user lists, but I think this could
be quite interesting news for EuroISDN people, expecially BRI owners.

A new ISDN architecture, called vISDN, has been developed to fully
support EuroISDN protocol with HFC based cards: HFC-S PCI, HFC-4S and
HFC-8S (with HFC-E1 and HFC-S USB support coming soon).

vISDN is not based on Zaptel, libpri, chan_zap, zaphfc, qozap, etc...
but has been designed from scratch to be a standard compliant EuroISDN
implementation plus a channel crossconnector, plus protocol analisys
support thru Ethereal, plus a ppp terminator, plus other stuff :)

Main features:

- Open, modular, flexible and versatile architecture
- Fully GPLed
- Full support for PRI and BRI
- Full support for Network and Terminal Equipment role
- Traffic analisys with (patched) Ethereal
- E-channel sniffing
- D-channel sharing between applications (in TE-multipoint mode)
- Good integration with latest 2.6 kernels and extensive usage of their
newer features (e.g.: sysfs)
- Protocol stacks are implemented following the finite-state-machine
models described in the ETSI specification for better
compliance/debuggability.
- Termination of PPP connections without exiting from the kernel
- Takes advantage from HFC's framer for HDLC traffic
- Preliminary/experimental support for hardware PCM bus
- Preliminary support for hardware channel bridging
- Support for dynamic (optionally automatic) activation/deactivation of
layer2 (DLC connection)
- Unintrusive with respect to the Linux kernel and Asterisk (no patches 
needed)

Missing features:

- Echo cancellation (likely going in in the next release)
- Explicit Call Transfer

Please note: This is a Linux-only, 2.6-only architecture which supports
only EuroISDN. It is currently in beta-stage, brave testers will be more
than welcome; people who want to contribute will be welcome too. You
will not need to sign a disclaimer to partecipate but you will receve
good and valuable consideration, plus discrete amounts of beer :^)

vISDN will undergo a layer2/layer3 certification by an independent lab
and will have a "Declaration of conformance" valid in the EU territory.
AFAIK, the declaration is valid for the product as a whole, so, it is to
be seen if the declaration could be extended to "mixed" products.

Please see http://www.visdn.org/ for further informations. If you are
near Milan, Italy, vISDN is live at SMAU expo. We will be happy to show
the driver working live and exchange comments and ideas about it
(Voismart - hall 12 booth H22)

Bye!


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Re: [Asterisk-Users] Free DID's

[Sergey Okhapkin]

On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote:


On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote:
> Callpacket.com has a free plan (up to 100 mins/month outbound,
> unlimited inbound, free DID).

Do you have hints on using callpacket w/ Asterisk?




register => sipusername:[EMAIL PROTECTED]

[callpacket-out]
type=peer
username=sipusername
secret=sipsecret
fromuser=sipusername
host=ser.callpacket.com
dtmfmode=rfc2833

[callpacket-in]
type=user
host=ser.callpacket.com
dtmfmode=rfc2833 
context=from-callpacket



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Re: [Asterisk-Users] IAX only speech one way

[Mir]

Bingo, that was the answer.

Originally, both ends had trunk=yes, and I had tried to comment it
out, but it didnt help.

But when I wrote trunk=no in end B, it worked.

Thanks for your help, everybody.

Michael

2005/10/19, David Uzzell <[EMAIL PROTECTED]>:
> Mir wrote:
> > Thanks for your suggestion.
> >
> > Unfortunately, it didnt change anything, A can still not hear B, but B
> > can hear A, strange..
> >
>
> I had the same problem with one of my IAX providers in AUS.
>
> Both ends turned of trunking and all was fine with the world again.
>
> Not sure what was the cause but that was my solution for EXACTLY the
> same problem that you explain.
>
> David
>
>
>
> > Michael
> >
> > 2005/10/19, Rich Adamson <[EMAIL PROTECTED]>:
> >
> >>>I have two Asterisk's connected via IAX, they are sitting on the same
> >>>network, via a VPN, so there should be no problems with firewalls.
> >>>
> >>>My problem is that when a person calls from A to B, A will not hear B
> >>>speak. B hears A fine.
> >>>
> >>>I doesn't matter who initiates the call.
> >>>
> >>>One of the Asterisk'ses is a new installation, just installed, but
> >>>with the Conf-files from an earlier setup, that worked fine.
> >>>
> >>>Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31
> >>>Asterisk version on computer B is Asterisk 
> >>>CVS-D2005.05.28.22.00.00-10/17/05
> >>>
> >>>Two different versions, but I dont think it should matter?
> >>
> >>Not sure this applies, but I was having the same problem with teliax.com
> >>and turning off the jitterbuffer in iax.conf fixed the problem. Kind
> >>of looks like we are running two different versions of asterisk as
> >>well, but I'd suspect that teliax has modified their system for
> >>other business purposes.
> >>
> >>Try jitterbuffer=no and see if it helps.
> >>
> >>Rich
> >>
> >>
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Re: [Asterisk-Users] Please recommend a phone

[Jesse Keating]

On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote:
> 
>I'm in need of a phone that would blink a led to let the callee
> know that there is an incoming call. The GXP-2000 does this but I want
> an alternative to Grandstream. Any help is appreciated.

Polycom IP301s and 501s have a red LED that blinks when calls are coming
in.

-- 
Jesse Keating
GameHouse -- Systems Engineer

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Re: [Asterisk-Users] Free DID's

[Jesse Keating]

On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote:
> Callpacket.com has a free plan (up to 100 mins/month outbound,
> unlimited inbound, free DID).

Do you have hints on using callpacket w/ Asterisk?

-- 
Jesse Keating
GameHouse -- Systems Engineer

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Re: [Asterisk-Users] INBOUND DID SERVICE FOR THE ASTERISK COMMUNITY

[Olle E. Johansson]

Federico Alves wrote:
> My company can supply inbound numbers in 49 States, via IAX, SIP or H323. I
> noticed that nobody offers this much needed service to the Asterisk

This kind of commercial notices should *not* be sent to this mailling
list. Please use the asterisk-biz mailing list for all commercial
services and products.

Thank you.

/Olle
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Re: [Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

[Jason Becker]

Steve Totaro wrote:

Anyone know how to get around this?  I am stumped.

# make mpg123
[ -f mpg123-0.59r.tar.gz ] || fetch
http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
make -C mpg123-0.59r linux
make[1]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \
audio_oss.o term.o' \
CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \
-DREAD_MMAP -DOSS -DTERM_CONTROL\
-Wall -O2 -m486 \
-fomit-frame-pointer -funroll-all-loops \
-finline-functions -ffast-math' \
mpg123-make
make[2]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make[3]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  -DREAD_MMAP 
-DOSS -DTERM_CONTROL-Wall -O2 -m486 -fomit-f

rame-pointer -funroll-all-loops -finline-functions -ffast-ma
th   -c -o mpg123.o mpg123.c
`-m486' is deprecated. Use `-march=i486' or `-mcpu=i486' instead.
cc1: error: CPU you selected does not support x86-64 instruction set
make[3]: *** [mpg123.o] Error 1
make[3]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[2]: *** [mpg123-make] Error 2
make[2]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make: *** [mpg123] Error 2


Use madplayer instead. There are several reasons why Digium & the 
Asterisk community should part ways with mpg123.


Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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[Asterisk-Users] some faxes get cut off

[Jay Austad]

Some incoming faxes are being cut off about halfway through.  The  
page looks fine, and then it gets some garbage and ends.


I've played around a whole bunch with my rxgain and txgain settings.   
The problem seems to depend on the type of fax machine sending.  Some  
machines send it flawlessly to me every time, and some always fail.


Could there be anything else that is causing this to happen besided  
the tx and rxgain?


~jay
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Re: [Asterisk-Users] Free DID's

[Sergey Okhapkin]

Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID).

On Wed, 2005-10-19 at 13:31 -0700, trixter aka Bret McDanel wrote:


On Wed, 2005-10-19 at 16:19 -0400, Yu Safin wrote:
> am I correct in believing that only goiax.com offers free DID's?

nope you are not correct.

ipkal.com offers (to any sip proxy now not just to FWD) washington state
DIDs free

stanaphone.com offers new york state DIDs free

calluk.com offers UK numbers free

sipgate.co.uk offers german and UK numbers free

voipbuster.com offers free outbound to 14 european countries (landline
only) and free to US mobiles and landlines.  Just gotta put $5 on your
account via paypal to get past the 1 minute call limit and get to 1 hour
call limits.

IF I MISSED ANY FREE PROVIDERS THAT SOMEONE ELSE KNOWS OF PLEASE REPLY
TO THE LIST SO THAT A MORECOMPLETE LIST CAN BE CREATED.


goiax.com offers free inbound with at least Mass DIDs and free US
outbound (although that is currently suspended Matthew indicated he
liked one method and that should be back up soon).


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[Asterisk-Users] Please recommend a phone

[Jesus Mogollon]

Hi All:   I'm in need of a phone that would blink a led to let the callee know that there is an incoming call. The GXP-2000 does this but I want an alternative to Grandstream. Any help is appreciated.
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Re: [Asterisk-Users] Asterisk 1.2.0beta1 for Debian Sarge

[Paul]

José Pablo Ezequiel Fernández wrote:

Is there anyone working on debs of Asterisk 1.2.0beta1 and related software 
for Debian Sarge ?
Because I am interested on them and I'll make them if necessary, but I don't 
want to re-invent the wheel (or anything).
 


http://rapid.dotsrc.org/experimental/


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Re: [Asterisk-Users] Free DID's

[trixter aka Bret McDanel]

On Wed, 2005-10-19 at 13:31 -0700, trixter aka Bret McDanel wrote:
> ipkal.com offers (to any sip proxy now not just to FWD) washington state
> DIDs free
> 
typo ipkall.com

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] INBOUND DID SERVICE FOR THE ASTERISK COMMUNITY

[Federico Alves]

My company can supply inbound numbers in 49 States, via IAX, SIP or H323. I
noticed that nobody offers this much needed service to the Asterisk
community. We charge $5/month per DID, plus 500 minutes of inbound traffic
at only 1 cent per minute. Our DID's are not single-line, but multi-line,
without any limit in terms of simultaneous inbound calls. I have clients who
push 48 or more calls on a single DID. Minutes are rounded to 60 seconds.
We support G729 only. There is a $30 setup fee and a $20 terminating fee. We
don't sell to end-users, but to companies looking to resell Voip services.
We take paypal or wire transfer.

Yours truly,

Federico Alves
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk 1.2.0beta1 for Debian Sarge

[Tzafrir Cohen]

On Wed, Oct 19, 2005 at 05:31:32PM -0300, José Pablo Ezequiel Fernández wrote:
> Is there anyone working on debs of Asterisk 1.2.0beta1 and related software 
> for Debian Sarge ?
> Because I am interested on them and I'll make them if necessary, but I don't 
> want to re-invent the wheel (or anything).

  deb http://rapid.dotsrc.org/ experimental
  deb-src http://rapid.dotsrc.org/ experimental

Not well tested. But bug reports are welcomed.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] Free DID's

[trixter aka Bret McDanel]

On Wed, 2005-10-19 at 16:19 -0400, Yu Safin wrote:
> am I correct in believing that only goiax.com offers free DID's?

nope you are not correct.

ipkal.com offers (to any sip proxy now not just to FWD) washington state
DIDs free

stanaphone.com offers new york state DIDs free

calluk.com offers UK numbers free

sipgate.co.uk offers german and UK numbers free

voipbuster.com offers free outbound to 14 european countries (landline
only) and free to US mobiles and landlines.  Just gotta put $5 on your
account via paypal to get past the 1 minute call limit and get to 1 hour
call limits.

IF I MISSED ANY FREE PROVIDERS THAT SOMEONE ELSE KNOWS OF PLEASE REPLY
TO THE LIST SO THAT A MORECOMPLETE LIST CAN BE CREATED.


goiax.com offers free inbound with at least Mass DIDs and free US
outbound (although that is currently suspended Matthew indicated he
liked one method and that should be back up soon).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Asterisk 1.2.0beta1 for Debian Sarge

[José Pablo Ezequiel Fernández]

Is there anyone working on debs of Asterisk 1.2.0beta1 and related software 
for Debian Sarge ?
Because I am interested on them and I'll make them if necessary, but I don't 
want to re-invent the wheel (or anything).
-- 
José Pablo Ezequiel Fernández
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Re: [Asterisk-Users] sixtel DID

[trixter aka Bret McDanel]

On Wed, 2005-10-19 at 15:51 -0400, Yu Safin wrote:
> has anybody tried to register with Sixtel to obtain a DID?
> I signed up 9/27 and I am yet to receive my DID.
> Also, how do I change my aix.conf to connect to Sixtel?
> I have a userid and password but I don't have details about all the 
> parameters.

I cant comment about the did but if you log into your account (you
should have gotten an email refering you to http://control.sixtel.net)
they have iax config stuff on their webpage.  It really is a simple
matter of cutting and pasting.

As for the did try contacting sixtel9 on aim, emailing them or calling
them.

I havent had any problems with their service so far.
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Free DID's

[Yu Safin]

am I correct in believing that only goiax.com offers free DID's?
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[Asterisk-Users] Problems Installing MPG123 on a 64 Bit System

[Steve Totaro]

Anyone know how to get around this?  I am stumped.

# make mpg123
[ -f mpg123-0.59r.tar.gz ] || fetch
http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
[ -d mpg123-0.59r ] || tar xfz mpg123-0.59r.tar.gz
make -C mpg123-0.59r linux
make[1]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make CC=gcc LDFLAGS= \
OBJECTS='decode_i386.o dct64_i386.o decode_i586.o \
audio_oss.o term.o' \
CFLAGS='-DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX \
-DREAD_MMAP -DOSS -DTERM_CONTROL\
-Wall -O2 -m486 \
-fomit-frame-pointer -funroll-all-loops \
-finline-functions -ffast-math' \
mpg123-make
make[2]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
make[3]: Entering directory `/usr/local/src/asterisk/mpg123-0.59r'
gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  -DREAD_MMAP 
-DOSS -DTERM_CONTROL-Wall -O2 -m486 -fomit-f
rame-pointer -funroll-all-loops -finline-functions -ffast-ma
th   -c -o mpg123.o mpg123.c
`-m486' is deprecated. Use `-march=i486' or `-mcpu=i486' instead.
cc1: error: CPU you selected does not support x86-64 instruction set
make[3]: *** [mpg123.o] Error 1
make[3]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[2]: *** [mpg123-make] Error 2
make[2]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make[1]: *** [linux] Error 2
make[1]: Leaving directory `/usr/local/src/asterisk/mpg123-0.59r'
make: *** [mpg123] Error 2

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[Asterisk-Users] Re: uable to establish link between asterisk to external phone

[kotesh m]

My mistake it is [EMAIL PROTECTED] 1.5. 
 
--k 
On 10/19/05, kotesh m <[EMAIL PROTECTED]> wrote:

 
Hi,
 
I am new Asterisk. I configured asterisk1.5 and be able to communicate from iaxComm dial pad to external computer i.e out side my router/LAN. When I make call from iamComm of external computer to my cell phone, I am getting the ring but not able to listen voice on both sides. Do I need to make any special configuration to make voice link.  

 
I found the same problem when used Sipura SIP device. 
 
Please let me know if I am missing anything.
 
Appreciate any help
 
--k
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RE: [Asterisk-Users] Help with Dial Plan

[steve]

On Wed, 19 Oct 2005, Dave Morrow wrote:

> Thanks Steve.  It almost works, but never dials the extension.  Also, is
> there a way I could mute the line while the remote attendant comes on? 


Oops sorry - the dangers of posting without testing.

The ","s are wrong - they should be w.  Each w is 1/2 second of waiting.

So that makes it:

exten => _6XXX,1,Dial(Zap/gX/1234567890,60,D(${EXTEN}))

As for the muting - bit of a loss about that one.

Steve

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[Asterisk-Users] Extension dialing out

[Anders Svensson]

Newbie warning

 

Hi!

 

Can I setup an extension that dial out directly to the
phone number I have with my sip provider. Like dial exten 110 and it connects
to my sip phone number

 

 

 

Regards

Anders Svensson



 

 






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Re: [Asterisk-Users] SIP to IAX

[Yu Safin]

On 10/19/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
> YES
>
> - Original Message -
> From: "Frank Kostin" <[EMAIL PROTECTED]>
> To: 
> Sent: Wednesday, October 19, 2005 8:58 AM
> Subject: [Asterisk-Users] SIP to IAX
>
>
> Hello everybody,
> Is it possible to route "any" incoming SIP call
> (without authentication - register) from an Asterisk A
> to a remote Asterisk B(throught IAX2), transparently ?
> Otherwise said, I would like to pass any incoming SIP
> call from Asterisk A to Asterisk B without SIP need to
> be registered, like a phone call in zap.
> I would apreciate any hint,
> Thanks,
> Frank
>
short answer yes,
read on,
what you really need to know is the compression.  You want to avoid
having to compress/uncompress different formats more than once.  I
normally have my SIP phones on 711 (same LAN to Asterisk A), then the
calls travel via IAX2 to Asterisk B (yes, it is transparent).  From
Asterisk B, they might go to Zap phones, SIP phones, IAX phones, FXO
(Zap), channel banks, etc.
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[Asterisk-Users] sixtel DID

[Yu Safin]

has anybody tried to register with Sixtel to obtain a DID?
I signed up 9/27 and I am yet to receive my DID.
Also, how do I change my aix.conf to connect to Sixtel?
I have a userid and password but I don't have details about all the parameters.
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Re: [Asterisk-Users] Re: Polycom IP501 and record on demand

[Matthew T. O'Connor]

Mojo with Horan & Company, LLC wrote:
That is perfect for one-button remaps! I guess I migrated away from 
one-button features in * but I see the light now.


Yeah, I have this working for one button remaps (remapping to the 
transfer button to #) but I never got the SpeedDial trick working.  
Could you resend me your exact I do it it remaps to volume up which is really annoying.


The trouble Matthew and I were having was to stimulate presses of more 
than one button in a sequence -- "SpeedDial" function was the only one 
I could find that was close, but this opens a new call appearance for 
the call rather than just playing the dtmf over the open one. 


This would be wonderful if anyone could get this working.  The really 
frustrating thing is that Polycom lists the "reprogramable buttons" as 
one of their selling points.  Anyway, I have given up for now.


Matt

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[Asterisk-Users] more voip patent madness

[trixter aka Bret McDanel]

Teles obtains US patent also for VoIP telephony method
http://www.heise.de/english/newsticker/news/65126

They are already involved in a lawsuit in germany over their patent
there, now that they have a US patent expect lawsuits in the US as
well.  

This just adds to the "about 100 patents on VoIP" that sprint-nextel
has, and sprint-nextels willingness to sue.  Course sprint-nextel cant
do boost mobile services anymore becuase prepaid mobile service is
patented.  What goes around comes around, and its all insane.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Polycom IP501 and record on demand

[Matthew T. O'Connor]

Jonathan k. Creasy wrote:

I probably can't provide any better information for you, however, have
you looked through the Polycom configuration files. The button mappings
are there. I haven't spent much time with it so I can not attest to what
you can map them to do. 


I have spent a lot of time looking through the config files, and also 
through the Admin guide which does show you how to remap a button (and I 
have it working for simple one button remaps), and it give you a list of 
functions that you can use while remapping buttons, but it gives you no 
infomratoin about those functions, what they do, or specifically how 
they work.  This is the really annoying part, is that they give you 
enough info to get really close, but not enougth to actually make it work.


Oh well.


Matt

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Re: [Asterisk-Users] possible bug, what do you think?

[Kevin P. Fleming]

Andy Goss wrote:


error.  The end result is that the user gets a new message envelope in
their INBOX (msg.txt) but there is no associated .WAV file to go
along with it.  The desired behavior here is to a) notify the user who
is attempting to forward this message that the process failed so that
the asterisk admin (me) can fix the issue or b) convert the file to the
proper format and then merge the two together.  What do you all think?


Yes, that is a bug, but there is no simple way to handle it. 
app_voicemail would have to work quite a bit differently to make this 
work seamlessly.


Basically, what you need to do when you change the formats in your 
voicemail.conf file is manually go through the spool area and ensure 
that every message is present in all the selected formats, just as if 
those formats had been selected initially. Otherwise you can get very 
bizarre behavior, depending on the codec in use when a user deals with 
the mailbox.

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Re: [Asterisk-Users] DID setup from goiax.com

[trixter aka Bret McDanel]

I dont know then that was cut and paste from what I have working ...

maybe actual log dumps of the error?

On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
> That is What I stated in the email.. my GOIAX #. not the DID #.
> 
> That is not the issue.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] uable to establish link between asterisk to external phone

[Olle E. Johansson]

kotesh m wrote:
>  
> Hi,
>  
> I am new Asterisk. I configured asterisk1.5 and be able to communicate
That is amazing. You are new and already at version 1.5. I have been
around for a while and only reached 1.1dev, working on 1.2 :-)

Guess I have some catching up to do...

/O ;-)
(Sorry, could not resist)
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RE: [Asterisk-Users] Help with Dial Plan

[Dave Morrow]

Thanks Steve.  It almost works, but never dials the extension.  Also, is
there a way I could mute the line while the remote attendant comes on? 


David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615 

< Poor planning on your part does not necessarily constitute an
emergency on my part! >

This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help with Dial Plan



On Wed, 19 Oct 2005, Dave Morrow wrote:

> Hi all. So far this list is proving it's worth, even on my first day 
> using it!  I hope that someone might know an easy solution to this
one.
> I would like to create a dial plan which will allow me to have all 
> extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a 
> local number, wait for an answer, wait 2 seconds and then enter the
extension.
> Can I do this in a dial plan somehow? This will allow me to 
> pseudo-integrate a legacy telephone switch (whose extensions are all
> 6XXX) to my Asterisk system for direct extension dialing.

exten => _6XXX,1,Dial(Zap/gX/1234567890,60,D(,,${EXTEN}))

where:
gX needs to become the group of the channels of your T1, 1234567890 is
the number of your legacy system.
60 is the dial timeout

You may need to adjust the number of commas to get the right delay.

Hope that helps,
Steve

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Re: [Asterisk-Users] sip rfc bye violated?

[Olle E. Johansson]

Matt Hess wrote:
> I should have mentioned that I can't do a full sip log.. with several
> calls a second whipping through this system it's almost impossible to
> weed out the info for the proper call.. and usually I don't see the dead
> channel until well after the fact.

http://bugs.digium.com/view.php?id=5475

/Olle
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[Asterisk-Users] possible bug, what do you think?

[Andy Goss]

We recently changed file formats on our server to wav49 from gsm.
Several users had saved messages in gsm format.  When a user attempts to
forward an old message to a user and they prepend the message with a
recording, the process seems to be flawed.  From what I can tell, the
prepend message is recorded to a temporary file, in my case
msg-prepend.WAV then after the prepend is finished recording,
asterisk attempts to merge the two audio files into one.  Since it
cannot find a msg.WAV file (the file is msg.gsm) it throws an
error.  The end result is that the user gets a new message envelope in
their INBOX (msg.txt) but there is no associated .WAV file to go
along with it.  The desired behavior here is to a) notify the user who
is attempting to forward this message that the process failed so that
the asterisk admin (me) can fix the issue or b) convert the file to the
proper format and then merge the two together.  What do you all think?

Thanks,
Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
 
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Re: [Asterisk-Users] sip rfc bye violated?

[Olle E. Johansson]

Matt Hess wrote:
> I should have mentioned that I can't do a full sip log.. with several
> calls a second whipping through this system it's almost impossible to
> weed out the info for the proper call.. and usually I don't see the dead
> channel until well after the fact.
> 
Looked at this with cooperation from Matt and it turned out to be a bug,
not in the way we handle the BYE, but in the way we handle a response to
a re-invite AFTER the bye...

Matt's got a patch to test. Hopefully we can fix this in cvs head quickly.

/O
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Re: [Asterisk-Users] initiate call recording from phone.

[Mojo with Horan & Company, LLC]

Well... I don't know anything about [EMAIL PROTECTED]  I know even more nothing about 
dialparties.agi... but I can summarize 
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial for you:


Let's say you want to call out on a PSTN line.  A command such as the 
following will be in your outgoing context:

exten => x,1,Dial(Zap/2/18005551212,,W)
before the first comma means dial 18005551212 out the second Zap line, 
the fact that there's nothing between the 2nd and 3rd comma means wait 
forever for an answer, and the W means let the _calling_ user (you) 
start a recording (in my case, with *#)


Let's say you want to be able to record incoming calls from PSTN.  A 
command such as the following would be in your incoming context:

exten => s,1,Dial(SIP/110,20,w)
The SIP/110 is where to ring when an incoming call comes in, the 20 
means wait 20 seconds before proceeding (to voicemail, or whatever you 
want) and the small w means let the _called_ user (you, again) start a 
recording however configured.


So... if you don't have direct control over your extensions.conf (as 
I said, I don't know [EMAIL PROTECTED]) I don't know if you can get your hands dirty 
with things like this.  Probably there's a check-box in [EMAIL PROTECTED] somewhere 
that allows this.


good luck!



todd wrote:

Moj
First great to see someone has figured this out, I have been struggling with 
it.
If not to much trouble; could you spare an example of where that "w or W" 
exist in the Dial command. Also will this command in the Dial plan work if I 
am using [EMAIL PROTECTED]
And how does this work into the whole picture with the dialparties.agi 
script, if at all?
Obviously I am a little confused on how this all works any help would be 
GREATLY appreciated.

Todd
- Original Message - 
From: "Mojo with Horan & Company, LLC" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, October 17, 2005 10:56 AM
Subject: Re: [Asterisk-Users] initiate call recording from phone.




And the w or W options must be specified in the Dial() cmd, as in:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial

Moj

Mojo with Horan & Company, LLC wrote:


If you have httpd with php on the * server, you can do what I did:

I set up the key combination *# in features.conf to monitor and created a 
few php files to interact with the results.  Save the four php files at:


http://horanappraisals.com/asterisk/

into a folder on the * web server, eg: /var/www/html/recordings/ -- 
rename them all to .php instead of .phps, and edit config.php to point to 
the asterisk monitor directory (usually /var/spool/asterisk/monitor). Now 
make a soft link so the recorded waves appear in the web tree:


ln -s /var/spool/asterisk/monitor /var/www/html/recordings/monitor

Then direct a web browser to http://asterisk_server/recordings/ and it 
should be pretty self-explanatory.  No recordings will appear in the list 
if you don't have the sox packages installed.


Andy Goss wrote:



I am looking for a way to allow a user to record a call simply by
pressing a button or some combination of buttons on their phone.  Is
this possible?

I have read the stuff about the monitor command; however, this doesn't
seem to be very interactive for the user.
Thanks,
Andy

--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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[Asterisk-Users] teliax audio issues - response

[Rich Adamson]

For those on the list using iax with teliax.com, the intermitant one-way
audio problem that I reported to them received the following response:

"We currently use our own version of 1.2 with our own patches on our boxes
and the iax code is updated. You cannot use jitterbuffers with g729 or gsm 
as this causes audio issues. So yes turning jitterbuffers will fix IAX 
issues."

So much for standards; rfc, defacto or otherwise.

Can anyone add anything more to the above?



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[Asterisk-Users] unable to make connectivity between asterisk to external phone

[kotesh m]

Hi All,
 
I am new to Asterisk. I configured asterisk and be able to communicate from iaxComm dial pad to external computer i.e out side my router/LAN. When I make call from iamComm of external computer to my cell phone, I am getting the ring but not able to listen voice . Do I need to make any special configuration to make voice link.  

 
I found the same problem when used SIPURA SIP device. Does anybody can provide the configuartion settings for Sipura SIP.. 
 
Please let me know if I am missing anything.
 
Appreciate any help
 
--kotesh
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

That did it... Thank you.

putting / after the register line caused it to not
register any more... and i would get error

server1.goiax.com/ could not be found.

anyways.. thanks for your help guys :)


> Replace
> [goiax]
> with
> [PHONENUMBER]
>
> username= don't work for users in IAX channel.
>
> On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
>> That is What I stated in the email.. my GOIAX #. not the DID #.
>>
>> That is not the issue.
>>
>> > for the incoming context put your goiax.com  phone
>> > number
>> > not the free DID number but the other one.
>> >
>> > On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> >>
>> >> Trixter:
>> >>
>> >> Thanks for the guide to setting this up:... I have tried the below
>> >> configuration with my settings, and when I place /goiax-in after my
>> >> register command, my register statement fails.
>> >>
>> >> If i remove it. I get a Rejected connect attempt from goiax's server
>> IP,
>> >> trying to reach 's@'
>> >>
>> >> I have put my GoIAX # in default, local, as the extension, and
>> nothing.
>> >>
>> >> I dont know where to look next on why i'm getting the rejected
>> connect
>> >> attempt.
>> >>
>> >> Thanks..
>> >>
>> >> ./Ben
>> >>
>> >> > On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> >> >> Can anybody post a step by step setup guide, please?
>> >> >
>> >> > Its like anything else once you have signed up ...
>> >> >
>> >> > in iax.conf
>> >> > register =>
>> >> PHONENUMBER:[EMAIL PROTECTED]/goiax-in> >> PROTECTED]/goiax-in>
>> >> >
>> >> > [goiax]
>> >> > type = peer
>> >> > host = server1.goiax.com 
>> >> > context = default
>> >> > secret = PASSWORD
>> >> > allow = gsm
>> >> > ;allow = ulaw
>> >> > ;disallow = all
>> >> > notransfer = yes
>> >> > qualify = yes
>> >> > auth = md5
>> >> > username = PHONENUMBER
>> >> >
>> >> >
>> >> > replace PHONENUMBER with the 8782 number you were issued.
>> Replace
>> >> > PASSWORD with your password from you account signup.
>> >> >
>> >> > Then in extensions.conf
>> >> > ; for outbound
>> >> > exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
>> >> > exten => _1NX,2,Busy
>> >> > exten => _1NX,102,Congestion
>> >> > exten => _1NX,202,playback(tt-weasels)
>> >> >
>> >> > ; for inbound
>> >> > exten => goiax-in,1,DO WHATEVER HERE
>> >> >
>> >> > asterisk -rx reload
>> >> >
>> >> > you should be set.
>> >> >
>> >> >
>> >> > --
>> >> > Trixter http://www.0xdecafbad.com Bret McDanel
>> >> > UK +44 870 340 4605 Germany +49 801 777 555 3402
>> >> > US +1 360 207 0479 or +1 516 687 5200
>> >> > FreeWorldDialup: 635378
>> >> > ___
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>> >> Easynews.com--
>> >> >
>> >> > Asterisk-Users mailing list
>> >> > Asterisk-Users@lists.digium.com
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> > To UNSUBSCRIBE or update options visit:
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >>
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>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>> >
>> >
>> > --
>> > Tom Vile
>> > Baldwin Technology Solutions, Inc
>> > Consulting - Web Design - VoIP Telephony
>> > www.baldwintechsolutions.com 
>> > Phone: 518-631-2855 x205
>> > Phone: 845-652-4578 x205
>> > Phone: 978-203-3848 x205
>> > Fax: 518-631-2856
>> > ___
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>> > Asterisk-Users@lists.digium.com
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Re: Polycom IP501 and record on demand

[Noah Miller]

Hi Mojo -


The trouble Matthew and I were having was to stimulate presses of more
than one button in a sequence -- "SpeedDial" function was the only  
one I
could find that was close, but this opens a new call appearance for  
the

call rather than just playing the dtmf over the open one.


Yeah, I'd like to be able to make that work, too.  Your method is  
pretty ingenious, and gets further that I got to mapping a key  
sequence to a single key.


I think we might just have to petition Polycom for this feature.   
Back in the pre-1.5.x firmware days, I did a feature request with  
them for the ability to disable their call waiting.  I don't know if  
my request really had an effect on them or not, but you can  
effectively do this in the 1.5.x firmware.  Maybe if everybody that  
want this submits a feature request to Polycom, they might just add  
it in a future firmware release.


There also might be some hope to hack together a solution on our  
own.  Anybody good with XML?



- Noah
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Re: [Asterisk-Users] DID setup from goiax.com

[Sergey Okhapkin]

Replace 
[goiax]
with
[PHONENUMBER]

username= don't work for users in IAX channel.

On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
> That is What I stated in the email.. my GOIAX #. not the DID #.
> 
> That is not the issue.
> 
> > for the incoming context put your goiax.com  phone
> > number
> > not the free DID number but the other one.
> >
> > On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >>
> >> Trixter:
> >>
> >> Thanks for the guide to setting this up:... I have tried the below
> >> configuration with my settings, and when I place /goiax-in after my
> >> register command, my register statement fails.
> >>
> >> If i remove it. I get a Rejected connect attempt from goiax's server IP,
> >> trying to reach 's@'
> >>
> >> I have put my GoIAX # in default, local, as the extension, and nothing.
> >>
> >> I dont know where to look next on why i'm getting the rejected connect
> >> attempt.
> >>
> >> Thanks..
> >>
> >> ./Ben
> >>
> >> > On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
> >> >> Can anybody post a step by step setup guide, please?
> >> >
> >> > Its like anything else once you have signed up ...
> >> >
> >> > in iax.conf
> >> > register =>
> >> PHONENUMBER:[EMAIL PROTECTED]/goiax-in >> PROTECTED]/goiax-in>
> >> >
> >> > [goiax]
> >> > type = peer
> >> > host = server1.goiax.com 
> >> > context = default
> >> > secret = PASSWORD
> >> > allow = gsm
> >> > ;allow = ulaw
> >> > ;disallow = all
> >> > notransfer = yes
> >> > qualify = yes
> >> > auth = md5
> >> > username = PHONENUMBER
> >> >
> >> >
> >> > replace PHONENUMBER with the 8782 number you were issued. Replace
> >> > PASSWORD with your password from you account signup.
> >> >
> >> > Then in extensions.conf
> >> > ; for outbound
> >> > exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
> >> > exten => _1NX,2,Busy
> >> > exten => _1NX,102,Congestion
> >> > exten => _1NX,202,playback(tt-weasels)
> >> >
> >> > ; for inbound
> >> > exten => goiax-in,1,DO WHATEVER HERE
> >> >
> >> > asterisk -rx reload
> >> >
> >> > you should be set.
> >> >
> >> >
> >> > --
> >> > Trixter http://www.0xdecafbad.com Bret McDanel
> >> > UK +44 870 340 4605 Germany +49 801 777 555 3402
> >> > US +1 360 207 0479 or +1 516 687 5200
> >> > FreeWorldDialup: 635378
> >> > ___
> >> > --Bandwidth and Colocation sponsored by
> >> Easynews.com--
> >> >
> >> > Asterisk-Users mailing list
> >> > Asterisk-Users@lists.digium.com
> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >> > To UNSUBSCRIBE or update options visit:
> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>
> >> ___
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> >> --
> >>
> >> Asterisk-Users mailing list
> >> Asterisk-Users@lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> >
> > --
> > Tom Vile
> > Baldwin Technology Solutions, Inc
> > Consulting - Web Design - VoIP Telephony
> > www.baldwintechsolutions.com 
> > Phone: 518-631-2855 x205
> > Phone: 845-652-4578 x205
> > Phone: 978-203-3848 x205
> > Fax: 518-631-2856
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

That is What I stated in the email.. my GOIAX #. not the DID #.

That is not the issue.

> for the incoming context put your goiax.com  phone
> number
> not the free DID number but the other one.
>
> On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>>
>> Trixter:
>>
>> Thanks for the guide to setting this up:... I have tried the below
>> configuration with my settings, and when I place /goiax-in after my
>> register command, my register statement fails.
>>
>> If i remove it. I get a Rejected connect attempt from goiax's server IP,
>> trying to reach 's@'
>>
>> I have put my GoIAX # in default, local, as the extension, and nothing.
>>
>> I dont know where to look next on why i'm getting the rejected connect
>> attempt.
>>
>> Thanks..
>>
>> ./Ben
>>
>> > On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> >> Can anybody post a step by step setup guide, please?
>> >
>> > Its like anything else once you have signed up ...
>> >
>> > in iax.conf
>> > register =>
>> PHONENUMBER:[EMAIL PROTECTED]/goiax-in> PROTECTED]/goiax-in>
>> >
>> > [goiax]
>> > type = peer
>> > host = server1.goiax.com 
>> > context = default
>> > secret = PASSWORD
>> > allow = gsm
>> > ;allow = ulaw
>> > ;disallow = all
>> > notransfer = yes
>> > qualify = yes
>> > auth = md5
>> > username = PHONENUMBER
>> >
>> >
>> > replace PHONENUMBER with the 8782 number you were issued. Replace
>> > PASSWORD with your password from you account signup.
>> >
>> > Then in extensions.conf
>> > ; for outbound
>> > exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
>> > exten => _1NX,2,Busy
>> > exten => _1NX,102,Congestion
>> > exten => _1NX,202,playback(tt-weasels)
>> >
>> > ; for inbound
>> > exten => goiax-in,1,DO WHATEVER HERE
>> >
>> > asterisk -rx reload
>> >
>> > you should be set.
>> >
>> >
>> > --
>> > Trixter http://www.0xdecafbad.com Bret McDanel
>> > UK +44 870 340 4605 Germany +49 801 777 555 3402
>> > US +1 360 207 0479 or +1 516 687 5200
>> > FreeWorldDialup: 635378
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>
>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com 
> Phone: 518-631-2855 x205
> Phone: 845-652-4578 x205
> Phone: 978-203-3848 x205
> Fax: 518-631-2856
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Re: [Asterisk-Users] DID setup from goiax.com

[Tom Vile]

for the incoming context put your goiax.com phone number not the free DID number but the other one.On 10/19/05, 
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Trixter:Thanks for the guide to setting this up:... I have tried the belowconfiguration with my settings, and when I place /goiax-in after myregister command, my register statement fails.If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'I have put my GoIAX # in default, local, as the extension, and nothing.I dont know where to look next on why i'm getting the rejected connectattempt.Thanks.../Ben
> On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:>> Can anybody post a step by step setup guide, please?>> Its like anything else once you have signed up ...>> in 
iax.conf> register => PHONENUMBER:[EMAIL PROTECTED]/goiax-in>> [goiax]> type= peer> host= 
server1.goiax.com> context = default> secret  = PASSWORD> allow   = gsm> ;allow  = ulaw> ;disallow   = all> notransfer  = yes
> qualify = yes> auth= md5> username= PHONENUMBER>>> replace PHONENUMBER with the 8782 number you were issued.  Replace> PASSWORD with your password from you account signup.
>> Then in extensions.conf> ; for outbound> exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)> exten => _1NX,2,Busy> exten => _1NX,102,Congestion
> exten => _1NX,202,playback(tt-weasels)>> ; for inbound> exten => goiax-in,1,DO WHATEVER HERE>> asterisk -rx reload>> you should be set.>>
> --> Trixter http://www.0xdecafbad.com Bret McDanel> UK +44 870 340 4605   Germany +49 801 777 555 3402> US +1 360 207 0479 or +1 516 687 5200> FreeWorldDialup: 635378
> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> Asterisk-Users mailing list> 
Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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Re: [Asterisk-Users] Asterisk management portal

[Jason Becker]

Tomislav Parčina wrote:

Does anybody have detailed instruction how to Install AMP? I have tried to 
install it using Installation Guide on their pages but I'm unable to satisfy 
AMP's PERL module dependencies.


Please post to the amportal-users list:

http://lists.sourceforge.net/lists/listinfo/amportal-users

and/or Help forum:

http://sourceforge.net/forum/forum.php?forum_id=414452

Please include in your post what PERL dependencies you are unable to 
satisfy and why. Please provide standard output in your post.


Regards,

--
Jason Becker
Director & CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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[Asterisk-Users] NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap

[Paul Hussein]

I am running [EMAIL PROTECTED] ( asterisk 1.2beta1 with two X100P cards ) on 
centos 4.1 box with a 2.6.12 kernel.


I ran  genzaptelconf 

and added two trunks for each of the devices however the incoming calls 
when I ring just get ignored.


asterisk -r tells me that it just gets hangupcall, and in the the log 
files I see exceptions.


I am running asterisk 1.2 beta.   Can someone help as to how to debug this

I am new to the asterisk game so any hints would be greatfully received.


== Manager 'admin' logged on from 127.0.0.1
   -- Starting simple switch on 'Zap/1-1'
   -- Executing Macro("Zap/1-1", "hangupcall") in new stack
   -- Executing ResetCDR("Zap/1-1", "w") in new stack
   -- Executing NoCDR("Zap/1-1", "") in new stack
   -- Executing Wait("Zap/1-1", "5") in new stack
   -- Executing Hangup("Zap/1-1", "") in new stack
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'Zap/1-1' in macro 'hangupcall'

 == Spawn extension (from-internal, s, 1) exited non-zero on 'Zap/1-1'
   -- Executing Macro("Zap/1-1", "hangupcall") in new stack
   -- Executing ResetCDR("Zap/1-1", "w") in new stack
   -- Executing NoCDR("Zap/1-1", "") in new stack
   -- Executing Wait("Zap/1-1", "5") in new stack
   -- Executing Hangup("Zap/1-1", "") in new stack
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'Zap/1-1' in macro 'hangupcall'

 == Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/1-1'





Oct 19 19:01:58 VERBOSE[10782] logger.c: -- Starting simple switch 
on 'Zap/1-1'
[EMAIL PROTECTED] ~]# Oct 19 19:02:03 NOTICE[10782] chan_zap.c: Got event 
18 (Event 18)...
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
Macro("Zap/1-1", "hangupcall") in new stack
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
ResetCDR("Zap/1-1", "w") in new stack
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: inserting a 
CDR record.
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: SQL command 
as follows:  INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) 
VALUES ('2005-10-19 19:02:03','\"device\" 
<400>','400','s','from-internal', 'Zap/1-1','','ResetCDR','w',0,0,'NO 
ANSWER',3,'')
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
NoCDR("Zap/1-1", "") in new stack

Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' not posted
Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' lacks end
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
Wait("Zap/1-1", "5") in new stack

Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on 
channel 1 (index 0)
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to 
ring. Old polarity was 0

Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Event 18(18) on 
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Dunno what to do with event 18 
on channel 1
Oct 19 19:02:08 DEBUG[10782] acl.c: # Testing 192.168.0.108 with 
192.168.0.0
Oct 19 19:02:08 NOTICE[10782] chan_sip.c: Registration from 'new 
' failed for '192.168.0.108' - Wrong password
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Hangup("Zap/1-1", "") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c:   == Spawn extension 
(macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall'
Oct 19 19:02:08 VERBOSE[10782] logger.c:   == Spawn extension 
(from-internal, s, 1) exited non-zero on 'Zap/1-1'
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Macro("Zap/1-1", "hangupcall") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
ResetCDR("Zap/1-1", "w") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
NoCDR("Zap/1-1", "") in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Wait("Zap/1-1", "5") in new stack

Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on 
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to 
ring. Old polarity was 0





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Re: [Asterisk-Users] Agent recording and muxmon

[Kevin P. Fleming]

Julian Lyndon-Smith wrote:

Torrow your time I presume - it's today in the uk:). Will this be in 
1.2, or is it a post 1.2 ?


It will be in 1.2.

I don't understand why they would be incompatible changes - could you 
not add a MuxMon facility as another option. e.g. in agents.conf:


RecordAgentCalls=no
MuxMonAgentCalls=yes


The existing monitor application supports behavior that is not 
implemented by the new one, like applications changing the monitor 
filename while the call is being monitored, started/stopping under 
application control, etc. The new application can eventually support 
that, but it does not do so currently.

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[Asterisk-Users] uable to establish link between asterisk to external phone

[kotesh m]

 
Hi,
 
I am new Asterisk. I configured asterisk1.5 and be able to communicate from iaxComm dial pad to external computer i.e out side my router/LAN. When I make call from iamComm of external computer to my cell phone, I am getting the ring but not able to listen voice on both sides. Do I need to make any special configuration to make voice link. 

 
I found the same problem when used Sipura SIP device. 
 
Please let me know if I am missing anything.
 
Appreciate any help
 
--k
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

Trixter:

Thanks for the guide to setting this up:... I have tried the below
configuration with my settings, and when I place /goiax-in after my
register command, my register statement fails.

If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'

I have put my GoIAX # in default, local, as the extension, and nothing.

I dont know where to look next on why i'm getting the rejected connect
attempt.

Thanks..

./Ben

> On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> Can anybody post a step by step setup guide, please?
>
> Its like anything else once you have signed up ...
>
> in iax.conf
> register => PHONENUMBER:[EMAIL PROTECTED]/goiax-in
>
> [goiax]
> type= peer
> host= server1.goiax.com
> context = default
> secret  = PASSWORD
> allow   = gsm
> ;allow  = ulaw
> ;disallow   = all
> notransfer  = yes
> qualify = yes
> auth= md5
> username= PHONENUMBER
>
>
> replace PHONENUMBER with the 8782 number you were issued.  Replace
> PASSWORD with your password from you account signup.
>
> Then in extensions.conf
> ; for outbound
> exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
> exten => _1NX,2,Busy
> exten => _1NX,102,Congestion
> exten => _1NX,202,playback(tt-weasels)
>
> ; for inbound
> exten => goiax-in,1,DO WHATEVER HERE
>
> asterisk -rx reload
>
> you should be set.
>
>
> --
> Trixter http://www.0xdecafbad.com Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
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Re: [Asterisk-Users] Caller-ID via database lookup

[Doug Lytle]

O'Connor, Jonathan wrote:


I have my Definity attached to my Asterisk box with a PRI Trunk.  The
guides and seemingly most people say to use a tie type connection,
however I did not get correct caller-id and setup until I:

1) Set the trunk-group on the Definity to isdn
2) Carrier/Medium to PRI
3) Trunk group numbering format to Public
 


Thanks for the Info Jonathan,

I'll give that info to our Definity manager.  I should have been clearer 
on our setup though.


The Definity that hooks up to our Asterisk box is a Definity G3R and I 
am getting CIDNumber and CIDName.  The other Definity hooks up to the 
first via a DCS link.  It's this linked Definity that I am having issues 
with on CIDNumber.


Again,

Thanks for your input!

Doug


--

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


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Re: [Asterisk-Users] Re: Polycom IP501 and record on demand

[Mojo with Horan & Company, LLC]

That is perfect for one-button remaps! I guess I migrated away from 
one-button features in * but I see the light now.


The trouble Matthew and I were having was to stimulate presses of more 
than one button in a sequence -- "SpeedDial" function was the only one I 
could find that was close, but this opens a new call appearance for the 
call rather than just playing the dtmf over the open one.


Moj

Noah Miller wrote:

Hi Matthew -



You could also take a look at features.conf, and use ** for blind
transfers, ## for attended transfers, *0 for recording, and *1 to  
hangup.


I haven't tried mapping them to polycom buttons, but there was
recently a discussion about that, just this week you can search the
archives.


There was a discussion (of which I was a part of) however there was no
resolution.  I have not found any good documentation on how to remap
Polycom buttons.  At this point I'm willing to pay for some help.
Anybody got some better info on this?



The best documentation I found is the Polycom manual.  It is fairly  
clear, though they don't provide a lot of examples.  Also, for some  
reason they put the button remapping documentation in one section of  
the manual and the button map in a completely different section.  A  
bit annoying.


I've remapped the transfer key to "#", so I can do an asterisk  
unattended transfer using the transfer key.  To do this, I just added  
the following line to my ipmid.cfg (or sip.cfg if you are using  
firmware version 1.5.x or later):


key.IP_500.37.function.prim="DialpadPound"  
key.IP_600.37.function.prim="DialpadPound"/>


Thanks,
Noah
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] Caller-ID via database lookup

[O'Connor, Jonathan]

I have my Definity attached to my Asterisk box with a PRI Trunk.  The
guides and seemingly most people say to use a tie type connection,
however I did not get correct caller-id and setup until I:

1) Set the trunk-group on the Definity to isdn
2) Carrier/Medium to PRI
3) Trunk group numbering format to Public

At this point I had to delete all the 23 ports from the trunk, busy it
out, change to the above, add the ports and release the trunk (a royal
pain).

My Asterisk box just uses:

loadzone= us
defaultzone = us
span=1,0,0,esf,b8zs
bchan=1-23 
dchan=24 



Once I had all of this done it was back to the Definity and into:

change isdn public-unknown-numbering

In mine, trunk group 4 is the Sprint PRIs used for normal calling.
Using 3742 as an example extension:

Ext Len 4
Ext Code37
Trk Grp 4   
CPN Prefix  614791
Ext Len 10

Therefore on trunk 4 it sends a caller id number of 6147913701

To make that send just 4 digits to Asterisk I added entries for:

Ext Len 4
Ext Code37
Trk Grp 1   
CPN Prefix  
Ext Len 4

And when 3742 calls an Asterisk box the Avaya send sonly its 4 digits as
caller ID on trunk 1.


I think the main change is the PRI type instead of tie, my system works
great since I did that, transferring back and forth no problem.  Oddly
tie only worked well with CSUs in place, PRI doesn't seem to care with
just a twist cable

Hope that helped, and didn't confuse you :)


-Jonathan


 
Jonathan O'Connor
System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
 
 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Doug Lytle
> Sent: Wednesday, October 19, 2005 12:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Caller-ID via database lookup
> 
> Hey everybody,
> 
> I'm having issues with one of our facilities, concerning caller-id.   
> The system is a Definity that hits a second Definity.  The 
> 2nd Definity trunks the call to my Asterisk server via a 
> TE110P.  I can only get Caller-ID name.  Nothing in the From: 
> field.  I thought I would be able to do a database lookup 
> against name to match extension, but
> 
> When doing this and setting Caller-ID number, it still shows 
> on the Polycom IP501 as Unknown/Unknown. Dial plan below:
> 
> exten => s,1,Set(dnd=${DB(DND/${ARG1})}) exten => 
> s,2,Set(CIDNUMB=${DB(cidname/${CALLERIDNAME})})
> exten => s,3,Set(CALLERID(Name)=${CALLERIDNAME})
> exten => s,4,Set(CALLERID(Number)=${CIDNUMB})
> 
> 
> CLI output below:
> 
> CLI> -- Accepting AUTHENTICATED call from 192.168.101.10:
>> requested format = gsm,
>> requested prefs = (),
>> actual format = gsm,
>> host prefs = (gsm),
>> priority = mine
> -- Executing Macro("IAX2/bc-asterisk-16384", 
> "sip.extensions|4483|") in new stack
> -- Executing Set("IAX2/bc-asterisk-16384", "dnd=") in new stack
> -- Executing Set("IAX2/bc-asterisk-16384", 
> "CIDNUMB=5574") in new stack
> -- Executing Set("IAX2/bc-asterisk-16384", "CALLERID(Name)=Lytle,
> Doug") in new stack
> -- Executing Set("IAX2/bc-asterisk-16384", 
> "CALLERID(Number)=5574") in new stack
> -- Executing GotoIf("IAX2/bc-asterisk-16384", "0?8:6") in 
> new stack
> -- Goto (macro-sip.extensions,s,6)
> -- Executing SetMusicOnHold("IAX2/bc-asterisk-16384", 
> "epi-cd") in new stack
> -- Executing Dial("IAX2/bc-asterisk-16384", 
> "SIP/4483|28|t") in new stack
> -- Called 4483
> -- SIP/4483-04ba is ringing
> 
> Debug output from the 'receiving Asterisk' server via IAX below:
> 
> 
> -- SIP/4483-14d3 is ringing
> Reliably Transmitting (no NAT) to 192.168.101.64:5060:
> CANCEL sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.104.40:5060;branch=z9hG4bK73d230ce;rport
> From: "Unknown" ;tag=as23c39fbe
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
> What variable needs to be set to change it from "Unknown" to 5574?
> 
> Any help would be appreciated.
> 
> Doug
> 
> -- 
>  
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a 
> little Temporary Safety, deserve neither Liberty nor Safety."
> 
> 
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[Asterisk-Users] Re: Polycom IP501 and record on demand

[Noah Miller]

Hi Matthew -


You could also take a look at features.conf, and use ** for blind
transfers, ## for attended transfers, *0 for recording, and *1 to  
hangup.


I haven't tried mapping them to polycom buttons, but there was
recently a discussion about that, just this week you can search the
archives.


There was a discussion (of which I was a part of) however there was no
resolution.  I have not found any good documentation on how to remap
Polycom buttons.  At this point I'm willing to pay for some help.
Anybody got some better info on this?


The best documentation I found is the Polycom manual.  It is fairly  
clear, though they don't provide a lot of examples.  Also, for some  
reason they put the button remapping documentation in one section of  
the manual and the button map in a completely different section.  A  
bit annoying.


I've remapped the transfer key to "#", so I can do an asterisk  
unattended transfer using the transfer key.  To do this, I just added  
the following line to my ipmid.cfg (or sip.cfg if you are using  
firmware version 1.5.x or later):


key.IP_500.37.function.prim="DialpadPound"  
key.IP_600.37.function.prim="DialpadPound"/>


Thanks,
Noah
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Re: [Asterisk-Users] Voicemail as an email attachement

[Tzafrir Cohen]

On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote:
> I have configured the voicemail.conf file as per the wiki to email 
> voicemails as an attachment. I cannot find any instructions/locations to 
> set the outgoing server login information. Furthermore, I can get no 
> emails from asterisk. Can anyone point me to the next step to setup the 
> attachment of voicemail messages to an email?

Set up a "sendmail". Or basically: an MTA. Any linux distro comes with
at least one (postfix seems to be the preffered choice nowadays). Which
one do you use?

There are a bunch of programs that provide /usr/sbin/sendmail but don't
spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are
probably others.

The downside is that messages that have, for some reason, not been
delivered in the first shot (e.g: due to some transient network error)
will be dropped rather than queued.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Caller-ID via database lookup

[Doug Lytle]

Hey everybody,

I'm having issues with one of our facilities, concerning caller-id.   
The system is a Definity that hits a second Definity.  The 2nd Definity 
trunks the call to my Asterisk server via a TE110P.  I can only get 
Caller-ID name.  Nothing in the From: field.  I thought I would be able 
to do a database lookup against name to match extension, but


When doing this and setting Caller-ID number, it still shows on the 
Polycom IP501 as Unknown/Unknown. Dial plan below:


exten => s,1,Set(dnd=${DB(DND/${ARG1})})
exten => s,2,Set(CIDNUMB=${DB(cidname/${CALLERIDNAME})})
exten => s,3,Set(CALLERID(Name)=${CALLERIDNAME})
exten => s,4,Set(CALLERID(Number)=${CIDNUMB})


CLI output below:

CLI> -- Accepting AUTHENTICATED call from 192.168.101.10:
  > requested format = gsm,
  > requested prefs = (),
  > actual format = gsm,
  > host prefs = (gsm),
  > priority = mine
   -- Executing Macro("IAX2/bc-asterisk-16384", "sip.extensions|4483|") 
in new stack

   -- Executing Set("IAX2/bc-asterisk-16384", "dnd=") in new stack
   -- Executing Set("IAX2/bc-asterisk-16384", "CIDNUMB=5574") in new stack
   -- Executing Set("IAX2/bc-asterisk-16384", "CALLERID(Name)=Lytle, 
Doug") in new stack
   -- Executing Set("IAX2/bc-asterisk-16384", "CALLERID(Number)=5574") 
in new stack

   -- Executing GotoIf("IAX2/bc-asterisk-16384", "0?8:6") in new stack
   -- Goto (macro-sip.extensions,s,6)
   -- Executing SetMusicOnHold("IAX2/bc-asterisk-16384", "epi-cd") in 
new stack
   -- Executing Dial("IAX2/bc-asterisk-16384", "SIP/4483|28|t") in new 
stack

   -- Called 4483
   -- SIP/4483-04ba is ringing

Debug output from the 'receiving Asterisk' server via IAX below:


   -- SIP/4483-14d3 is ringing
Reliably Transmitting (no NAT) to 192.168.101.64:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.104.40:5060;branch=z9hG4bK73d230ce;rport
From: "Unknown" ;tag=as23c39fbe
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

What variable needs to be set to change it from "Unknown" to 5574?

Any help would be appreciated.

Doug

--

Ben Franklin quote:

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


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Fwd: Re: [Asterisk-Users] IAX only speech one way

[Jerry Richmond]

IAX may be as bad as what we are doing?Note: forwarded message attached.--- Begin Message ---
Mir wrote:
> Thanks for your suggestion.
> 
> Unfortunately, it didnt change anything, A can still not hear B, but B
> can hear A, strange..
> 

I had the same problem with one of my IAX providers in AUS.

Both ends turned of trunking and all was fine with the world again.

Not sure what was the cause but that was my solution for EXACTLY the
same problem that you explain.

David



> Michael
> 
> 2005/10/19, Rich Adamson <[EMAIL PROTECTED]>:
> 
>>>I have two Asterisk's connected via IAX, they are sitting on the same
>>>network, via a VPN, so there should be no problems with firewalls.
>>>
>>>My problem is that when a person calls from A to B, A will not hear B
>>>speak. B hears A fine.
>>>
>>>I doesn't matter who initiates the call.
>>>
>>>One of the Asterisk'ses is a new installation, just installed, but
>>>with the Conf-files from an earlier setup, that worked fine.
>>>
>>>Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31
>>>Asterisk version on computer B is Asterisk CVS-D2005.05.28.22.00.00-10/17/05
>>>
>>>Two different versions, but I dont think it should matter?
>>
>>Not sure this applies, but I was having the same problem with teliax.com
>>and turning off the jitterbuffer in iax.conf fixed the problem. Kind
>>of looks like we are running two different versions of asterisk as
>>well, but I'd suspect that teliax has modified their system for
>>other business purposes.
>>
>>Try jitterbuffer=no and see if it helps.
>>
>>Rich
>>
>>
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[Asterisk-Users] E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a UA, but i'm in state 1" (long)

[Dinesh Nair]

 Original Message 
Subject: E1 PRI error: "!! Got I-frame while link state 2" and "!! Got a 
UA, but i'm in state 1"

Date: Wed, 19 Oct 2005 23:46:01 +0800
From: Dinesh Nair <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 
, Asterisk on BSD discussion 



hey * folk,

i've got a TE410P (generation 1 firmware) stuck in a box with a single xeon
2.8Ghz and 1GB RAM. there's a loopback E1 cable connecting span 1 to span 4
(zaptel.conf and zapata.conf below). upon starting up asterisk, i see the
following errors consistently on the screen,

!! Got I-frame while link state 2
!! Got a UA, but i'm in state 1

they seem to be coming from libpri.so.1 and the spans seem to be restarting
each other infinitely. i also get a number of the following messages from
chan_zap.so:

B-channel 0/6 restarted on span 1
B-channel 0/6 restarted on span 4
B-channel 0/7 restarted on span 1
B-channel 0/7 restarted on span 4
B-channel 0/8 restarted on span 1
B-channel 0/8 restarted on span 4
B-channel 0/9 restarted on span 1
B-channel 0/9 restarted on span 4

No D-channels available! Using Primary Channel 16 as D-channel anyway!
No D-channels available! Using Primary Channel 109 as D-channel anyway!

both spans show "Provisioned,Up, Active" in pri show span, and zttest shows
100% all the way.

a snapshot of pri debug span 1, shows:

> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 83]
> 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: 3 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 0/0x0) (Terminator)
< Message type: RESTART ACKNOWLEDGE (78)
< [18 03 a9 83 83]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 84]
> 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: 4 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

!! Got I-frame while link state 2
> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 83]
> 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: 3 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

< Protocol Discriminator: Q.931 (8)  len=13
< Call Ref: len= 2 (reference 0/0x0) (Originator)
< Message type: RESTART (70)
< [18 03 a9 83 84]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

 Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Terminator)
> Message type: RESTART ACKNOWLEDGE (78)
> [18 03 a9 83 84]
> 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: 4 ]
> [79 01 80]
> Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

!! Got I-frame while link state 2
> Protocol Discriminator: Q.931 (8)  len=13
> Call Ref: len= 2 (reference 0/0x0) (Originator)
> Message type: RESTART (70)
> [18 03 a9 83 85]
> 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
> 

[Asterisk-Users] Trunk Dialing rules

[bails]

Hi i have posted before about this problem, and have had several 
suggestions, that i can use contexts to overcome this.


The situation. [EMAIL PROTECTED] 1.5

I have 3 sets of users say sales, admin and tech with the numbers

sales200 201
admin 202 203
tech   204 205

They all need to be able to ring each other hence they are all in 
[ext-local]


Each group has its own trunk, which is unavailable to other users, at 
this point I am failing.  If i make say sales members of [ext-local-1], 
admin members of [ext-local-2] and tech members of [ext-local-3], they 
cannot call each other until i add then to


[ext-local]
include => ext-local-1
include => ext-local-2
include => ext-local-3

then of course thay can call each other, the trouble is they can then 
call [all-routes-outbound] which is not what i want.


if i remove [all-routes-outbound]
noone can call out over the trunks.

so i create
[from-internal-local-1]
include => ext-local-1
include => outbound-outrt-1  ;iax route out 1

and
[from-internal]
include => from-internal-local-1

However this means that anyone can dial out over outbound-outrt-1

Which is what i was trying to avoid in the outset.

Is this possible with [EMAIL PROTECTED]

if so how? (my head is bruised from repeatedly banging it against the wall)


I would love to have this funtionality available from the amportal, 
something like add extensions totrunks.



Thanks

Bails
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RE: [Asterisk-Users] Polycom IP501 and record on demand

[Jonathan k. Creasy]

I probably can't provide any better information for you, however, have
you looked through the Polycom configuration files. The button mappings
are there. I haven't spent much time with it so I can not attest to what
you can map them to do. 

Hope this helps you a little. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew T.
O'Connor
Sent: Wednesday, October 19, 2005 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP501 and record on demand

Matt Gibson wrote:
> You could also take a look at features.conf, and use ** for blind 
> transfers, ## for attended transfers, *0 for recording, and *1 to
hangup.
>
> I haven't tried mapping them to polycom buttons, but there was 
> recently a discussion about that, just this week you can search the 
> archives. 

There was a discussion (of which I was a part of) however there was no 
resolution.  I have not found any good documentation on how to remap 
Polycom buttons.  At this point I'm willing to pay for some help.  
Anybody got some better info on this?

Thanks,

Matthew O'Connor


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RE: [Asterisk-Users] DNIS/DNID

[James Steven]

That worked great. Thanks


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Steve 
TotaroSent: 19 October 2005 14:45To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
DNIS/DNID

exten => xx,2,SetCIDName(*-SALES-* 
${CALLERIDNAME})

  - Original Message - 
  From: 
  James Steven 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, October 19, 2005 9:33 
  AM
  Subject: [Asterisk-Users] DNIS/DNID
  
  Hi
  Is it possible 
  with Asterisk to tell the called party which number was dialled by the 
  caller?  Or in place of the number dialled have a description such as 
  'Sales' or 'Accounts'?  Ideally, I would like to show a description 
  corresponding to the number dialled followed by CIDName.  
  How might this be set up?  
   
  Currently my 
  extensions.conf is:
   
  exten => 
  xx,1,LookupCIDNameexten => xx,2,Dial(SIP/xx,50)exten => 
  xx,3,Voicemail(xx)exten => xx,4,Hangup
   
  Thanks for your 
  help.
   
  
  

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Re: [Asterisk-Users] Polycom IP501 and record on demand

[Matthew T. O'Connor]

Matt Gibson wrote:
You could also take a look at features.conf, and use ** for blind 
transfers, ## for attended transfers, *0 for recording, and *1 to hangup.


I haven't tried mapping them to polycom buttons, but there was 
recently a discussion about that, just this week you can search the 
archives. 


There was a discussion (of which I was a part of) however there was no 
resolution.  I have not found any good documentation on how to remap 
Polycom buttons.  At this point I'm willing to pay for some help.  
Anybody got some better info on this?


Thanks,

Matthew O'Connor


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Re: [Asterisk-Users] Recomendations for utility togenerateAsteriskconfiguration

[Tom Rymes]

On Oct 19, 2005, at 11:04 AM, asterisk wrote: AMP's dialplan and setup is quite complex.   Requires, e.g, a number of  AGIs.This is normally not the type of thing you'd like   to hand-edit later  after the initial adaptation to the target   system.Who said anything about hand editing?That is why you would want to keep the old computer running [EMAIL PROTECTED].  Insteadof hand editing anything, make the changes on the [EMAIL PROTECTED] box's AMP GUI andcopy them over again.  Very simple and most tech folks have an old computerlaying around somewhere that could be put to use.  Why wouldn't you just install [EMAIL PROTECTED] on your main server then? Why install a   second server and go through the trouble of using scp to copy files back and   forth?Tom     Maybe because you snipped the beginning of the   thread without reading the entire thread's context, but he is running on   Solaris.  I am not sure what all is involved with installing [EMAIL PROTECTED] on solaris but I assume it is no trivial   task.  WinSCP is very trivial IMHO and there is no "copying files back   and forth", just one direction, takes about twenty seconds and maybe 30 if you   are slow.  Now you also have an almost hot swap server in case the   Solaris machine goes down, just swap IP addresses and hardware.OK, my bad, but the point is still valid. If you are dead set on running Solaris, install AMP on the Solaris server, don't go to the trouble of creating a second machine to generate your configuration when you can just eliminate the extra steps and create the config on the main machine. It would be simpler to set up, maintain, and make testing config changes much easier and faster.Tom___
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Re: [Asterisk-Users] Voicemail as an email attachement

[[EMAIL PROTECTED]]

Yes. I am interested. I will make provisions for the upload. How big are 
the files?


Thanks

BEN

Goran Skular wrote:

I changed my app_voicemail.c to work not with sendmail but with sendEmail
that connects to any SMTP and sends email with attachment...

It's dirty, but it works.

If you are interested I can upload app_voicemail.c and sendEmail package
somewhere..




I have configured the voicemail.conf file as per the wiki to email
voicemails as an attachment. I cannot find any instructions/locations to
set the outgoing server login information. Furthermore, I can get no
emails from asterisk. Can anyone point me to the next step to setup the
attachment of voicemail messages to an email?

Thanks

BEN
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[Asterisk-Users] Connection question

[Joao Carneiro - DLS]

Asterisk seems to be a very good peace of software, but i am interested
to know if i can use plain ISDN cards with it, i mean use the isdn cards
as a passthrough device between my alcatel pbx and voip users.

thanks 


DLS - Projectos, Automação e Manutenção, Lda. 
João Carneiro, Tecnico 
Dep. Sistemas de Informação 
Rua da Boavista S/N - P.O.Box 313
4416-901 Grijó 
www.dls.pt

Email: [EMAIL PROTECTED]

Tel : +351 227 470 786

Fax : +351 227 470 787

Tlm : 




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Re: SV: [Asterisk-Users] Queues and call waiting indication

[Tom Rymes]

On Oct 18, 2005, at 9:08 AM, Adam Goryachev wrote:


On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote:


 Hi,

This issue has been discussed probably a million times on every  
asterisk forum in the world and I have the same problem too.  
Another problem you would have with the agents is that when they  
make an outgoing call they are not regarded as "busy" by asterisk  
and it sends more calls to the agent if it has call waiting enabled.


This behaviour is totally senseless since the whole purouse of  
queues is to _queue_ the callers until the agent is available.  
"available" usually means "not on the phone" -- whether or not  
it's an incoming or outgoing call.


I "solved" this problem by using single-line clients and phones  
where you can turn off call wating.




Actually this can simply be solved in your dialplan Just use the
setgroup/checkgroup values, and use the AgentCallbackLogin instead of
AgentLogin 

This is what I used, and it seems to work quite well so far... well, I
haven't actually added the bits for the outbound calls yet on my own
system, but I've done it on others, and they seem to be quite happy  
with

it...


Can you provide some more specifics? Maybe an example for the  
dialplan? Does this keep the queue from sending multiple calls to  
agents who have call waiting enabled?


Tom
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Re: [Asterisk-Users] more dids added to goiax.com

[John Novack]

Steve Totaro wrote:

  
  
  
I made two attempts this morning to send some comments off list, and both got returned due to some sort of spam filter, so would hope that any future controls will not suffer from that inability to communicate.

  
  
Maybe if you posted what the returned email said then he could remove or alter that filter.  "both got returned due to some sort of spam filter", doesn't help anyone.

  

I saw no reason to clog up this list with  details that aren't Asterisk
related.
If he wants further information he can communicate directly with me.

JN


  
  



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Re: [Asterisk-Users] Caller ID

[Nathan Pralle]

The question I have is, if an incoming call from the POTS line has
caller ID information, does/is/can that information be passed onto
the analog phone so it's caller id display will show the info?  If
so, is there anything I need to do to make this happen or does it
*just work*?  Thanks.


It should "just work".  You can modify it if you want, but if you don't, 
it should just pass it on through.


Nathan


--
-
Nathan E. Pralle
Give the Director a Serpent Deflector
www.nathanpralle.com
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Re: [Asterisk-Users] Problem with select correct network interface (oh323)

[Moises Silva]

it seems to me that your problem is not Asterisk configuration, but
iproute configuration. Look in google about iproute and kernel routing
tables.
In order to help you, it would be desireable to know how are you dialing.

Best Regards.On 10/19/05, Oleh Mukha <[EMAIL PROTECTED]> wrote:
i build asterisk on pc with 3 network inerfaceeth0 (yyy.yyy.yyy.yyy) main public ipeth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connectioneth2 (zzz.zzz.zzz.zzz) local ipi config oh323 to bind eth1 interface
i try make callfrom my local network -> Asterisk -> provider h323when i try to call from ata 186 throught my astersik oh323 moduleasetrisk resive calls from ata but send it to my oh323 providet not from eth1
(with ip xxx.xxx.xxx.xxx) or from eth0 (ip yyy.yyy.yyy.yyy)how can i tel asterisk send data from me to my provider from eth1 (ipxxx.xxx.xxx.xxx)Oleh MukhaIClub380322722738
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