Re: [asterisk-users] Necessary hardware

[Kosa]

I found a sap400, wich is FXO and seems to work fine with asterisk. It
offers the posibility to plug 4 analog phonelines.

Thanks again.

Kosa

- Un mundo mejor es posible -

Ioan Indreias escribió:
 Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
 other ATA which have FXO ports).
 
 HTH,
 Ioan.
 
 On Thu, Apr 1, 2010 at 12:29 AM, Kosa k...@piradio.org wrote:
 
 I have two linksys spa2102 and a sap9000 but as far as I know I need
 something else to connect the asterisk box to the analog phoneline.
 


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[asterisk-users] Necessary hardware

[Kosa]

Hi there!

just a quick question: what would you recommend to get to connect an
asterisk box to the analog phoneline?

I have two linksys spa2102 and a sap9000 but as far as I know I need
something else to connect the asterisk box to the analog phoneline. I
just have two analog phone lines, so getting one device to plug both is
fine, but two devices to connect one by one would be good too. The
cheaper the better this time.

Thanks in advance.

--

Kosa

- Un mundo mejor es posible -

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[asterisk-users] Questions about asterisk and spa2102

[Kosa]

Hi there! First mail on the list :)

1.- is it possible to use an spa2102 to make and revice calls from a
normal phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.

2.-  I'm trying to unlock the spa2102 with no succes at the moment, any
links or hint will be very appreciated.


I'm and absolute newbie on asterisk, btw.


Thanx!

Kosa

- Un mundo mejor es posible -


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Re: [asterisk-users] Problem with SPA3000 -- dropping calls

[Adam KOSA]

  Most of the times it works, other times phone on FXS rings, I pick it
  up and all I get is a dial tone.

this might also be MWI in SPA-style.

regards
adam

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[asterisk-users] gtalk and dtmf

[Adam KOSA]

Hi,

i've just finished setting up gtalk connection with asterisk.  it works 
nice, audio is full duplex.

i just have one question which i could not find an exact answer to.  Is 
gtalk able to send dtmf codes?  Because i'd like to listen to my 
voicemails while away from home.

I've been googling for half an hour, i found some sort of jingle 
protocol which i'm not sure what to use for but it might be the 
solution?  It seems to me that my problem is sending the dtmf tones, not 
receiving them, so this is really gtalk related.

I'm writing here because i read many of you have successfully integrated 
gtalk to asterisk and hoping somebody have a solution or at least some 
direction where i can move forward to.

thanks
Adam

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Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

[Adam KOSA]

 permit udp any host 192.168.5.0 range 1 2 and then I didn't

home users typically use /24 netmask.  If this is the case, i don't 
understand why do you write keyword host following a network address.

either specify a valid host address, or write 192.168.5.0 255.255.255.0 
to specify the whole subnet.

if the netmask isn't /24 then, of course the above 5.0 may be a valid 
host address.

regards
adam

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Re: [asterisk-users] FXO ATA Options?

[Adam KOSA]

Hi,

 I'm currently looking at FXO options to provide a POTS line to Asterisk to 
 trunk calls with.



i've had some problems setting the disconnect tone correctly to my 
country.  As a matter of fact, i still do, as the calculated values does 
not always hang up the phone.

Other than this i have a small issue which i did not understand 
completely.  Sometimes the SPA webpage starts to load, and in the middle 
i get a connection reset page by firefox.  Sometimes i can only load the 
page by refreshing 10-15 times or even more.  This only happens in 
advanced admin mode, when using any other modes everything works fine. 
This refresh-error only occures from remote networks, not from a PC that 
is within the same subnet as the SPA (subnets are connected via 
site2site vpn tunnel).

I haven't had time to correctly debug this issue (tcpdump etc)  but it's 
so annoying that i will go and debug this once.  It may be an MTU issue, 
an SPA performance issue, a firefox issue...

This is an SPA3k which i'm using (actually not one but four, all 
involved in this problem).

regards
Adam

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Re: [asterisk-users] Backuping VoIP provider with PRI

[Adam KOSA]

Marc Patino Gómez wrote:
 in most cases it works well but, if my internet connection is down 
 Asterisk tries to Dial voipprovider, but it can't resolve the dns name, 
 so it waits 60 seconds to jump to the following priority...
 
 Any ideas to solve this problem? I can't use the IP of the provider (it 
 has a pool of servers), I try to use dnsmgr without solving the isue
 

Why don't you fill the ip addresses to your /etc/hosts file?  In that 
way lookups won't need any dns resolving and still could keep the load 
balancing by having multiple ip addresses to the same SIP hostname.

regards
Adam

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Re: [asterisk-users] Voicemail.conf

[Adam KOSA]

Paul Hales wrote:
   
 Is there a way to specify multiple email addresses in voicemail.conf for
 a specific user?
 

why don't you use the /etc/aliases file for this purpose?

regards
adam

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[asterisk-users] asterisk and vad/cng

[Adam KOSA]

Hi List,

i've set up a cisco 7912 for my asterisk box.  I've had problems with 
VAD and CNG.  After googling a bit, i've found an article about asterisk 
not supporting these two protocols, therefore it's better to turn them off.

Since then i did not found answer to my two questions, maybe somebody 
here could help me:

a) am i even able to turn off vad/cng on cisco 7912?  SIP image 8.0 
version.  I've been through the cisco admin guide, but it did not help.
b) the article mentioned above was dated 2006, does asterisk still not 
support VAD/CNG?  I'm using 1.4.8, the log says it does not, but maybe a 
patch, or 1.4.11?

I encounter no big problems, only with MOH: i have to make some noise 
(breathing loud etc) to hear the music.  Not a big deal after all.

And some other questions not really related:  anybody has experience 
with the following phones: linksys 921/941 and cisco 3911?  Do these 
have the same problem?  According to their datasheet, linksys supports 
VAD, but the user guide says nothing about turning it off.

Thank you for any replies, it would help me a lot!

Regards
Adam

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[asterisk-users] callback and bridge problem

[Adam KOSA]

Greetings,

i've been posted a message to this list in july, which had one response. 
  Thanks for that idea!  Unfortunately asterisk is only a hobby, and did 
not have much time dealing with the problem since.  My original letter 
was long, i wouldn't post it again, the archive url is

http://archives.free.net.ph/message/20070710.053008.c02209c0.en.html

Since than i've upgraded to 1.4.8 from 1.2 series, i thought this might 
help.  It did not.

Answering to the question asked from me in july, no, i'm not behind nat, 
  and i did not have reinvite=yes in my configs.  I've put it into the 
sip.conf, tried, but the call hung up again.

I'd be greatful for more ideas of solving the problem.

Fresh logs when hanging up, from asterisk console:

 -- SIP/neophonex99-out-08213ac8 is making progress passing it to 
SIP/neophonex57-out-081e8a78
[Aug  2 21:54:51] WARNING[24739]: chan_sip.c:11948 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '[EMAIL PROTECTED]'. Giving up.
 -- SIP/neophonex99-out-08213ac8 answered SIP/neophonex57-out-081e8a78
 -- Native bridging SIP/neophonex57-out-081e8a78 and 
SIP/neophonex99-out-08213ac8
[Aug  2 21:54:57] WARNING[24739]: chan_sip.c:11948 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '[EMAIL PROTECTED]'. Giving up.
   == Spawn extension (internal, 9520620*, 3) exited non-zero on 
'SIP/neophonex57-out-081e8a78'
[Aug  2 21:54:57] NOTICE[24749]: pbx_spool.c:351 attempt_thread: Call 
completed to SIP/[EMAIL PROTECTED]


Thanks for any help
Adam

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[asterisk-users] callback and bridge problem

[Adam KOSA]

Hi guys,

sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).  I've sent this e-mail a couple of days ago, but 
it bounced back today.

i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.

i have troubles getting two calls bridged together.  Scenario is the
following:

- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:

[internal]
; callback is directed here
exten = s,1,WaitExten,50
include = voicemail-context
include = internal_extensions-context
include = dialout_prefix-context


because my call file looks like this:

Channel: SIP/[EMAIL PROTECTED]
Context: internal
Extension: s
Priority: 1

where 0620222 is my cell.

- after picking up, i dial 9520630111 where 952 is the dialing
prefix, 0630... is another cell.  952 is a prefix for another
registered account at the same provider (one account is allowed to place
one call at a time).

After this as you can see, the second number (..) is dialed.
However when i pick up the phone, the call hangs up.

This also happens when i use another prefix (another provider, even
PSTN) for the second call too.

The relevant part from asterisk console is at the end of this e-mail, i
don't really understand the warning messages.

- configs:

In sip.conf, the configuration for the two SIP accounts are:

register = 0621380:[EMAIL PROTECTED]
register = 0621381:[EMAIL PROTECTED]

[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380
authname=0621380
fromuser=0621380
secret=password
callerid=0621380
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no

[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381
authname=0621381
fromuser=0621381
secret=password
callerid=0621381
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no


extension.conf:

exten = _952.,1,Playback(kapcsolas,noanswer)
exten = _952.,n,Set(CALLERID(name)=0621380)
exten = _952.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

I have tried every possible setting i know about, but still, when i call
outside, via 'turning around' in asterisk, both cells hung up when
answering the call.  I have tried calling a regular landline phone
number but still hanging up.

Both accounts are valid, registered and have enough credit to dial
outside its voice network.

The only way the call does not hung up is when i dial extensions within
asterisk.

The asterisk log:

  -- Called [EMAIL PROTECTED]
  -- Call on SIP/neophonex-out-081a9cc0 left from hold
  -- SIP/neophonex-out-081a9cc0 is making progress passing it to
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
  -- Call on SIP/neophonex-out-081a9cc0 left from hold
  -- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
  -- Native bridging SIP/neophonex-081ab240 and
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
== Spawn extension (internal, 9520630111, 3) exited non-zero on
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/[EMAIL PROTECTED]


Please help me to figure out why the calls are hung up.

Thanks
Adam

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[asterisk-users] callback and bridge problem

[Adam KOSA]

Hi guys,

sorry for the long e-mail, i'm only trying to give as much information 
as i think is relevant to my problem (console log, sip.conf and 
extension.conf parts).

i've been practicing with callback for a while, but i'm at a dead end. 
I hope somebody can help me to move on.

i have troubles getting two calls bridged together.  Scenario is the 
following:

- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:

[internal]
; callback is directed here
exten = s,1,WaitExten,50
include = voicemail-context
include = internal_extensions-context
include = dialout_prefix-context


because my call file looks like this:

Channel: SIP/[EMAIL PROTECTED]
Context: internal
Extension: s
Priority: 1

where 0620222 is my cell.

- after picking up, i dial 9520630111 where 952 is the dialing 
prefix, 0630... is another cell.  952 is a prefix for another 
registered account at the same provider (one account is allowed to place 
one call at a time).

After this as you can see, the second number (..) is dialed. 
However when i pick up the phone, the call hangs up.

This also happens when i use another prefix (another provider, even 
PSTN) for the second call too.

The relevant part from asterisk console is at the end of this e-mail, i 
don't really understand the warning messages.

- configs:

In sip.conf, the configuration for the two SIP accounts are:

register = 0621380:[EMAIL PROTECTED]
register = 0621381:[EMAIL PROTECTED]

[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380
authname=0621380
fromuser=0621380
secret=password
callerid=0621380
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no

[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381
authname=0621381
fromuser=0621381
secret=password
callerid=0621381
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no


extension.conf:

exten = _952.,1,Playback(kapcsolas,noanswer)
exten = _952.,n,Set(CALLERID(name)=0621380)
exten = _952.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

I have tried every possible setting i know about, but still, when i call 
outside, via 'turning around' in asterisk, both cells hung up when 
answering the call.  I have tried calling a regular landline phone 
number but still hanging up.

Both accounts are valid, registered and have enough credit to dial 
outside its voice network.

The only way the call does not hung up is when i dial extensions within 
asterisk.

The asterisk log:

 -- Called [EMAIL PROTECTED]
 -- Call on SIP/neophonex-out-081a9cc0 left from hold
 -- SIP/neophonex-out-081a9cc0 is making progress passing it to 
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '[EMAIL PROTECTED]'. Giving up.
 -- Call on SIP/neophonex-out-081a9cc0 left from hold
 -- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
 -- Native bridging SIP/neophonex-081ab240 and 
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839 
handle_response_invite: Re-invite to non-existing call leg on other UA. 
SIP dialog '[EMAIL PROTECTED]'. Giving up.
   == Spawn extension (internal, 9520630111, 3) exited non-zero on 
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call 
completed to SIP/[EMAIL PROTECTED]


Please help me to figure out why the calls are hung up.

Thanks
Adam



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[asterisk-users] voice mail format

[Adam KOSA]

Hi folks,

my goal is to access voicemail (there were some posts about this) but 
not by dialing numbers.  As asterisk sends voicemails in e-mail, it's 
cheaper for us to read e-mails on our cell phone (3g, gprs), and the 
message is attached there.


i've looking around in voicemail.conf and found:
[general]
; Default formats for writing Voicemail
format = wav49|gsm|wav

my phone understands wav49, but some phones do not.  wav is understood 
by all cellphones (every cellphone can play the message) but it's way 
too big.  GSM is again, not understood by all phones.  Is there a way to 
convert voicemail messages to mp3 format?  Should i poke around with 
some hand-crafted application to generate the file, or is there built-in 
way to achieve this?


since i don't know asterisk as much as i'd like, currently it seems that 
it's easier to catch the incoming e-mails sent by asterisk, detach the 
wav attachment, convert it to mp3 and resend to myself (this is not an 
asterisk-related task).  but perhaps asterisk can be told to create the 
messages in mp3 format.


thanks a lot
adam
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Re: [asterisk-users] How to check my voice mail from outside landline?

[Adam KOSA]

Hey Noah,

Noah Miller wrote:

whatever.  That would save you from having to have a dedicated line
for outside access to your extensions.  Or, if you have a live person



currently i just play with *, registering multiple voip accounts to be 
reachable from multiple sip networks.  I'm also exploring the 
possibilities, therefore one dedicated pstn line is no big price.  In 
the future, well, i'm thinking an E1 (as i live in europe) card and a 
professional server to have an IVR, and give a little break to our phone 
support.


thanks for the ideas

best regards
adam
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Re: [asterisk-users] How to check my voice mail from outside landline?

[Adam KOSA]

Hi,

Crazy Boy wrote:
But, I want to check my voice mails by dialing our DID number from a 
outside telephone.




there must be an easier way, but since i only have asterisk and a couple 
of ATAs (spa 3k), i've set one up to give a dial tone to the incoming 
caller on the FXO port.  This way, dialing a pstn number i get another 
dial tone to access internal extensions, such as voicemail.


best regards
adam
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[asterisk-users] duration sec and billing sec in cdr

[Adam KOSA]

Hi guys,

i've installed asterisk to handle multiple voip accounts.  I've looked 
at CDR configs, and managed to have cdr-csv files growing after each 
call.  It would be easier to check my locak asterisk cdr's than logging 
into each account and check them at the provider website.


i found that if i ring my sip softphone from my ata, bill seconds are 
counted correctly.  however, if i call via a voip provider, bill seconds 
are counted incorrectly.  Example:


this call went to a pstn number

New call from 551 --- 94361abcdefg (context: internal)
Dialed: SIP/[EMAIL PROTECTED]
Call start: 2007-04-14 20:10:55
Answered  : 2007-04-14 20:10:55
Call end  : 2007-04-14 20:11:10
Duration  : 15 sec
Bill  : 15 sec


this call went to my ata from the sip softphone:

New call from 551 --- 505 (context: internal)
Dialed: SIP/505|45
Call start: 2007-04-15 07:58:11
Answered  : 2007-04-15 07:58:15
Call end  : 2007-04-15 07:58:43
Duration  : 32 sec
Bill  : 28 sec


i've searched and google'd the wiki, but found only accounting software 
and cdr extensions for providers, but that's not what i need.


thanks for any help
Adam
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Re: [asterisk-users] duration sec and billing sec in cdr

[Adam KOSA]

Hi, and thanks for the suggestions!

Matt wrote:

Sounds like your VoIP provider is incorrectly sending you an Answer before
the call actually completes.  I would contact your VoIP provider.



I suppose it could also be possible that YOU have an Answer() statement 
that

is only on your VoIP trunk.  I would double check that, and then contact
your VoIP provider to see if they have any suggestions.

this is what's most likely as i have no experience in asterisk configs. 
 I've checked the extension.conf settins, they are:


exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])

and for the internal numbers:

exten = _NXZ,1,Set(TIMEOUT(digit)=2)
exten = _NXZ,2,Dial(SIP/${EXTEN},45)
exten = _NXZ,3,VoiceMail(b${EXTEN})
exten = _NXZ,103,VoiceMail(u${EXTEN})


Basically, SOMEONE (your or voipstunt) is answering the call before it
should be answered.



i will check this with more voip providers to see if they or i have 
messed up something (but it's probably going to be me, i just don't know 
where to start looking).


thanks again
Adam
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Re: [asterisk-users] duration sec and billing sec in cdr

[Adam KOSA]

Yossi Ben Hagai wrote:
The Playback command is auto-answering the call. you can use 
Playback(please_wait,noanswer) to fix it.
 


thanks a lot to everyone who answered, this, of course solved this 
issue, it's also in the doc, i just didn't have the idea to look at 
playback's manual :(


regards
adam
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