Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Tzafrir Cohen]

On Tue, Feb 20, 2007 at 11:53:50PM -0600, Eric ManxPower Wieling wrote:
> Tzafrir Cohen wrote:
> 
> >obviously, the makefile used an incorrect "kernel source tree" to build
> >your systems. The package kernel-devel provides a partial kernel source
> >tree which is good enough for building modules (or at least: for
> >building Zaptel). However it seems that the wrong one was used in your
> >case.
> 
> In my experience (Mandrake) the EXTRAVERSION variable in the kernel 
> Makefile does not match running kernel version and so the Zaptel modules 
> are installed in the wrong directory.  Fix that to match what you are 
> running, then run a make clean && make install in the Zaptel directory.

In my experince (Debian) EXTRAVERSION in the .config file is not set,
but the extra version is set from elsewhere (in
include/config/kernel.release I believe).

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Hint a sip account

[Christian Gansberger]

Hi all!

I have Asterisk 1.2.14 (bristuff0.3.0preR8x) installed and i have 2 sip
accounts (A and B) registered at a sip-provider

I want my leds (functionkey 1) on the snom 190 to be lighted when a call
comes in on account A and my
functionkey 2 on account B.

Is there a way to do this with making a hint extension for the SIP account A
and B?
I tried it this way (without success):

sip.conf:
[11]
...
subscribecontext=Snom11

extensions.conf:
[Snom11]
exten => 141,hint,DS/141

[from_Account_A]
exten => ${ACCOUNT_A},1,Devstate(141,2)
exten => ${ACCOUNT_A},2,Dial(SIP/21)
...
exten => h,1,Devstate(141,0)
exten => h,2,Hangup
on the Snom 190: Functionkey 1 - destination - [EMAIL PROTECTED]

in asteriskdb: database put DEVSTATES 141 0

The database value changes on an incoming call, but i get no leds lighted.

Sip log:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.100.231:2060;branch=z9hG4bK-kwu8eayxe3qg;received= 192.168.100.231
;rport=2060
From: >;tag=ho8i2el01x
To: < sip:[EMAIL PROTECTED] 
;user=phone>;tag=as70240b9f
Call-ID: [EMAIL PROTECTED]
CSeq: 29 SUBSCRIBE
User-Agent: Buero Hauer PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5,
realm="asterisk-server
",
nonce="094bd9b6"
Content-Length: 0

I know that this is not the best solution, because when i got 2 calls from
one Sip-account my leds would light
not as they should. But, is there another way to do this?


christian gansberger

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[asterisk-users] How to read channel occupation from PRI INTENSE DEBUG ?

[Olivier]

Hi,

(Apologies for readers of Bristuff mailing as I already posted this message
to the list)

My setup is:
SIP hardphone -  --- Asterisk server
-
Asterisk server is :
Gentoo enabled with 1.0.8 bristuffed Asterisk
equipped with Junghanns Quad BRI with 2 BRI ports connected to ISDN


From log files (with line numbering enabled) I've got :


 2140  < Informational frame:
 2141  < SAPI: 00  C/R: 1 EA: 0
 2142  <  TEI: 064EA: 1
 2143  < N(S): 000   0: 0
 2144  < N(R): 001   P: 0
 2145  < 8 bytes of data
 2146  -- ACKing all packets from 0 to (but not including) 1
 2147  -- Since there was nothing left, stopping T200 counter
 2148  -- Stopping T203 counter since we got an ACK
 2149  -- Nothing left, starting T203 counter
 2150  < Protocol Discriminator: Q.931 (8)  len=8
 2151  < Call Ref: len= 1 (reference 153/0x99) (Terminator)
 2152  < Message type: RELEASE COMPLETE (90)
 2153  < [08 02 87 a2]
 2154  < Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: International network (7)
 2155  <  Ext: 1  Cause: Circuit/channel congestion
(34), class = Network Congestion (2) ]
 2156  Sending Receiver Ready (1)

 2157  > [ 02 81 01 02 ]

 2158  > Supervisory frame:
 2159  > SAPI: 00  C/R: 1 EA: 0
 2160  >  TEI: 064EA: 1
 2161  > Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
 2162  > N(R): 001 P/F: 0
 2163  > 0 bytes of data
 2164  -- Restarting T203 counter
 2165  -- Restarting T203 counter
 2166  -- Channel 0/1, span 2 got hangup
 2167  Feb 19 16:21:03 VERBOSE[10990]: -- Zap/4-1 is circuit-busy
 2168  Feb 19 16:21:03 DEBUG[10990]: Set option AUDIO MODE, value:
ON(1) on Zap/4-1
 2169  Feb 19 16:21:03 DEBUG[10990]: Hangup: channel: 4 index = 0,
normal = 18, callwait = -1, thirdcall = -1
 2170  Feb 19 16:21:03 DEBUG[10990]: Already hungup...  Calling hangup
once, and clearing call
 2171  Feb 19 16:21:03 DEBUG[10990]: disabled echo cancellation on
channel 4
 2172  Feb 19 16:21:03 DEBUG[10990]: Set option TDD MODE, value: OFF(0)
on Zap/4-1
 2173  Feb 19 16:21:03 DEBUG[10990]: Updated conferencing on 4, with 0
conference users
 2174  Feb 19 16:21:03 DEBUG[10990]: Set option AUDIO MODE, value:
OFF(0) on Zap/4-1
 2175  Feb 19 16:21:03 DEBUG[10990]: disabled echo cancellation on
channel 4
 2176  Feb 19 16:21:03 VERBOSE[10990]: -- Hungup 'Zap/4-1'
 2177  Feb 19 16:21:03 VERBOSE[10990]:   == Everyone is busy/congested
at this time
 2178  Feb 19 16:21:03 DEBUG[10990]: Exiting with
DIALSTATUS=CONGESTION.

My feeling (not based on hard facts yet), is that all 4 channels were not
all busy.
How can I can check this reading PRI INTENSE DEBUG SPAN (enormous) output ?

I guess this information should be somehow readable from Channel D .
As PRI INTENSE DEBUG shows both Supervisory and Informational frames, I
guess channel occupation should be in Supervisory frames which I don't know
yet how to read.

How can I check this ?
Regards
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Re: [asterisk-users] Asterisk Inbound Problem

[Arun Kumar]

My service provider only supports g729 and I tried what you have mentioned
here but same thing is happening here. Is there any why that I can see which
codec my service provider is pushing when I'm receiving call on my asterisk
server. When call comes comes to my server and then I type show g729 it
shows 0/0 out of 15 lic.

thanks
arun

On 2/21/07, Mike Lynchfield <[EMAIL PROTECTED]> wrote:


Well, could be the fact provider not pushing as g729 or someting else.

Can you set debug 999 and set verbose 999
then redump that ? you are missing the before the answer part also..

Also try G711 first then work your way to other codecs


On 2/20/07, Rajeev Natarajan <[EMAIL PROTECTED]> wrote:
>
> Am working with Arun on this project - here's a longer description of
> the problem:
>
> We've been fighting with our service provider on this issue - we seem to
> be getting a BYE just after we receive an ACK. They claim that it is an
> asterisk issue! The service provider provides only IP based authentication
> for inbound.
>
> We have used username-password based authentication with the same setup
> with *no problems*  whatsoever!
>
> If we configure an Audiocodes MEdia gateway to receive the calls, there
> is no issue - so there's something that asterisk is doing? or
> asterisk-Provider gateway combo?
>
> In our efforts to mask IP, I have used PROVIDER-IP for the IP of my
> service provider (host) and AsteriskIP to indicate my asterisk server
>
> sip.conf
> [PROVIDER]
> type=peer
> disallow=all
> allow=g729
> context=default
> host=
> fromuser=y.y.y.y
> port=5060
> insecure=very
> canreinvite=no
> nat=yes
> qualify=yes
>
> CLI output:
>
>-- Executing Answer("SIP/PROVIDER-IP-b7a076a8", "") in new stack
> We're at 124.7.195.102 port 47698
> Adding codec 0x100 (g729) to SDP
> Reliably Transmitting (NAT) to PROVIDER-IP:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 
PROVIDER-IP:5060;branch=z9hG4bK6bd3121243ee9f936c4aeb96d6785b7a;received=PROVIDER-IP
>
> From: ;tag=3380976385-794612
> To: ;tag=as52d36855
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: 
> Content-Type: application/sdp
> Content-Length: 183
>
> v=0
> o=root 2172 2172 IN IP4 AsteriskIP
> s=session
> c=IN IP4 AsteriskIP
> t=0 0
> m=audio 47698 RTP/AVP 18
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=silenceSupp:off - - - -
>
> ---
>
>  -- Executing Playback("SIP/PROVIDER-IP-b7a076a8", "park") in new stack
> -- Playing 'park' (language 'en')
> AstSQL*CLI>
> <-- SIP read from PROVIDER-IP:5060:
> ACK sip:[EMAIL PROTECTED] SIP/2.0
> Max-Forwards: 5
> To: ;tag=as52d36855
> From: ;tag=3380976385-794612
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 ACK
> Via: SIP/2.0/UDP 
221.135.102.100:5060;branch=z9hG4bK02505a1dcc5937d9a648eebc0052b422
> Content-Length: 0
>
>
> --- (9 headers 0 lines) ---
> AstSQL*CLI>
> <-- SIP read from PROVIDER-IP:5060:
> BYE sip:[EMAIL PROTECTED] SIP/2.0
> Max-Forwards: 5
> To: ;tag=as52d36855
> From: ;tag=3380976385-794612
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 BYE
> Via: SIP/2.0/UDP 221.135.102.100:5060
> ;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f
> Content-Length: 0
>
>
> --- (9 headers 0 lines) ---
> Sending to PROVIDER-IP : 5060 (NAT)
> Transmitting (NAT) to PROVIDER-IP:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 
PROVIDER-IP:5060;branch=z9hG4bK50997e7192792d429780fc49f7b3f24f;received=PROVIDER-IP
>
> From: ;tag=3380976385-794612
> To: ;tag=as52d36855
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: 
> Content-Length: 0
>
> 

>
> The following is an ngrep of the traffic for an inbound call - 'U' marks
> the begin of the packet grabbed.
>
>
> U :5060 -> :5060
>   INVITE sip:800942@ SIP/2.0..Max-Forwards:
> 5..Session-Expires: 3600;Refresher=uac..Suppor ted: timer..To: <
> sip:[EMAIL PROTECTED]:5060>..From:
> >;tag=3380960452-790279..Co ntact:
> :5060>..Remote-Party-Id:
> >;party=calling;screen=no;privacy =off..Call-ID:
> [EMAIL PROTECTED]: 1
> INVITE..Via: SIP/2.0/UDP 221. 
135.102.100:5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4..Allow-Events:
> telephone-event..Content-T ype: application/sdp..Content-Length:
> 206v=0..o=nextone-msw1 1774 4816 IN IP4 ..s=sip call..c=IN
> IP4 ..t=0 0..m=audio 18932 RTP/AVP 18 19..a=ptime:20..a=rtpmap:19
> CN/8000..a=fm tp:18 annexb=yes..a=rtpmap:18 G729/8000..
>
>
> #
> U :5060 -> :5060
>   SIP/2.0 100 Trying..Via: SIP/2.0/UDP
> :5060;branch=z9hG4bKdc6e0e4db237086a63608e77d7a2eff4;
> received=..From: >;tag=3380960452-790279..To:
> < sip:[EMAIL PROTECTED] 11.2:5060>..Call-ID:
> [EMAIL PROTECTED]: 1
> INVITE..User-Agent: Ast erisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
> REFER, SU

Re: [asterisk-users] Asterisk with Radius users authentication

[Ricardo Carvalho]

Thanks yusuf,

Any other experience on this subject? Anyone knows if Asterisk 1.4 
already implement Radius authentication properly? Has anyone ever 
patched Asterisk with the patch from the Digium Issue Tracker available 
in the URL: http://bugs.digium.com/view.php?id=5424 and got well succeeded?


Thanks once again,
Ricardo.






yusuf wrote:

Ricardo Carvalho wrote:

Dear all,

I've searched the web about Asterisk with Radius integration for user 
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's 
Radius client patch, an still open branch of Digium Issue Tracker 
"SIP peer authentication on an external database (RADIUS - LDAP)", 
etc. Although, none of these seems to give me the confidence to 
implement it in a production environment...


What do you people recommend me? Which Asterisk+Radius solution 
should in your opinion be the best choice? Does Asterisk 1.4 already 
implement it properly?



Thanks in advance,
Ricardo.



Here is a mock-up of what I used to hook-up to a Radius Server, with 
Porta's patch.  It worked quite well for us.  I have'nt used it in 2 
years or so, cant remember much  :)  .  I thin we got it to work by 
seeing the debug (set it in /etc/asterisk/logger.conf) and seeing what 
values were getting sent and recieved.



;exten => _X.,1,SetVar(RADIUS_Server=x.x.x.x)
exten => _X.,2,SetVar(RADIUS_Secret=secret)
exten => _X.,3,SetVar(NAS_IP_Address=x.x.x.x)
exten => _X.,4,SetVar(CALLERID=${CALLERIDNUM})
exten => _X.,5,SetVar(DNID=${EXTEN})
;
; Set account to authorize by
; It can be a prepaid calling card PIN, ANI, or SIP ID depending on 
your application

;
;exten => _X.,6,SetAccount(${CALLERIDNUM})
exten => _X.,6,SetAccount(${CALLERIDNAME})
;
; RADIUS Authorize
; Called as:  
agi-rad-auth.pl|parametr1=value1¶metr2=value2¶metr3=value3

; Possible parametrs:
; Routing=XXX will will send h323-ivr-out = 'PortaBilling_Routing:XXX' 
attribure (XXX is usually SIP)
; AuthorizeBy=SIP requires 
SIPGetHeader(SIP_Authorization=Proxy-Authorization) first + 
externalauth=yes in sip.conf

; AuthorizeBy=Account requires SetAccount() first
; Password=Password optional and may be used together with 
AuthorizeBy=Account
; IfFailed=DoNotHangup optional, used for custome authentication error 
processing i.e. IVR

;
;
exten => 
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=Account&Password=${CALLERIDNUM}&IfFailed=DoNotHangup 

;exten => 
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=Account&Password=Account&IfFailed=DoNotHangup 

;exten => 
_X.,7,agi,agi-rad-auth.pl|AuthorizeBy=Account&IfFailed=DoNotHangup

;
exten => _X.,8,NoOp(${h323-credit-time})
exten => _X.,9, Set(TIMEOUT(absolute)=${h323-credit-time:17})
;exten => _X.,10, AbsoluteTimeout(${h323-credit-time})
exten => _X.,10,Goto(sip-calls,${EXTEN},1)
exten => _X.,11,Hangup
exten => T,1,NoOp(timeout)



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[asterisk-users] AGI DTMF Problem

[Jon Farmer]

Hi

I am writing a IVR app using phpagi and are coming up against a problem when 
trying to detect DTMF. If I use the get_data function I dont seem to be able to 
reliably detect 16 digits. If I try 10 digits then its fine but anything above 
that seems to have a problem.

Any ideas anyone?


 
Jon Farmer
Telford, Shropshire, UK





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[asterisk-users] Dialout option problem in voicemail.conf

[srinivas Antarvedi]

hello all,

i have a set up of 2 contexts with ivr features
and it works fine with voicemail also using callback=somecontext  i can
callback
persons on that context

but problem is if i included third context i can only callback any one
context users
not all users

how can i solve this issue !
plz help me out !

thanks in advance
srinivas antarvedi
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Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

[Charles Wang]

Dear Phil,

Thank you for your reply.

I have changed by extensions.conf as below.
And I also put my debug information for your reference.

It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
If I use my SIP trunking gateway(outside), I got the return value is zero.

** extensions.conf **
exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
exten=> _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
exten=> _00[1-9].,102,Hangup

***
myserver*CLI> agi debug
AGI Debugging Enabled
   -- Seeding '24012100' at 61.217.xxx.xxx:8400 for 60
   -- Accepting AUTHENTICATED call from 61.217.xxx.xxx:
  > requested format = ilbc,
  > requested prefs = (),
  > actual format = ilbc,
  > host prefs = (ilbc),
  > priority = mine
   -- Executing Dial("IAX2/24012100-1", "zap/g1/008621") in new stack
   -- Requested transfer capability: 0x00 - SPEECH
   -- Called g1/0028621
   -- Zap/29-1 is proceeding passing it to IAX2/24012100-1
   -- Zap/29-1 is ringing
   -- Zap/29-1 answered IAX2/24012100-1
   -- Hungup 'Zap/29-1'
 == Spawn extension (default, 008621, 1) exited non-zero on
'IAX2/24012100-1'
   -- Hungup 'IAX2/24012100-1'




2007/2/21, Phil Reynolds <[EMAIL PROTECTED]>:


Quoting Charles Wang <[EMAIL PROTECTED]>:

> Dear all,
>
> I tried to make a call with extensions.conf.
>
> exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
> exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME})
> exten=> _00[1-9].,102,Hangup
>
> But the 2 and 102 will not be executed.
>
> So I can get the correct answered time via 2.
>
> Is any idea about it?

The Dial() exits when the call is finished - then control passes to
the h extension if present.

Therefore, I think you need to put the NoOp in the h extension. It
only continues at 2 if the Dial() times out.

Not sure but that's how I understand it.

--
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95





--

Best Regards
Charles
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[asterisk-users] Channels hanging when SIP phone gets reset during call

[Steve Langstaff]

Hi All.

This is on Asterisk 1.2.13

I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes).

I reset the phones (so they don't have time to say BYE).

Asterisk seems to think that the call is still ongoing. This persists
until I do a 'restart now'.

asterisk1*CLI> show channels
Channel  Location State   Application(Data)
SIP/5301-089fc890(None)   Up  Bridged
Call(SIP/5303-089f1558
SIP/5303-089f1558[EMAIL PROTECTED]:10  Up  Dial(SIP/5301||)
2 active channels
1 active call

asterisk1*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold
Last Message
192.168.5.203line0   2eeb3516264  00103/0  ulaw  No
Tx: ACK
192.168.5.203530328948-0xca0  00102/1  ulaw  No
Tx: ACK
2 active SIP channels


asterisk1*CLI> show hints
asterisk1*CLI>
-= Registered Asterisk Dial Plan Hints =-
   5303: SIP/5303  State:InUse
Watchers  1
   5301: SIP/5301  State:InUse
Watchers  0

- 2 hints registered

I was wondering whether there is anything that I can do about this on
either the Asterisk server or the phone?

__
Steve Langstaff

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[asterisk-users] Re: Setting Caller-ID / Point Codes

[Matt]

Bump.  Nothing heard.

On 2/19/07, Matt <[EMAIL PROTECTED]> wrote:


Greetings folks,
I'm currently dealing with a company to let me set Caller-ID-Name on
outbound calls.  So far pretty happy with their services.  The basic service
works like this:

* CLEC sets Point Code to point to this company
* CLEC has to sign LOA saying they give me permission to set the
Caller-ID-Name through this company.
* I go into web interface and set name.

However, the CLEC is currently asking questions about the LOA, and I am
concerned they may not sign it.

What do other people here know about this procedure.  Have any of you
signed up with a company to allow you to set the Caller-ID-Name?  If so, was
an LOA required?  Did your CLEC sign it?  Who do you all work with?

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[asterisk-users] how to detect who starts one touch recording

[Pavel Jezek]

Is there any way, how to detect, what party starts touch monitor 
recording? is some variable set?
I would like to deliver recorded file after call hangup to that user 
using some shell script.

PJ
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[asterisk-users] Re: The High Performance Echo Canceller (HPEC)

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Boris Bakchiev <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> Has anyone noticed degraded voice quality with HPEC?
> I have a client running TE4XX card who configured HPEC for couple of
> channels with echocancel=1024.
> 
> Whenever HPEC is used you get a background static in voice.
> When HPEC is not used everything is crystal clear.
> 
> What could cause this static?

Try using a utility like "top" to see what the CPU loading is with HPEC.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] Re: The High Performance Echo Canceller (HPEC)

[Boris Bakchiev]

Hi Tony,

Its a dual core system and combined CPU usage was 2%.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Thursday, 22 February 2007 12:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: The High Performance Echo Canceller (HPEC)

In article
<[EMAIL PROTECTED]>,
Boris Bakchiev <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> Has anyone noticed degraded voice quality with HPEC?
> I have a client running TE4XX card who configured HPEC for couple of
> channels with echocancel=1024.
> 
> Whenever HPEC is used you get a background static in voice.
> When HPEC is not used everything is crystal clear.
> 
> What could cause this static?

Try using a utility like "top" to see what the CPU loading is with HPEC.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Re: The High Performance Echo Canceller (HPEC)

[Tzafrir Cohen]

On Wed, Feb 21, 2007 at 01:06:36PM +, Tony Mountifield wrote:
> Try using a utility like "top" to see what the CPU loading is with HPEC.

top itself can give you a pretty god idea, though it may be hard to
separate between zaptel itself and hpec.

oprofile can be handy. You'll probably need to rebuild your kernel to
get a vmlinux (not vmlinu*z*). Debian also has a nice kerneltop. The
latter two (especially kerneltop) proved to nbe not so relieble when it
comes to action in an interrupt context.

-- 
   Tzafrir Cohen   
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[asterisk-users] IAX Realtime - show peers works?

[Enrico Pasqualotto]

hi all, I'm trying to set up some iax2 trunks in Realtime architecture 
with the same backend.
All work better (make call, receive etc etc) but when I do "iax2 show 
peers" some asterisk don't show anything and other show the iax2 peers 
but with status "unknow".


Name/UsernameHost Mask Port 
Status


ctm1/trixbox 10.0.0.131  (S)  255.255.255.255  4569 
UNKNOWN


I have set in iax.conf rtcachefriends=yes but the status not change.
There are anyone with this situation  that iax2 show peers work?

One of my mysql records:

INSERT INTO `iax_buddies` (`name`, `username`, `type`, `secret`, 
`md5secret`, `dbsecret`, `notransfer`, `inkeys`, `outkey`, `auth`, 
`accountcode`, `amaflags`, `callerid`, `context`, `defaultip`, `host`, 
`language`, `mailbox`, `deny`, `permit`, `qualify`, `disallow`, `allow`, 
`ipaddr`, `port`, `regseconds`) VALUES
('ctm2', 'trixbox', 'friend', 'X', NULL, NULL, NULL, NULL, NULL, 
NULL, NULL, NULL, NULL, 'fromiax', NULL, '10.0.0.254', NULL, NULL, NULL, 
NULL, 'yes', 'all', 'alaw', NULL, 4569, 0);


--
Pasqualotto 'Pasqu' Enrico
enrico AT pasqualotto DOT org
web: http://www.pasqualotto.org
skype: epasqualotto
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RE: [asterisk-users] Re: Setting Caller-ID / Point Codes

[Trevor G. Hammonds]

Matt,

A Letter of Agency is almost always signed by the end subscriber and given
to the ILEC/CLEC.  Its purpose is to allow someone other than the subscriber
(e.g. an "Enhanced Service Provider" or consultant) to make changes to, or
get information about, the customer's account (e.g. your account with the
CLEC).  

 

I am not sure how you are set up.  Are you a subscriber of the CLEC, or does
your ITSP get the DIDs from the CLEC?  

 

I would be interested in knowing more about the company you are using for
the CNAM stuff.  If you feel you should not disclose the information
publicly, please e-mail me off list.

 

Sincerely,

Trevor Hammonds

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, February 21, 2007 4:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk Discussion
Subject: [asterisk-users] Re: Setting Caller-ID / Point Codes

 

Bump.  Nothing heard.

On 2/19/07, Matt <[EMAIL PROTECTED]> wrote:

Greetings folks,
I'm currently dealing with a company to let me set Caller-ID-Name on
outbound calls.  So far pretty happy with their services.  The basic service
works like this:

* CLEC sets Point Code to point to this company 
* CLEC has to sign LOA saying they give me permission to set the
Caller-ID-Name through this company.
* I go into web interface and set name.

However, the CLEC is currently asking questions about the LOA, and I am
concerned they may not sign it. 

What do other people here know about this procedure.  Have any of you signed
up with a company to allow you to set the Caller-ID-Name?  If so, was an LOA
required?  Did your CLEC sign it?  Who do you all work with? 

 

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[asterisk-users] Trunk - strange behavior

[Dave Cotton]

Throughout the time I've been using * I've always made tests by calling
out on my SIP provider and calling my fixed line, it's often the only
way of getting an intelligent conversation :).

Since I've been trying trunk I find calls are being put on hold, I even
get music on hold on the calling phone when the called phone answers.

I have just tried calling my mobile using the fixed line and saw this

Call on SIP/2001-081f7e58 left from hold so did have two sound.

The fixed line is connected via an SPA3000 so SIP is there also.

Thoughts anyone.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [asterisk-users] Re: Setting Caller-ID / Point Codes

[Matt]

I am a direct subscriber to the CLEC.   The DIDs have a SPID that belongs to
the CLEC, however the CLEC has given us full control of the numbers, and as
far as they are concerned, they are our numbers.

However, the CNAM company wants the CLEC to sign the LOA, instead of us.

On 2/21/07, Trevor G. Hammonds <[EMAIL PROTECTED]> wrote:


 Matt,

A Letter of Agency is almost always signed by the end subscriber and given
to the ILEC/CLEC.  Its purpose is to allow someone other than the subscriber
(e.g. an "Enhanced Service Provider" or consultant) to make changes to, or
get information about, the customer's account (e.g. your account with the
CLEC).



I am not sure how you are set up.  Are you a subscriber of the CLEC, or
does your ITSP get the DIDs from the CLEC?



I would be interested in knowing more about the company you are using for
the CNAM stuff.  If you feel you should not disclose the information
publicly, please e-mail me off list.



Sincerely,

Trevor Hammonds





*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Matt
*Sent:* Wednesday, February 21, 2007 4:14 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; Commercial
and Business-Oriented Asterisk Discussion
*Subject:* [asterisk-users] Re: Setting Caller-ID / Point Codes



Bump.  Nothing heard.

On 2/19/07, *Matt* <[EMAIL PROTECTED]> wrote:

Greetings folks,
I'm currently dealing with a company to let me set Caller-ID-Name on
outbound calls.  So far pretty happy with their services.  The basic service
works like this:

* CLEC sets Point Code to point to this company
* CLEC has to sign LOA saying they give me permission to set the
Caller-ID-Name through this company.
* I go into web interface and set name.

However, the CLEC is currently asking questions about the LOA, and I am
concerned they may not sign it.

What do other people here know about this procedure.  Have any of you
signed up with a company to allow you to set the Caller-ID-Name?  If so, was
an LOA required?  Did your CLEC sign it?  Who do you all work with?



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RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Carlos Alperin]

Ok,

I understand about the the wrong link, and I agree with that.

There are no kernel-devel-xen. I also tried with yum install
kernel-devel-xen and there was no match for that. But I found 
That the right one is kernel-xen-devel, which I already finished installing.
I find out that there are no xen-headers or headers-xen.

I put a symlink on /usr/src to linux-2.6

  linux-2.6 -> /lib/modules/2.6.19-1.2911.fc6xen/build  
 
I'm going to try again to compile & let you know.

Thanks,

Carlos
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, February 20, 2007 11:52 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

Let's step back a bit,

On Tue, Feb 20, 2007 at 12:57:01PM -0500, Carlos Alperin wrote:
> Tzafrir,
> 
> Sorry, I didn't see this e-mail before:
> 
> > What exactly?
> >  rpm -qa | grep kernel
> 
> [EMAIL PROTECTED] zaptel-1.4.0]# rpm -qa | grep kernel
> kernel-xen-2.6.18-1.2798.fc6
> kernel-xen-2.6.19-1.2911.fc6
> kernel-headers-2.6.19-1.2911.fc6
> kernel-devel-2.6.19-1.2911.fc6
> kernel-devel-2.6.19-1.2911.fc6

obviously, the makefile used an incorrect "kernel source tree" to build your
systems. The package kernel-devel provides a partial kernel source tree
which is good enough for building modules (or at least: for building
Zaptel). However it seems that the wrong one was used in your case.

Does FC6 have a package of the sort of kernel-devel-xen ? Or is one of the
two kernel-devel packages listed above for you intended for the kenel-xen
you have?

Have you generatd the symlink
/lib/modules/kernel-xen-2.6.19-1.2911.fc6/build manually? It seems to point
to an incorrect kernel tree.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Re: Open CallerID Database?

[Natambu Obleton]

Just out of interest: From former posts I understood that there is a
CALLERID service in US (for an extra fee, I assume) that gives both
number _and_ name of the caller...? I am aware of the fact that e.g.
EuroISDN lines can transmit alphanumeric callerid (and in fact I already
use that on an ISDN phone here that connects to an Asterisk - showing a
few "special" names like "wakeup call"). Not for names yet, as I was too
lazy to implement that. Does that also work over analogue lines?


ISDN is america does the same thing, but in Canada( which could be
like europe) the callerid is transfered on the SS7 ISUP message, so
the owner of the number can provide their own callerid. In America
though, this isn't the case. You can not transfer the callerid over
the ISUP message, it must be looked up using TCAP(verisign) or IP
servce(targus).

It's not really getting the CallerID info to the local switch, but
getting it updates in this database in the sky that we are forced to
use. :)


On 2/20/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:

Am Dienstag, den 20.02.2007, 16:33 -0700 schrieb Natambu Obleton:
> I would guess that registration would be by the telco for the blocks
> just like with reverse dns today, so then each telco would have a
> local server to manage their 'reverse' cnam lookup and the people
> in charge would be NANPA, just like how ARIN is regulated today.
> Although who owns the root namservers.. I wonder if ARIN and RIPE
> share ownership of them?
>
> Although now that i Think about it dns wouldn't work because want to
> deligate .. XXX-XXX-XYYY and XXX-XXX-XXYY and then there is single
> numbers. For that right now I do weird PTR CNAME to A record thing for
> single reverse dns. This would be little larger... ohh shit.. LNP. So
> now Qwest would need to deligate a single CNAM to me and crap..naw
> this will never work.

I do not get your point here.

Take ENUM. A made-up phone number like +49 (228) 91234567 would be found
as
7.6.5.4.3.2.1.9.8.2.2.9.4.enum-something

This means, you delegate the 2.1.9.8.2.2.9.4.enum-something to the telco
that owns the 912 block in Bonn, Germany.

Number portability screws this, as even single numbers out of a
contiguous range of MSNs on a ISDN line can be moved over to another
provider, with the others staying with the old provider. They would have
to play well together, and that will "fubar" for sure (they even
sometimes "block" the DSL frequencies on lines when a customer moves to
another company, with "unblocking" taking a 14-day security period,
"because they can" - there is no technical block, but the DSLAM
manipulation database software will mark your DSL line as blocked,
making moving to another provider a pain with up to 6 weeks without
internet).

In Germany we have an agency that manages all phone book data, and
(nearly?) all the 411 type services (called 118xx here) buy data there.
How they manage their data internally is none of my business (although I
would guess it's something like MSSQL with an Access frontend, thinking
about the Deutsche Telekom ;-).

They will no way accept DNS-type queries for free. Some months ago,
there was a heise.de (German IT newsticker) article about them charging
enourmous sums, paying the real data storage and administration costs
back by about factor 5 or so. Well-paying businesses rarely give away
their business turnpoint.

Any non-official system will suffer even worse inaccuracy than the
providers' own and managed system (as someone else already wrote). Their
data is quite bad enough. This relates, of course, to the fact that they
may only reverse-lookup numbers to find names if the customer
explicitely allowed them to do it, on the line rental agreement. There
are usually several checkboxes, allowing you to "get listed in
phonebook", "get listed in digital listings", and "get listed for
reverse lookup".

For those who allow it there is a free web-frontend to reverse-lookup
numbers, which is a pain to script-access, but it is possible. It
suffers from problems with DIDs, as for example a shop might have the
number 94144-0, and assigned the numbers up to 94144-29. If you try to
lookup 9414488 (which might be a private person's analogue line, and
this is absolutely valid in the German numbering system) it will return
the wrong entry because the logic in their webinterface always assumes
that 94144-0 means all numbers starting 94144- belong to that line.
_That_ really sucks - you think business and then it's a friend calling
for private talk.

Just out of interest: From former posts I understood that there is a
CALLERID service in US (for an extra fee, I assume) that gives both
number _and_ name of the caller...? I am aware of the fact that e.g.
EuroISDN lines can transmit alphanumeric callerid (and in fact I already
use that on an ISDN phone here that connects to an Asterisk - showing a
few "special" names like "wakeup call"). Not for names yet, as I was too
lazy to implement that. Does that also work ov

Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Tzafrir Cohen]

On Wed, Feb 21, 2007 at 08:54:33AM -0500, Carlos Alperin wrote:
> Ok,
> 
> I understand about the the wrong link, and I agree with that.
> 
> There are no kernel-devel-xen. I also tried with yum install
> kernel-devel-xen and there was no match for that. But I found 
> That the right one is kernel-xen-devel, which I already finished installing.
> I find out that there are no xen-headers or headers-xen.
> 
> I put a symlink on /usr/src to linux-2.6
> 
>   linux-2.6 -> /lib/modules/2.6.19-1.2911.fc6xen/build  

This seems to be wrong.

Any Fedora kerneller in the crowd?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
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RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Carlos Alperin]

Well,

I tried with the xen-devel and same result.

I tried changing the symlink to  linux-2.6 ->
/lib/modules/2.6.19-1.2911.fc6/build but that is incomplete since build
directory doesn't exists.

Also, I tried with no symlink and on each case:

[EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel
FATAL: Error inserting zaptel
(/lib/modules/2.6.19-1.2911.fc6xen/misc/zaptel.ko): Invalid module format

It looks like allways keeps going to fc6xen/misc directory.

Carlos Alperin

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[asterisk-users] Re: Asterisk CDR MySQL

[Mike Hammett]

I removed Asterisk and reinstalled it from scratch.  It seems to be working
now as module show like cdr now reports many more lines and now mentions
MySQL.

The database is the same as I didn't remove that, just the various files.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, February 21, 2007 3:12 AM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 31, Issue 90

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When replying, please edit your Subject line so it is more specific
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Today's Topics:

   1. Re: Mask the caller-ID (Joanna Liza Mariazeta)
   2. The High Performance Echo Canceller (HPEC) (Boris Bakchiev)
   3. Asterisk behind OpenSER - Getting SIP reinvites towork with
  an ITSP  (Hugo Livude)
   4. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 (Tzafrir Cohen)
   5. Re: They ignore my DTMF! (Pierre Marceau)
   6. Re: Passing a variable from one Asterisk box to   another
  (Justin Newman)
   7. Re: They ignore my DTMF! (Benjamin Jacob)
   8. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
  (Eric "ManxPower" Wieling)
   9. Re: They ignore my DTMF! (Pierre Marceau)
  10. Re: Best FXO Gateway (Martin Joseph)
  11. Re: They ignore my DTMF! (Benjamin Jacob)
  12. Help! How to get ANSWEREDTIME after DIAL a ZAPchannel?
  (Charles Wang)
  13. Re: Asterisk CDR MySQL (Goke Aruna)
  14. Open Source VOIP at Toronto Conference (Evan Leibovitch)
  15. How to repeat pri show span and zap show channel  commands
  (Olivier)
  16. Re: They ignore my DTMF! (Joanna Liza Mariazeta)
  17. How to read "pri intense debug span" data ? (Olivier)
  18. Re: How to repeat pri show span and zap show channel  commands
  (Tzafrir Cohen)
  19. Re: Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64 (Tzafrir Cohen)
  20. Hint a sip account (Christian Gansberger)
  21. How to read channel occupation from PRI INTENSE   DEBUG ? (Olivier)


--

Message: 13
Date: Wed, 21 Feb 2007 07:48:46 +0100
From: Goke Aruna <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Asterisk CDR MySQL
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=windows-1252

Mike Hammett wrote:
>
> Im attempting to setup Asterisk 1.4.0 CDRs to use MySQL.
>
> Modules show like cdr_mysql.so tells me it is loaded.
>
> Reload cdr with MySQL started or stopped makes no difference in the
> errors.
>
> Ideas?
>
> 
>
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>   
do you have cdr_mysql.conf well configured and write permmission granted
to sql user.?

give a verbose and debug to ur logger to know whether asterisk is
attempting login or not.

goksie

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Re: [asterisk-users] trixbox not sending ring back to caller

[Lacy Moore - Aspendora]

On 2/20/07, Marcelo Y <[EMAIL PROTECTED]> wrote:

Any ideas?


Yes, www.trixbox.org
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Re: [asterisk-users] They ignore my DTMF!

[Julio Arruda]

Benjamin Jacob wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the 
ulaw codec.




Out of curiosity, there is any 'document' about how RFC2833 would be 
better or worse than SIP INFO ?





Pierre Marceau wrote:


Okay, in the SPA-941 admin I changed:

;DTMF Tx Method: Auto
DTMF Tx Method: Inband

and now it works.

Thanks!
Pierre

 


[EMAIL PROTECTED] 2/21/2007 12:09 AM >>>
  

Pierre,
Thats exactly what  Joanna  said in her reply.
Check the client DTMF settings on your phones.
set it to rfc2833 or out-of-band, whatever the config says.

Grandstream by default have inband DTMF set, and usualy ulaw is 
supported as well, and thats the reason ur grandstream works but 
others dont.


cheerz
- Ben.

Pierre Marceau wrote:

 


Hi Joanna,

Thanks for your reply.

In my mind I think it must be some setting in the client (phone) 
becasue the Grandstream GXP 2000 does work and it is using the same 
sip.conf


Extensions:
6000 is xlite softfone
6003 is Linksys SPA941
6004 is Grandstream GXP 2000
6005 is Linksys PAP2NA

Please have a look at my sip conf and suggest any changes I could try...

[general]
context=internal
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
type=friend
secret=XXX
nat=no
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
canreinvite=no
register=8885551234:[EMAIL PROTECTED]
[atlasvoice]
type=friend
host=proxy.atlasvoice.com
username=8885551234
secret=XXX
fromuser=8885551234
fromdomain=proxy.atlasvoice.com
canreinvite=no
insecure=very
nat=yes
context=incoming

[6000]
[EMAIL PROTECTED]
[6001]
[6003]
[6004]
[6005]
[6006]
[6007]
[6008]


Thanks,
Pierre




  

[EMAIL PROTECTED] 2/20/2007 10:47 PM >>>
 


Hi Pierre,

Just a thought..check your dtmfmode in your SIP client configuration, if
your using inband but your codec is not ulaw or alaw the DTMF tones 
will be
misrepresented and thus will not be recognised due to the audio 
compression,
on the other hand if your phones are rfc2833 and asterisk is set to 
inband

you wont hear anything.

Hope that helps.

Best Regards,
Joanna

On 2/21/07, Pierre Marceau <[EMAIL PROTECTED]> wrote:


  

Hello,

I can call out to the PSTN and talk to people but when I have to 
enter a
dtmf tone in an ivr or voicemail system those systems do not 
recognise that
I have sent a tone. This is the case when I make the call with the 
Xlite
softfone or a regular telephone plugged into a PAP2NA or a Linksys 
SPA941.


However... a Grandstream GXP2000 works just great ???

All are extensions on my Asterisk 1.4 box. I am using a voip trunk 
through

Atlasvoice. All extensions are setup identical in sip.conf.

One last thing, if a system wants me to respond 1 for sales 2 for 
service
I can hit the 1 button quickly 4 or 5 times and the remote system 
will get
it. That does not work for a three digit extension as you may well 
imagine.


Any help would be appreciated.


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[asterisk-users] Re: Jabber/Asterisk Integration

[Chris Earle]

"agent monitoring screen"?

curious,
which app are you using for that?

--
Chris Earle


"Julian Lyndon-Smith" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
>
> Kyle Sexton wrote:
> > Started playing with 1.4 and I'm curious what uses people have come up
> > with for the Jabber integration?  So far I can think of presence based
> > call routing, but I'm sure there are other ideas.  How are YOU using
> > the new Jabber features in 1.4? :)
> >
>
> We've been using it since July last year (brave / stupid - make your
> choice) for integrating our custom application with the asterisk system.
> The phone system sends all sorts of call information to the agent about
> to receive the call, whilst the agent monitoring screen is used to
> monitor the presence of the agents and their dialplan status (dialling /
> calling / etc etc)
>
> Julian.
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Re: [asterisk-users] SIP interface status and calllimit

[James Fromm]

Anybody seen this behavior?

To determine if it's my config or a bug, could I trouble someone running 
Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as 
a test?  After a few hours a 'sip show inuse' should indicate the 
interface is on calls that it isn't. The incorrect count can be cleared 
up by ringing the interface for how ever many calls are incorrect.


Beware, removing call-limit will require a restart to take effect. 
Thanks in advance for any help.


James Fromm wrote:

It does.

Eric "ManxPower" Wieling wrote:
Maybe Queue doesn't consider a SIP account that returns "BUSY" as "in 
use".  That would be the only case where I could see needing call-limit.


James Fromm wrote:
We do the same thing only we use ringinuse=no and autopause=yes for 
the queue.  With autopause, if the agent is busy their interface in 
the queue gets paused.  Setting call-limit for the SIP interface is 
the only way to make ringinuse=no work.


Eric "ManxPower" Wieling wrote:

James Fromm wrote:

There is an issue when using call-limit for a SIP interface in
sip.conf.  The call count does not properly reset when some calls
end.  The problem happens regardless of which side of the connection
ends the call.  It happens on all calls including calls from SIP
interface to SIP interface (with no reinvite) within the same Asterisk
server.  I have not been able to determine a definite pattern.  I 
can call from one interface to another 50 times before it happens 
and sometimes it happens after only 2 calls.


We have to enable call-limit for our customer service queue agents 
so that the ringinuse option in queues.conf will work properly.


Has anyone else seen this issue?  Any ideas?


This doesn't really help you, but might help others when deciding 
how to design their Asterisk system.  On our phones we set call 
waiting off and each line appearance registers as a separate SIP 
user.  This avoids all this silliness with call limits, group 
limits, etc.  This also allows us total control about which call 
appearance a call shows up on, roll over and hunting features, etc.  
It does require a little more work in the dialplan, but for our 
needs it is well worth it.

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[asterisk-users] jingle + asterisk 1.4

[Rodrigo Gonzalez]

Hi,

can someone give me a link to a howto about that?

I want to use jabbin with asterisk but dont find how to register jabbin 
client in asterisk so it can make calls.


Thanks

Rodrigo
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[asterisk-users] Using asterisk with vpb driver (OpenLine4)

[Yifan Zhang]

Hello, list,

I am using Asterisk with an OpenLine4 card. It worked well with Asterisk 
1.0. Then I upgraded the system, Asterisk 1.4 had some problem to 
compile chan_vpb, but I managed to compile it manually. Still Asterisk 
does not work because it refused to load chan_vpb module.  I had to 
revert back to install Asterisk 1.2. Then I get a error:

asterisk: hip.cpp:793: HipDataPCI::HipDataPCI(int): Assertion `0' failed.

I double checked the vpb-driver, make and make install, modprobe shows 
nothing wrong. dmesg says:


vtcore: VoiceTronix device class interface 4.0 for linux 2.6.19-gentoo-r5
vtopenpci: VoiceTronix OpenPCI card driver 4.0 for linux 2.6.19-gentoo-r5
vtopenpci: module loaded, driver bound to 0 cards

Is there any one have experienced the same problem?

Thanks




--



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Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved

[asterisk]

Did you solved this Problem?

I have the same problem, and i can't solve it, did you know anything 
about?


Thanks

Nico


On Thu, 14 Sep 2006, Kai Militzer wrote:


Hello everyone,

since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that

SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN

What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is no NAT involved and
firewall rules allow the RTP ports defined in rtp.conf on both asterisk
(A and B) machines. The SIP packages look good, no errors messages from
asterisk or anything else, so I have really no idea what causes it and I
cannot reproduce it except by waiting till it happens again. :(

Now the strange thing is, that if I restart the asterisk all works fine
again. A reload does not help, only a restart. Until now I came across
this phenomenon two times on different machines and it all started about
three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
then updated to 1.2.11. I looked through the Changelog but coulnd't find
anything that seems related, but I guess it's a bug that was introduced
somewhere between 1.2.10 and 1.2.11 ...

Does anyone else have similar problems?

Regards,
Kai

--
Kai Militzer WESTEND GmbH  |  Internet-Business-Provider
Technik  CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10  Tel 0241/701333-14
[EMAIL PROTECTED]   D-52064 Aachen  Fax 0241/911879

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[asterisk-users] Zaptel 1.4.0

[Mike Hammett]

I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don't see any errors.  This is out of my modprobe.conf:

 

install tor2 /sbin/modprobe --ignore-install tor2  && /sbin/ztcfg

install torisa /sbin/modprobe --ignore-install torisa  && /sbin/ztcfg

install wcusb /sbin/modprobe --ignore-install wcusb  && /sbin/ztcfg

install wcfxo /sbin/modprobe --ignore-install wcfxo  && /sbin/ztcfg

install wctdm /sbin/modprobe --ignore-install wctdm  && /sbin/ztcfg

install wctdm24xxp /sbin/modprobe --ignore-install wctdm24xxp  &&
/sbin/ztcfg

install ztdynamic /sbin/modprobe --ignore-install ztdynamic  && /sbin/ztcfg

install ztd-eth /sbin/modprobe --ignore-install ztd-eth  && /sbin/ztcfg

install wct1xxp /sbin/modprobe --ignore-install wct1xxp  && /sbin/ztcfg

install wcte11xp /sbin/modprobe --ignore-install wcte11xp  && /sbin/ztcfg

install pciradio /sbin/modprobe --ignore-install pciradio  && /sbin/ztcfg

install ztd-loc /sbin/modprobe --ignore-install ztd-loc  && /sbin/ztcfg

install ztdummy /sbin/modprobe --ignore-install ztdummy  && /sbin/ztcfg

alias wcfxs wctdm

alias wct2xxp wct4xxp

install zttranscode /sbin/modprobe --ignore-install zttranscode  &&
/sbin/ztcfg

install wct4xxp /sbin/modprobe --ignore-install wct4xxp  && /sbin/ztcfg

 

However:

 

[EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel

FATAL: Module zaptel not found.

 

/var/log/dmesg doesn't say anything about zaptel. 

 

 

 

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[asterisk-users] HELP!! Dropping calls on Bridge

[Jason Wolfe]

All calls through the system are being dropped when they are bridged 
(Asterisk, Linux, pure VoIP system). The calling party here's half of 
the word 'hello' for instance and the call is dropped.


I've noticed that hangup() was being called for some time now when the 
call was bridged, but the call was still continuing.


Any thoughts on where to start debugging?

Jason


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Re: [asterisk-users] Zaptel 1.4.0

[Tzafrir Cohen]

On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote:
> I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
> install and I don't see any errors.  This is out of my modprobe.conf:
> 

[ snip ]

> 
> However:
> 
>  
> 
> [EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel
> 
> FATAL: Module zaptel not found.
> 

Any chance that this is just a missing depmod run?

  depmod
  modinfo zaptel

Or maybe you installed the modules to an incorrect directory:

  uname -r
  find /lib/modules -name zaptel.ko

If so, it probably means you built it with incorrect kernel source /
configuration.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Looking for starting point?

[Race Vanderdecken]

As a starting point for Linux installs I would recommend Ubuntu Linux.
 
Easy to setup, you don't need a Linux Administer degree to get started.
 
I stopped using Fedora after the 4th hard disk failure for no reason on
EXT 3.
 
PS
I too am an older developer. Let me know if you need help
programming. I could use some advice on telephone circuits.
 
 
 
Race Vanderdecken
Code Tyrant, Inc.
Somewhere near Asheville, NC.
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary H.
Thompson
Sent: Sunday, February 18, 2007 2:05 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Looking for starting point?
 
Hi,
I am a retired telephone tech/manager who recently had a bad experience
with a local company offering digital phone service (VoIP). I have spent
the last thirty years in the PSTN network, switching, PBX and key system
field and am interested in learning more about VoIP. My background also
includes programming, mostly specialized applications to interact with
the PSTN network. Most of my experience in this field have been with
Borland products, specifically Delphi. I also have been involved with
database programming, same platform as the communications.
My computer experience started with the operating system CPM (I'm not
really that old, only 56). The best platform now seems to be Linux so
now since I am retired now, it seems a good time to learn something new.
I also have been looking at Asterisk which most companies seem to be
using for a PBX platform. I found out by accident that the local company
I had the problem with uses this PBX software.
Could someone steer me in the right direction as to where to start? I
spent most of my career in the telephone industry in a 'bush' area of
Alaska so pretty much had to teach myself what I needed to know about
computers but I can learn almost anything from a book and by asking
questions when I get stuck. Most of my experience was before the
Internet so I plan on using this avenue to advance my knowledge. 
I understand what a broad scope I am asking about so would appreciate
any tips to help me get started. Since there are many 'brands' of Linux
what is the best one to start with? Which Linux will be better when I
get to the point of working with Asterisk? Any tips or ideas on books,
online tutors, discussions or anything of this nature would be much
appreciated.
I hope to add to this group if I can be any assistance from the 'other
side', the PSTN network.
Thank You,
Gary H. Thompson
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Re: [asterisk-users] Digium TE110P

[younss azzayani]

Hi,
thank You,
when i run "zttool i get

Alarms  Span   â
  â UNCONFIGUREDDigium Wildcard TE110P T1/E1 Card 0 â  â
  â UNCONFIGUREDZTDUMMY/1 1 â  â
  â â  â
  â â  â
  â #  â
  â â  â
  â â  â
  â â  â
  â â  â
  â â  â
  ââ
  ââ
  â  ââ    â
  â  â Select â  â Quit â  â
  â  ââ    â
  ââ
  ââ
*
run lsmod
Module  Size  Used by
md5 4033  1
ipv6  235137  12
autofs424773  0
i2c_dev11329  0
i2c_core   22081  1 i2c_dev
sunrpc162725  1
ztdummy 3924  0
wctdm  34880  0
wcfxo  13088  0
wcte11xp   27936  0
wct1xxp19488  0
wct4xxp65600  0
tor2   91936  0
zaptel207748  9
ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
crc_ccitt   2113  1 zaptel
dm_mirror  27825  0
dm_mod 57557  1 dm_mirror
uhci_hcd   31065  0
snd_intel8x0   33897  0
snd_ac97_codec 63889  1 snd_intel8x0
snd_pcm_oss49017  0
snd_mixer_oss  17985  1 snd_pcm_oss
snd_pcm96841  2 snd_intel8x0,snd_pcm_oss
snd_timer  29893  1 snd_pcm
snd_page_alloc  9673  2 snd_intel8x0,snd_pcm
snd_mpu401_uart 8769  1 snd_intel8x0
snd_rawmidi26597  1 snd_mpu401_uart
snd_seq_device  8137  1 snd_rawmidi
snd55461  9
snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device
soundcore   9889  1 snd
3c59x  39293  0
mii 5185  1 3c59x
floppy 58481  0
ext3  116809  2
jbd71385  1 ext3
**
no led is on .. :(
thank you
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[asterisk-users] Trixbox ;TE110P ;DELL OPTIPLEX GX240

[younss azzayani]

Hello every body,
I ve installed Trixbox 1.2 on "DELL OPTIPLEX GX 240", i upgreded it to
the latest version. I've 2 cards installed in the same pci channel via
a bridge or plug i don't know exactly what's his name( but a card (1
pci) that gives 2 pci channels)
the first card is TDM400P: it's ok
the second is TE110P: it's not ok . No led is on neither the red nor the green
when i type zttool, i remark that the TE110P is detected but not configured

My Primary access line has 31 channel that's mean E1 access (no?)
i ve made a cable RJ45 (1->4,2-5);
so my problem is with TE110P,
can someone give me a help please ?
Thank You
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RE: [asterisk-users] HELP!! Dropping calls on Bridge

[Bryan M. Johns]

What asterisk version?

Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: 678.248.2637
Direct: 678.229.1809
http://www.sheltonjohns.com
**Sent from my mobile phone**

-Original Message-
From: "Jason Wolfe" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: 2/21/2007 11:01 AM
Subject: [asterisk-users] HELP!! Dropping calls on Bridge


All calls through the system are being dropped when they are bridged 
(Asterisk, Linux, pure VoIP system). The calling party here's half of 
the word 'hello' for instance and the call is dropped.

I've noticed that hangup() was being called for some time now when the 
call was bridged, but the call was still continuing.

Any thoughts on where to start debugging?

Jason


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[asterisk-users] Zaptel 1.2.14 Released

[Asterisk Development Team]

The Asterisk and Zaptel development team has released version 1.2.14 of
Zaptel.

This release contains only minor changes, the most important of which
relates to single-port module support on Digium's TDM800P analog
interface card (previously these modules were not properly recognized by
the driver).

Thanks for supporting Asterisk and Zaptel!
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Re: [asterisk-users] SIP interface status and calllimit

[Eric \"ManxPower\" Wieling]

Maybe Queue doesn't consider a SIP account that returns "BUSY" as "in 
use".  That would be the only case where I could see needing call-limit.


James Fromm wrote:
We do the same thing only we use ringinuse=no and autopause=yes for the 
queue.  With autopause, if the agent is busy their interface in the 
queue gets paused.  Setting call-limit for the SIP interface is the only 
way to make ringinuse=no work.


Eric "ManxPower" Wieling wrote:

James Fromm wrote:

There is an issue when using call-limit for a SIP interface in
sip.conf.  The call count does not properly reset when some calls
end.  The problem happens regardless of which side of the connection
ends the call.  It happens on all calls including calls from SIP
interface to SIP interface (with no reinvite) within the same Asterisk
server.  I have not been able to determine a definite pattern.  I can 
call from one interface to another 50 times before it happens and 
sometimes it happens after only 2 calls.


We have to enable call-limit for our customer service queue agents so 
that the ringinuse option in queues.conf will work properly.


Has anyone else seen this issue?  Any ideas?


This doesn't really help you, but might help others when deciding how 
to design their Asterisk system.  On our phones we set call waiting 
off and each line appearance registers as a separate SIP user.  This 
avoids all this silliness with call limits, group limits, etc.  This 
also allows us total control about which call appearance a call shows 
up on, roll over and hunting features, etc.  It does require a little 
more work in the dialplan, but for our needs it is well worth it.

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Re: [asterisk-users] Digium TE110P

[Carlos Chavez]

On Wed, 2007-02-21 at 16:14 +, younss azzayani wrote:
> Hi,
> thank You,
> when i run "zttool i get
> 
>  Alarms  Span   â
>â UNCONFIGUREDDigium Wildcard TE110P T1/E1 Card 0 â  â
>â UNCONFIGUREDZTDUMMY/1 1 â  â
>â â  â
>â â  â
>â #  â
>â â  â
>â â  â
>â â  â
>â â  â
>â â  â
>ââ
>ââ
>â  ââ    â
>â  â Select â  â Quit â  â
>â  ââ    â
>ââ
>ââ
> *

What is you /etc/zaptel.conf configuration like?  Have you defined the
span?

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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RE: [asterisk-users] Experiences with FoneBridge2 / TDMoE?

[Michel R Vaillancourt]

Hi, there.  I'm in the process of deploying one at a customer site
so I have a bit of experience with them.  Set up of the unit is trivial...
You create a text file for the config and then use the provided uploader to
send the config to the unit.  Because it *is* TDM, we went with a
direct-wired solution via cross-over cable between the FB2 and a dedicated
NIC on the AstBox;  saves chewing up 1.2Mb constant chatter on the LAN.

Fail-over is done via "heartbeat" monitoring;  when your first
machine drops, the second machine simply uploads a modified config file to
the FB2 and all traffic then starts pouring to the second machine.  Trivial.

The FB2 is *dumb*;  there is no NVRAM, so if the power goes for even
a second, it looses its config.  My answer is a CRON job that simply reloads
the config every minute;  worst case senario is 1 min of PRI unavailability.

Bottom line is this is "KISS"-level kit.  There is nothing to break,
and it just works.  We got one up and running, from the time the config file
was created, in 1 second   Run config util while dialing PRI on my
cellphone and call went through.

We do have one odd issue, which is outbound noise on the line...
Sounds like an electronic snake having a hissy fit in the background.
Inbound calls are crystal clear;  PRI E1(T2) in France.  I've no idea what
the issue is, but its also brand new kit.  I'll be contacting RF support
about it.

The only thing we've found is that on the Asterisk side, the TDMOE
driver is cranky.  Really cranky.  ZTDYNAMIC won't load first try 80% of the
time, ZTCFG fails to configure more than 20 channels 80% of the time, etc.
So, while the FB2 is working fine, it's the Asterisk side that is flakey.

Best of luck in your installation!

--Michel
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[asterisk-users] Monitoring which users are online in realtime

[Ricardo Carvalho]

Hi all,

Is there a way to keep track in Asterisk of which phones are online in 
realtime using some MySQL DB table for exemple, much like "sip show 
peers" does in the CLI?


Regards,
Ricardo.
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Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved

[François Delawarde]

Hi,

I have similar symptoms (usually one-way audio like you, but sometimes 
echoed, distorded, or low volume sound), in a simpler configuration, 
using just SIP with a few phones and a TDM400 card with two FXOs:

Asterisk --> PSTN

I have kernel 2.6.18-XEN and using Asterisk 1.4

François.



[EMAIL PROTECTED] wrote:


Did you solved this Problem?

I have the same problem, and i can't solve it, did you know anything 
about?


Thanks

Nico


On Thu, 14 Sep 2006, Kai Militzer wrote:


Hello everyone,

since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that

SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN

What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is no NAT involved and
firewall rules allow the RTP ports defined in rtp.conf on both asterisk
(A and B) machines. The SIP packages look good, no errors messages from
asterisk or anything else, so I have really no idea what causes it and I
cannot reproduce it except by waiting till it happens again. :(

Now the strange thing is, that if I restart the asterisk all works fine
again. A reload does not help, only a restart. Until now I came across
this phenomenon two times on different machines and it all started about
three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
then updated to 1.2.11. I looked through the Changelog but coulnd't find
anything that seems related, but I guess it's a bug that was introduced
somewhere between 1.2.10 and 1.2.11 ...

Does anyone else have similar problems?

Regards,
Kai

--
Kai Militzer WESTEND GmbH  |  Internet-Business-Provider
Technik  CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10  Tel 0241/701333-14
[EMAIL PROTECTED]   D-52064 Aachen  Fax 0241/911879

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[asterisk-users] Trixbox -- ACPI and IO-APIC?

[Stephen Bosch]

Hi:

Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server
isn't seeing the mainboard's APIC.

-Stephen-
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Re: [asterisk-users] HELP!! Dropping calls on Bridge

[Jason Wolfe]

1.2.1

Jason Wolfe, CTO
Click For A Call, Inc.
[EMAIL PROTECTED]
1-800-218-4951
o (770) 287-0273
c (770) 561-6956

This e-mail transmission may contain information that is proprietary, 
privileged and/or confidential and is intended exclusively for the person(s) to 
whom it is addressed. Any use, copying, retention or disclosure by any person 
other than the intended recipient or the intended recipient's designees is 
strictly prohibited. If you are not the intended recipient or their designee, 
please notify the sender immediately by return e-mail and delete all copies.



Bryan M. Johns wrote:

What asterisk version?

Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: 678.248.2637
Direct: 678.229.1809
http://www.sheltonjohns.com
**Sent from my mobile phone**

-Original Message-
From: "Jason Wolfe" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: 2/21/2007 11:01 AM
Subject: [asterisk-users] HELP!! Dropping calls on Bridge


All calls through the system are being dropped when they are bridged 
(Asterisk, Linux, pure VoIP system). The calling party here's half of 
the word 'hello' for instance and the call is dropped.


I've noticed that hangup() was being called for some time now when the 
call was bridged, but the call was still continuing.


Any thoughts on where to start debugging?

Jason


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Re: [asterisk-users] Monitoring which users are online in realtime

[Philipp Kempgen]

Ricardo Carvalho wrote:

> Is there a way to keep track in Asterisk of which phones are online in 
> realtime using some MySQL DB table for exemple, much like "sip show 
> peers" does in the CLI?

If you are using real realtime with rtupdate=yes in sip.conf
Asterisk stores the current time + sip registration time in
the regseconds column in your sipfriends table (the name depends
on your extconfig.conf). So you could do something like
SELECT * FROM `sipfriends` WHERE `regseconds` > UNIX_TIMESTAMP()
to get a list of the registered SIP friends.

Regards,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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[asterisk-users] Problem Installing Zaptel

[Dovid B]

While trying to compile zaptel 1.2.8 on a FC5 I get the following error:

/lib/modules/2.6.19-1.2288.fc5smp/build
make -C /lib/modules/2.6.19-1.2288.fc5smp/build SUBDIRS=/usr/src/zaptel-1.2.8 
modules
make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2288.fc5-smp-i686'
  CC [M]  /usr/src/zaptel-1.2.8/zaptel.o
In file included from /usr/src/zaptel-1.2.8/zaptel.c:40:
/usr/src/zaptel-1.2.8/zconfig.h:9:26: error: linux/config.h: No such file or 
directory
make[2]: *** [/usr/src/zaptel-1.2.8/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel-1.2.8] Error 2
make[1]: Leaving directory `/usr/src/kernels/2.6.19-1.2288.fc5-smp-i686'
make: *** [linux26] Error 2
[EMAIL PROTECTED] zaptel-1.2.8]# 

I have the sources installed:

[EMAIL PROTECTED] zaptel-1.2.8]# yum list kern*
Loading "installonlyn" plugin
Setting up repositories
core [1/3]
core  100% |=| 1.1 kB00:00 
extras   [2/3]
extras100% |=| 1.1 kB00:00 
updates  [3/3]
updates   100% |=| 1.2 kB00:00 
Reading repository metadata in from local files
primary.xml.gz100% |=| 1.9 MB00:09 
extras: ## 6294/6294
Added 3 new packages, deleted 34 old in 29.48 seconds
primary.xml.gz100% |=| 392 kB00:03 
updates   : ## 1073/1073
Added 5 new packages, deleted 2 old in 4.93 seconds
Installed Packages
kernel.i686  2.6.19-1.2288.fc5  installed   
kernel.i686  2.6.17-1.2174_FC5  installed   
kernel-devel.i6862.6.19-1.2288.fc5  installed   
kernel-smp.i686  2.6.19-1.2288.fc5  installed   
kernel-smp.i686  2.6.17-1.2174_FC5  installed   
kernel-smp-devel.i6862.6.17-1.2174_FC5  installed   
kernel-smp-devel.i6862.6.19-1.2288.fc5  installed   
Available Packages
kernel.i586  2.6.19-1.2288.fc5  updates 
kernel-debug.i6862.6.19-1.2288.fc5  updates 
kernel-debug-devel.i686  2.6.19-1.2288.fc5  updates 
kernel-devel.i5862.6.19-1.2288.fc5  updates 
kernel-doc.noarch2.6.19-1.2288.fc5  updates 
kernel-kdump.i6862.6.19-1.2288.fc5  updates 
kernel-kdump-devel.i686  2.6.19-1.2288.fc5  updates 
kernel-smp.i586  2.6.19-1.2288.fc5  updates 
kernel-smp-debug.i6862.6.19-1.2288.fc5  updates 
kernel-smp-debug-devel.i686  2.6.19-1.2288.fc5  updates 
kernel-smp-devel.i5862.6.19-1.2288.fc5  updates 
kernel-xen.i686  2.6.19-1.2288.fc5  updates 
kernel-xen-devel.i6862.6.19-1.2288.fc5  updates 
kernel-xen0.i686 2.6.19-1.2288.fc5  updates 
kernel-xen0-devel.i686   2.6.19-1.2288.fc5  updates 
kernel-xenU.i686 2.6.19-1.2288.fc5  updates 
kernel-xenU-devel.i686   2.6.19-1.2288.fc5  updates   

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Re: [asterisk-users] Problem Installing Zaptel

[Tzafrir Cohen]

On Wed, Feb 21, 2007 at 08:44:37PM +0200, Dovid B wrote:
> While trying to compile zaptel 1.2.8 on a FC5 I get the following error:
> 
> /lib/modules/2.6.19-1.2288.fc5smp/build
> make -C /lib/modules/2.6.19-1.2288.fc5smp/build SUBDIRS=/usr/src/zaptel-1.2.8 
> modules
> make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2288.fc5-smp-i686'
>   CC [M]  /usr/src/zaptel-1.2.8/zaptel.o
> In file included from /usr/src/zaptel-1.2.8/zaptel.c:40:
> /usr/src/zaptel-1.2.8/zconfig.h:9:26: error: linux/config.h: No such file or 
> directory

This has been resolved in later versions of zaptel. Please try a newer
version (1.2.14 has just been released). The issue should be resolved
there.


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Open CallerID Database?

[Brad Templeton]

On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote:
> Why not make it like DNS and have each provider have their lookups
> deligated to a local server and then each ISP will run a caching
> server that will use a serial number system to get updates.. just like
> DNS.
> 
> I know there are lot more DNS lookups then CNAM lookups per hour...
> isn't there? :)
> 

Hey, we could even build a system where DNS can be used to take any
phone number and look up data about it, not just a name, but even
a URI to redirect calls to for it, a source of presence info and
more.

What a great idea!   Unfortunately, since phone numbers are
believed to be owned by telcos and not by individuals, such
a system would probably make the mistake of delegating control
over the numbers to the telcos, who would feel no particular
motive to help people bypass what they sell, and so I predict
it will languish for a long time with no real deployment in the
USA.

:-)
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[asterisk-users] Problem on Asterisk to Register lines for out/in calls

[Frederico Madeira]

Hi guys,

I have a customer with asterisk registering 100 lines from my Voip Provider.

In some times during a day we receive this messages:

[Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to
schedule in the past?!?!
[Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to
schedule in the past?!?!
[Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to
schedule in the past?!?!
[Feb 21 17:09:34] NOTICE[26223]: chan_sip.c:7085 sip_reg_timeout: --
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #3)
[Feb 21 17:09:34] NOTICE[26223]: chan_sip.c:7085 sip_reg_timeout: --
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #3)

This message happen in all 100 lines.
After few minutes all lines can register in my softswitch.
This problem is not cyclical.

Sniffing the network, i saw that asterisk send a register message,
receive a 407 message and after instead send another register with
authentication header, send another register message without
authentication header.

In most part of the time this asterisk work fine, except for this
problems that happen 4 or 5 times per day.

What could be the cause of this problem ?

Thanks.

--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
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[asterisk-users] Re: Open CallerID Database?

[Benny Amorsen]

> "BT" == Brad Templeton <[EMAIL PROTECTED]> writes:

BT> Hey, we could even build a system where DNS can be used to take
BT> any phone number and look up data about it, not just a name, but
BT> even a URI to redirect calls to for it, a source of presence info
BT> and more.

BT> What a great idea!

I may be missing sarcasm here, but the URI thing is known as ENUM.


/Benny


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[asterisk-users] How does Asterisk use SIP info command

[Yuan LIU]

What Asterisk command I can use to send a SIP INFO command?  Thanks for 
pointers.


Yuan Liu


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[asterisk-users] How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

I have a sip.conf with stanzas for sip phones that have 
'context=sip-incoming for some Grandstream phones and another stanza for 
a Sipura SPA3000 with context=pstn-incoming.


Reviewing the code today, I was dismayed to see that all my outgoing 
extens were mixed into those two.  I have been told this is very insecure.


How can I separate the outgoing extens?

When I create a context [outgoing] in extensions.conf with various 
extens, they never get activated.  How to I get them to dial etc?


Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] monitoring cluster-based call-centers

[Lenz]

Hello list,

we are pleased ro announce that we have released a newer version of  
QueueMetrics (1.3.3) that is able to monitor multiple Asterisk servers at  
once, thus making it possible to monitor call centers running on clusters  
or on high-availability configurations. See  
http://queuemetrics.com/news.jsp


The code for this monitoring is not yet production-quality, but if you run  
such an environment, we would like you to check it out and let us know how  
it goes.


QueueMetrics is a commercial package but it is available free of charge to  
smaller call centers / SOHOs / well-mannered Asterisk hackers. We offer  
free evalutation licences and would really love input from developers and  
system integrators with real-life clustered call-center experience.


Best regards,
l.

--
Loway Research - Home of QueueMetrics
http://queuemetrics.com
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Re: [asterisk-users] Trixbox -- ACPI and IO-APIC?

[Lacy Moore - Aspendora]

On 2/21/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

Hi:

Does Trixbox support


www.trixbox.org
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[asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Benny Amorsen]

> "LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:

LA> I have a sip.conf with stanzas for sip phones that have
LA> 'context=sip-incoming for some Grandstream phones and another
LA> stanza for a Sipura SPA3000 with context=pstn-incoming.

LA> Reviewing the code today, I was dismayed to see that all my
LA> outgoing extens were mixed into those two. I have been told this
LA> is very insecure.

It shouldn't be insecure. It is perhaps a bit unusual to have calls
coming from your own phones labeled "incoming". That is probably the
source of the confusion.


/Benny


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Re: [asterisk-users] Trixbox -- ACPI and IO-APIC?

[Stephen Bosch]

Lacy Moore - Aspendora wrote:
> On 2/21/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:
>> Hi:
>>
>> Does Trixbox support
> 
> www.trixbox.org

Thanks -- I know where the website is :P

Where did you think I got it?

-Stephen-
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[asterisk-users] SIP 406 error - cause?

[Michelle Dupuis]

I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster).  The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
 
I have attached the SIP debug output below.  It looks like codecs overlaps -
can anyone see why the call is being refused?
 
(Note that I'm not registering with the remote SIP device, just sending
directly to it by IP address).
 
Thanks,
Michelle
 


 
Audio is at 99.99.26.93 port 16738
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 100.100.116.29:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" ;tag=as60543531
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 Feb 2007 21:24:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 467
 
v=0
o=root 5921 5921 IN IP4 99.99.26.93
s=session
c=IN IP4 99.99.26.93
t=0 0
m=audio 16738 RTP/AVP 0 3 8 112 5 10 7 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
---
-- Called [EMAIL PROTECTED]
 
<--- SIP read from 100.100.116.29:5060 --->
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" ;tag=as60543531
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-agent: Asterisk PBX
Max-Forwards: 69
Date: Wed, 21 Feb 2007 21:24:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
 
<->
--- (13 headers 0 lines) ---
-- Got SIP response 406 "Not Acceptable" back from 100.100.116.29
Transmitting (no NAT) to 100.100.116.29:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 99.99.26.93:5060;branch=z9hG4bK53811c65;rport
From: "Unknown" ;tag=as60543531
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


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[asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Axel Thimm]

On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64
> very good, but since FC keeps updating, I tried to follow newer kernel
> versions.

If you want to save these hassles, why not use the packages bits that
are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even
packages for the upcoming F7 and RHEL5 available:

http://atrpms.net/name/asterisk/
http://atrpms.net/name/zaptel/

If you add atrpms to your yum config all you have to do is

yum install asterisk zaptel zaptel-kmdl-`uname -r`

If you want yum to automatically install new kmdls for new kernels
also install yum-plugin-kmdl, and then you only need to use yum update
and not worry again (or worry less ...).

> I can't pass the zaptel compilation. Everything is OK, but when I finished,
> and tried to load it, allways got module not found when I run modprobe
> zaptel, and modprobe ztdummy.
>  
> I already tried to modify is with the sed 1 option but doesn't work.
>  
> I'm running make linux26, & make install. Also, I have the kernel sources,
> and a symlink to /lib/modules/
>  
> Also, I tried the make install-udev, since there was no zap device on
> /dev/zap but nothing.
>  
> The error is that when I run modprobe the result is FATAL NO ZAPTEL MODULE
> FOUND.
>  
> Any clue about this?
>  
> Thanks
>  
> Carlos Alperin
-- 
Axel.Thimm at ATrpms.net


pgprRIZT5q7Rf.pgp
Description: PGP signature
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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

Benny Amorsen wrote:

"LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:


LA> I have a sip.conf with stanzas for sip phones that have
LA> 'context=sip-incoming for some Grandstream phones and another
LA> stanza for a Sipura SPA3000 with context=pstn-incoming.

LA> Reviewing the code today, I was dismayed to see that all my
LA> outgoing extens were mixed into those two. I have been told this
LA> is very insecure.

It shouldn't be insecure. It is perhaps a bit unusual to have calls
coming from your own phones labeled "incoming". That is probably the
source of the confusion.


/Benny


Thanks for the response /Benny

My understanding is that the 'context=whatever' in sip.conf is the name 
of the context that handles incoming calls from the outside - that's 
what I meant by 'incoming'.


I'm quite clear that 'incoming' means calls coming in from the outside.

The 'insecure' part is that knowlegable callers _could_ call in and use 
various methods to make outside possibly expensive calls _if_ the 
outgoing extens were in the same context.  I don't want that!


That's why I want a clear separation between the incoming calls from 
outside and calls to the outside made by my phones on the inside.


The problem is I don't know how to use a context in extensions.conf 
without referring to it in sip.conf (context=something).


Do you know of a way?

Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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[asterisk-users] Snom 320 password

[Mike Hammett]

A client of mine has a Snom 320.  Usually when he comes in each morning, it
is asking him for a password.  A power cycle brings it back to normal
operation.  How do I troubleshoot this further?

 

--Mike

 

 

 

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Re: [asterisk-users] Snom 320 password

[Jessee J Holmes]

Mike,

A few things you can try, default administrator password should   
by default. Maybe it just needs that entered.


Otherwise, if the phone is being used with Asterisk, there was a bug  
on an issue like this which may have since been resolved, but non-the- 
less is documented here: http://voipstore.atacomm.com/Support/KB/ 
ViewArticle.aspx/27934028032-1-10.htm


Secondly, if this doesn't work, I'd really suggest the simple  
routine, manufacturer reset followed by the latest stable firmware  
release upgrade.


Instructions for a factory reset can be found here: http:// 
voipstore.atacomm.com/Support/KB/ViewArticle.aspx/27934028032-1-9.htm


Hope that helps,

Jessee Holmes
Atacomm / Ataractic Corporation
www.atacomm.com
V: 1-877-700-VOIP
[EMAIL PROTECTED]

Looking for voice over IP products?  Visit our VoIP store at http:// 
voipstore.atacomm.com/



On Feb 21, 2007, at 4:52 PM, Mike Hammett wrote:

A client of mine has a Snom 320.  Usually when he comes in each  
morning, it is asking him for a password.  A power cycle brings it  
back to normal operation.  How do I troubleshoot this further?




--Mike







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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Paul Hales]

Actually, 'context=' in sip.conf is the first place Asterisk looks for
when a number is dialled from the phone. It then uses 'includes' to
check for other options.

Usually, people use 'incoming' for their external lines, and something
else for the sip phones. I have used 'sip_phones' before.

regards,

PaulH

On Wed, 2007-02-21 at 16:11 -0600, Larry Alkoff wrote:
> Benny Amorsen wrote:
> >> "LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:
> > 
> > LA> I have a sip.conf with stanzas for sip phones that have
> > LA> 'context=sip-incoming for some Grandstream phones and another
> > LA> stanza for a Sipura SPA3000 with context=pstn-incoming.
> > 
> > LA> Reviewing the code today, I was dismayed to see that all my
> > LA> outgoing extens were mixed into those two. I have been told this
> > LA> is very insecure.
> > 
> > It shouldn't be insecure. It is perhaps a bit unusual to have calls
> > coming from your own phones labeled "incoming". That is probably the
> > source of the confusion.
> > 
> > 
> > /Benny
> 
> Thanks for the response /Benny
> 
> My understanding is that the 'context=whatever' in sip.conf is the name 
> of the context that handles incoming calls from the outside - that's 
> what I meant by 'incoming'.
> 
> I'm quite clear that 'incoming' means calls coming in from the outside.
> 
> The 'insecure' part is that knowlegable callers _could_ call in and use 
> various methods to make outside possibly expensive calls _if_ the 
> outgoing extens were in the same context.  I don't want that!
> 
> That's why I want a clear separation between the incoming calls from 
> outside and calls to the outside made by my phones on the inside.
> 
> The problem is I don't know how to use a context in extensions.conf 
> without referring to it in sip.conf (context=something).
> 
> Do you know of a way?
> 
> Larry

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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Eric \"ManxPower\" Wieling]

Put your phones in the context=toll-access in sip.conf or zapata.conf
Put the phone "lines" in context=incoming in sip.conf or zapata.conf

extensions.conf:

[extensions]

exten => 667,1,Dial(SIP/whatever)
...
more exten lines to dial your phones here

[incoming] ; this is where calls from untrusted sources should land

include => extensons

[toll-trunks]

exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}
...
more exten lines to dial outside numbers here

[toll-access]

include => extensions
inclide => toll-trunks



Larry Alkoff wrote:

Benny Amorsen wrote:

"LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:


LA> I have a sip.conf with stanzas for sip phones that have
LA> 'context=sip-incoming for some Grandstream phones and another
LA> stanza for a Sipura SPA3000 with context=pstn-incoming.

LA> Reviewing the code today, I was dismayed to see that all my
LA> outgoing extens were mixed into those two. I have been told this
LA> is very insecure.

It shouldn't be insecure. It is perhaps a bit unusual to have calls
coming from your own phones labeled "incoming". That is probably the
source of the confusion.


/Benny


Thanks for the response /Benny

My understanding is that the 'context=whatever' in sip.conf is the name 
of the context that handles incoming calls from the outside - that's 
what I meant by 'incoming'.


I'm quite clear that 'incoming' means calls coming in from the outside.

The 'insecure' part is that knowlegable callers _could_ call in and use 
various methods to make outside possibly expensive calls _if_ the 
outgoing extens were in the same context.  I don't want that!


That's why I want a clear separation between the incoming calls from 
outside and calls to the outside made by my phones on the inside.


The problem is I don't know how to use a context in extensions.conf 
without referring to it in sip.conf (context=something).


Do you know of a way?

Larry


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Re: [asterisk-users] Passing a variable from one Asterisk box to another

[C F]

You could also use Set(CALLERID(name)=1234*${CALLERID(name)})
and then on the other astereisk server use app_cut to reformat CID and
extract the Var.


On 2/20/07, Eric Bishop <[EMAIL PROTECTED]> wrote:

Hi all,

We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.

For example now on box 1 we have:

exten => _23XX,1,SetVar(Foo=1234)
exten => _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

When the call dials into Box 2 the variable Foo does not get passed...

Does anyone have any clever ideas?



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RE: [asterisk-users] Trixbox -- ACPI and IO-APIC?

[kodak]

Stephen Bosch <> wrote on Wednesday, February 21, 2007 12:26 PM:

> Hi:
> 
> Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox
> server isn't seeing the mainboard's APIC.

TB is really CentOS 4.4, which is really RHEL 4.4.

Now all you have to do is find out if RHEL supports it. :)

--J(K)

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Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Jens Vagelpohl]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1


On 21 Feb 2007, at 23:06, Axel Thimm wrote:


On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on  
FC5 x86_64
very good, but since FC keeps updating, I tried to follow newer  
kernel

versions.


If you want to save these hassles, why not use the packages bits that
are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even
packages for the upcoming F7 and RHEL5 available:


Hi Axel,

I tried to use the 1.2.x RPMs and they would not work for me  
attempting to use them with an Eicon Diva Server card and Melware's  
chan_capi. Only by looking at the SRPM did I notice that they are  
patched with BRIStuff patches, which I have assume causes  
incompatibilities. Compiling Asterisk and Zaptel from sources again  
solved all my problems. It may be helpful to spell out more clearly  
how severaly patched the Asterisk in those RPMs is.


jens



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Version: GnuPG v1.4.5 (Darwin)

iD8DBQFF3Ns2RAx5nvEhZLIRAtmSAJ4/ANMLSgUITOSaITMlxHhxJO1s7ACgjic6
zvPjhF6GkAvTW83JqJOtht0=
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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the inside 
gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand this 
better?


Larry


Eric "ManxPower" Wieling wrote:

Put your phones in the context=toll-access in sip.conf or zapata.conf
Put the phone "lines" in context=incoming in sip.conf or zapata.conf

extensions.conf:

[extensions]

exten => 667,1,Dial(SIP/whatever)
...
more exten lines to dial your phones here

[incoming] ; this is where calls from untrusted sources should land

include => extensons

[toll-trunks]

exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1}
...
more exten lines to dial outside numbers here

[toll-access]

include => extensions
inclide => toll-trunks



Larry Alkoff wrote:

Benny Amorsen wrote:

"LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:


LA> I have a sip.conf with stanzas for sip phones that have
LA> 'context=sip-incoming for some Grandstream phones and another
LA> stanza for a Sipura SPA3000 with context=pstn-incoming.

LA> Reviewing the code today, I was dismayed to see that all my
LA> outgoing extens were mixed into those two. I have been told this
LA> is very insecure.

It shouldn't be insecure. It is perhaps a bit unusual to have calls
coming from your own phones labeled "incoming". That is probably the
source of the confusion.


/Benny


Thanks for the response /Benny

My understanding is that the 'context=whatever' in sip.conf is the 
name of the context that handles incoming calls from the outside - 
that's what I meant by 'incoming'.


I'm quite clear that 'incoming' means calls coming in from the outside.

The 'insecure' part is that knowlegable callers _could_ call in and 
use various methods to make outside possibly expensive calls _if_ the 
outgoing extens were in the same context.  I don't want that!


That's why I want a clear separation between the incoming calls from 
outside and calls to the outside made by my phones on the inside.


The problem is I don't know how to use a context in extensions.conf 
without referring to it in sip.conf (context=something).


Do you know of a way?

Larry


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--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Eric \"ManxPower\" Wieling]

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the inside 
gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand this 
better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will not 
have access to any exten => lines that allow dialing out of the system. 
 include => is only "one-way"

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Re: [asterisk-users] Digium TE110P

[younss azzayani]

this is my zaptel.conf::
[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
# channel 1, WCT1, unhandled for now
# channel 2, WCT1, unhandled for now
# channel 3, WCT1, unhandled for now
# channel 4, WCT1, unhandled for now
# channel 5, WCT1, unhandled for now
# channel 6, WCT1, unhandled for now
# channel 7, WCT1, unhandled for now
# channel 8, WCT1, unhandled for now
# channel 9, WCT1, unhandled for now
# channel 10, WCT1, unhandled for now
# channel 11, WCT1, unhandled for now
# channel 12, WCT1, unhandled for now
# channel 13, WCT1, unhandled for now
# channel 14, WCT1, unhandled for now
# channel 15, WCT1, unhandled for now
# channel 16, WCT1, unhandled for now
# channel 17, WCT1, unhandled for now
# channel 18, WCT1, unhandled for now
# channel 19, WCT1, unhandled for now
# channel 20, WCT1, unhandled for now
# channel 21, WCT1, unhandled for now
# channel 22, WCT1, unhandled for now
# channel 23, WCT1, unhandled for now
# channel 24, WCT1, unhandled for now
# channel 25, WCT1, unhandled for now
# channel 26, WCT1, unhandled for now
# channel 27, WCT1, unhandled for now
# channel 28, WCT1, unhandled for now
# channel 29, WCT1, unhandled for now
# channel 30, WCT1, unhandled for now
# channel 31, WCT1, unhandled for now

# Span 2: ZTDUMMY/1 "ZTDUMMY/1 1"

# Global data

loadzone= us
defaultzone = us
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Re: [asterisk-users] Digium TE110P

[Paul Hales]

genzaptel is _not_ your friend when setting up E1.

PaulH

On Thu, 2007-02-22 at 00:46 +, younss azzayani wrote:
> this is my zaptel.conf::
> [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
> # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
> # Zaptel Configuration File
> #
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
> 
> # It must be in the module loading order
> 
> 
> # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0"
> # channel 1, WCT1, unhandled for now
> # channel 2, WCT1, unhandled for now
> # channel 3, WCT1, unhandled for now
> # channel 4, WCT1, unhandled for now
> # channel 5, WCT1, unhandled for now
> # channel 6, WCT1, unhandled for now
> # channel 7, WCT1, unhandled for now
> # channel 8, WCT1, unhandled for now
> # channel 9, WCT1, unhandled for now
> # channel 10, WCT1, unhandled for now
> # channel 11, WCT1, unhandled for now
> # channel 12, WCT1, unhandled for now
> # channel 13, WCT1, unhandled for now
> # channel 14, WCT1, unhandled for now
> # channel 15, WCT1, unhandled for now
> # channel 16, WCT1, unhandled for now
> # channel 17, WCT1, unhandled for now
> # channel 18, WCT1, unhandled for now
> # channel 19, WCT1, unhandled for now
> # channel 20, WCT1, unhandled for now
> # channel 21, WCT1, unhandled for now
> # channel 22, WCT1, unhandled for now
> # channel 23, WCT1, unhandled for now
> # channel 24, WCT1, unhandled for now
> # channel 25, WCT1, unhandled for now
> # channel 26, WCT1, unhandled for now
> # channel 27, WCT1, unhandled for now
> # channel 28, WCT1, unhandled for now
> # channel 29, WCT1, unhandled for now
> # channel 30, WCT1, unhandled for now
> # channel 31, WCT1, unhandled for now
> 
> # Span 2: ZTDUMMY/1 "ZTDUMMY/1 1"
> 
> # Global data
> 
> loadzone= us
> defaultzone = us
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[asterisk-users] Asterisk to Cisco's Rescue...again...Authenticate LD Calls

[JR Richardson]

Hi All,

 

Just wanted to share a story:

 

We turned up a new customer yesterday evening, typical situation, Cisco 2600
Router with T1 PRI card pointed to the customer's analog PBX with 2 data
T1's linked back to our network.  The router PRI was configured as a gateway
on our CCM 4, like we've done numerous times in the past.  The customer
needed LD Authorization codes enabled, got the list 400+, and configured
them in the CCM, no problem.  We started passing calls, local was fine but
the LD would not work, turned the LD codes off and LD would work.  After
engaging Cisco TAC, was informed that LD coded do not work with this type of
gateway device.

 

After strapping on my Asterisk-Orange Super-Engineer Cape and Goggles, I
told my Cisco Guy to prepend all the LD traffic with a 3 digit code and send
it to one of my Cluster Asterisk Servers.  I put in a pattern for matching
just this customers LD traffic as so:

 

exten => _5551NXXNXX,1,Answer

exten => _5551NXXNXX,2,Set(CDR(userfield)=Company LD)   

exten => _5551NXXNXX,3,Authenticate(/etc/asterisk/companyld.codes|a)

exten => _5551NXXNXX,4,Goto(ccmtrunkld|${EXTEN:3}|1)

 

;Answer because the authenticate cmd sends audio back to the caller

;set the CDR userfield to the company name

;Authenticate with file where the LD codes are stored, 'a' option puts the
LD code in the CDR accountcode field

;strip the 555 off and pass the LD call outbound

 

So now all the LD traffic from this customer can be authenticated from the
codes in the file companyld.codes, the CDR is updated properly for parsing
the LD and generates a nice monthly report for tracking who is using LD for
this customer.

 

I guess I'm feeling grateful that the Cisco Gateway is passing calls in the
first place, but it would have been nice for the Cisco CM and the Cisco
Gateway to play nice together.

 

The real hero here is Asterisk, Digium, and the Community that supports it!

 

Thank you All

 

JR

 

JR Richardson

Engineering for the Masses

 

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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the 
inside gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand this 
better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will not 
have access to any exten => lines that allow dialing out of the system. 
 include => is only "one-way"


I have a feeling that the answer is contained in your words but still 
don't quite get it.


Let me ask this:  How do inside devices get access to [toll-access]?  I 
would like my inside devices to have access to everything unless I 
specifically deny access.


Larry

--
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Using Thunderbird on Linux
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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Eric \"ManxPower\" Wieling]

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the 
inside gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand this 
better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will not 
have access to any exten => lines that allow dialing out of the 
system.  include => is only "one-way"


I have a feeling that the answer is contained in your words but still 
don't quite get it.


Let me ask this:  How do inside devices get access to [toll-access]?  I 
would like my inside devices to have access to everything unless I 
specifically deny access.


Contexts are both one of the most important and most difficult concepts 
to understand in Asterisk.


Calls from inside devices land in the toll-access context in 
extensions.conf.  This is because of the context=toll-access line in 
that device's section of sip.conf.  This context in extensions.conf 
include =>'s the toll-trunks context.  Therefore, the inside device gets 
access to the toll-trunks context.

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RE: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Carlos Alperin]

Axel,

Thanks for your advice, but as I tried to found the real problem overpass
the search is just like close my eyes.

I'm trying to learn in order to not repeat same mistake twice. I don't know
how the rpm's are build, and I don't think that 
You can apply on every kind of variation you can find. Also, there are
additional functions I going to add, that are directly related 
To the full system, so I cannot drive this on blind packages.

Thanks a lot, but I'm going to search for the solution and find what mistake
I made.

Carlos Alperin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Axel Thimm
Sent: Wednesday, February 21, 2007 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 
> x86_64 very good, but since FC keeps updating, I tried to follow newer 
> kernel versions.

If you want to save these hassles, why not use the packages bits that are
available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even packages
for the upcoming F7 and RHEL5 available:

http://atrpms.net/name/asterisk/
http://atrpms.net/name/zaptel/

If you add atrpms to your yum config all you have to do is

yum install asterisk zaptel zaptel-kmdl-`uname -r`

If you want yum to automatically install new kmdls for new kernels also
install yum-plugin-kmdl, and then you only need to use yum update and not
worry again (or worry less ...).

> I can't pass the zaptel compilation. Everything is OK, but when I 
> finished, and tried to load it, allways got module not found when I 
> run modprobe zaptel, and modprobe ztdummy.
>  
> I already tried to modify is with the sed 1 option but doesn't work.
>  
> I'm running make linux26, & make install. Also, I have the kernel 
> sources, and a symlink to /lib/modules/
>  
> Also, I tried the make install-udev, since there was no zap device on 
> /dev/zap but nothing.
>  
> The error is that when I run modprobe the result is FATAL NO ZAPTEL 
> MODULE FOUND.
>  
> Any clue about this?
>  
> Thanks
>  
> Carlos Alperin
--
Axel.Thimm at ATrpms.net

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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the 
inside gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand 
this better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will 
not have access to any exten => lines that allow dialing out of the 
system.  include => is only "one-way"


I have a feeling that the answer is contained in your words but still 
don't quite get it.


Let me ask this:  How do inside devices get access to [toll-access]?  
I would like my inside devices to have access to everything unless I 
specifically deny access.


Contexts are both one of the most important and most difficult concepts 
to understand in Asterisk.


Calls from inside devices land in the toll-access context in 
extensions.conf.  This is because of the context=toll-access line in 
that device's section of sip.conf.  This context in extensions.conf 
include =>'s the toll-trunks context.  Therefore, the inside device gets 
access to the toll-trunks context.


I _think_ we are getting somewhere.

You are essentially saying that, in order to have access to 
[toll-access] I would need a line context=toll-access

in a specific device(s).

In my case, the system is for my house.  So I have it setup to ring 
_all_ phones when a call comes in and would like my wife and I to be 
able to call _anywhere_.  Since we never know which phone will be handy, 
it's necessary to give full access to all phones, which I think means 
context=toll-access in sip.conf for all phones.


Doesn't that give access to any outside caller who can break into the 
system?


Searching voip-info
(my other bible besides "The Future of Telephony" book)
they specically say

"You should consider that if any channel, incoming line, etc can enter 
an extension context that it has the capability of accessing any 
extension within that context.


Therefore, you should NOT allow access to outgoing or toll services in 
contexts that are accessible (especially without a password) from 
incoming channels "


Doesn't that mean that
1.  I have to have context=toll-access]
in any phone that can make toll calls
2,  There is no way to give access to all internal phones unless I 
violate voip-info's security directive above?


Since I can give a password from sip.conf, is there an easy way to 
automatically give that password in calls made from my internal phones
in such a way that external callers won't know the password even if they 
 breach the system?


How do people breach a system anyway?  I've heard about hitting an '*' 
as soon as the connection is made but don't understand it.

Or much else apparently .

Larry

--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Eric \"ManxPower\" Wieling]

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to extensions
in [incoming] also access to toll-trunks?  How does anyone on the 
inside gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by including it
in another context.

I'm probably missing something here.  Can you help me understand 
this better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will 
not have access to any exten => lines that allow dialing out of the 
system.  include => is only "one-way"


I have a feeling that the answer is contained in your words but still 
don't quite get it.


Let me ask this:  How do inside devices get access to [toll-access]?  
I would like my inside devices to have access to everything unless I 
specifically deny access.


Contexts are both one of the most important and most difficult 
concepts to understand in Asterisk.


Calls from inside devices land in the toll-access context in 
extensions.conf.  This is because of the context=toll-access line in 
that device's section of sip.conf.  This context in extensions.conf 
include =>'s the toll-trunks context.  Therefore, the inside device 
gets access to the toll-trunks context.


I _think_ we are getting somewhere.

You are essentially saying that, in order to have access to 
[toll-access] I would need a line context=toll-access

in a specific device(s).

In my case, the system is for my house.  So I have it setup to ring 
_all_ phones when a call comes in and would like my wife and I to be 
able to call _anywhere_.  Since we never know which phone will be handy, 
it's necessary to give full access to all phones, which I think means 
context=toll-access in sip.conf for all phones.


Doesn't that give access to any outside caller who can break into the 
system?


Yes, any phone you want to dialout would have a context=toll-access in 
the device's sip.conf [section].  But that is not a security issue 
because contexts are really something only used for calls from a device 
to Asterisk.  The context= line of a device is ignored when sending 
calls to it.


My examples might be overly complex because I took them from my standard 
context design for production systems in a corporate enviroment where we 
also have contexts like [exten-access] (devices that can only dial 
extensions and 911) and [local-access]/[local-trunks] (devices that can 
only dial extensions, local calls, and 911)

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Re: [asterisk-users] Problem Installing Zaptel

[Dovid B]

- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>

To: 
Sent: Wednesday, February 21, 2007 9:15 PM
Subject: Re: [asterisk-users] Problem Installing Zaptel



On Wed, Feb 21, 2007 at 08:44:37PM +0200, Dovid B wrote:

While trying to compile zaptel 1.2.8 on a FC5 I get the following error:

/lib/modules/2.6.19-1.2288.fc5smp/build
make -C /lib/modules/2.6.19-1.2288.fc5smp/build 
SUBDIRS=/usr/src/zaptel-1.2.8 modules

make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2288.fc5-smp-i686'
  CC [M]  /usr/src/zaptel-1.2.8/zaptel.o
In file included from /usr/src/zaptel-1.2.8/zaptel.c:40:
/usr/src/zaptel-1.2.8/zconfig.h:9:26: error: linux/config.h: No such file 
or directory


This has been resolved in later versions of zaptel. Please try a newer
version (1.2.14 has just been released). The issue should be resolved
there.


That did the trick. Thanks.


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Re: [asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Larry Alkoff]

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Eric "ManxPower" Wieling wrote:

Larry Alkoff wrote:

Hello Eric.

I don't fully understand your example.

I _think_ you have in extensions.conf:

[incoming]
include => extensions

[extensions]
exten => 667
more exten here

[toll-trunks]
exten => 91NXXNXX
more exten here

[toll-access]
include => extensions
include => toll-trunks

My understanding of 'include' is it's as if the 'include'
were typed line by line into the context.

Since both extensions and toll-trunks are mixed together in 
[toll-access], doesn't that give anyone who gains access to 
extensions
in [incoming] also access to toll-trunks?  How does anyone on the 
inside gain access to [toll-access]?


Also I don't understand the 'doubling' of [extensions] by 
including it

in another context.

I'm probably missing something here.  Can you help me understand 
this better?


No.  Any device in the [incoming] context will only have access to 
anything in the [incoming] and [extensions] context.  i.e. it will 
not have access to any exten => lines that allow dialing out of the 
system.  include => is only "one-way"


I have a feeling that the answer is contained in your words but 
still don't quite get it.


Let me ask this:  How do inside devices get access to 
[toll-access]?  I would like my inside devices to have access to 
everything unless I specifically deny access.


Contexts are both one of the most important and most difficult 
concepts to understand in Asterisk.


Calls from inside devices land in the toll-access context in 
extensions.conf.  This is because of the context=toll-access line in 
that device's section of sip.conf.  This context in extensions.conf 
include =>'s the toll-trunks context.  Therefore, the inside device 
gets access to the toll-trunks context.


I _think_ we are getting somewhere.

You are essentially saying that, in order to have access to 
[toll-access] I would need a line context=toll-access

in a specific device(s).

In my case, the system is for my house.  So I have it setup to ring 
_all_ phones when a call comes in and would like my wife and I to be 
able to call _anywhere_.  Since we never know which phone will be 
handy, it's necessary to give full access to all phones, which I think 
means context=toll-access in sip.conf for all phones.


Doesn't that give access to any outside caller who can break into the 
system?


Yes, any phone you want to dialout would have a context=toll-access in 
the device's sip.conf [section].  But that is not a security issue 
because contexts are really something only used for calls from a device 
to Asterisk.  The context= line of a device is ignored when sending 
calls to it.


My examples might be overly complex because I took them from my standard 
context design for production systems in a corporate enviroment where we 
also have contexts like [exten-access] (devices that can only dial 
extensions and 911) and [local-access]/[local-trunks] (devices that can 
only dial extensions, local calls, and 911)


Thanks very much for your definitive statement that [any_context] must 
relate to a sip.conf context=any_context, either directly or via an 
include statement.  I've kinda verified this by experiment but have not 
seen this in the documentation.


If it's not a security issue I might as well have all phones with 
context=default in sip.conf even though voip-info specifically warns 
against that.  Wonder why?


Actually, context=default is what I had before today and nothing has 
happened _yet_.  I'll just have to look for other methods of preventing 
malefactors from accessing toll calls.  I've already blocked (have no 
access to) 900 calls - my wife and I don't use that 


Any final thoughts on my automatic password idea?

Thanks very much for your help.

Larry

--
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Using Thunderbird on Linux
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RE: [asterisk-users] Asterisk to Cisco's Rescue...again...AuthenticateLD Calls

[Jason Aarons \(US\)]

Glad to hear you had a workaround.
 
I would suggest re-queing your TAC case, perhaps you got a outsourced or less 
experienced engineer at Cisco. Their support has varied depending on which 
city/group you get. Some have more experience then others.
 
While your 2600 from 2001 timeframe it should work, you can't run any of  12.4T 
images over the last 3 years without maxing the DRAM/Flash.
 
I've got 1200+ Forced Authorization Codes with 4.1(3)SR1 using 
2811ISRs/VWIC2-1MFT-T1s running 12.4T with both  MGCP and H323 gateways across 
20 sites with no issues. Could be the old 2600s IOS as you mentioned.



From: [EMAIL PROTECTED] on behalf of JR Richardson
Sent: Wed 2/21/2007 8:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk to Cisco's Rescue...again...AuthenticateLD 
Calls



Hi All,

 

Just wanted to share a story:

 

We turned up a new customer yesterday evening, typical situation, Cisco 2600 
Router with T1 PRI card pointed to the customer's analog PBX with 2 data T1's 
linked back to our network.  The router PRI was configured as a gateway on our 
CCM 4, like we've done numerous times in the past.  The customer needed LD 
Authorization codes enabled, got the list 400+, and configured them in the CCM, 
no problem.  We started passing calls, local was fine but the LD would not 
work, turned the LD codes off and LD would work.  After engaging Cisco TAC, was 
informed that LD coded do not work with this type of gateway device.

 

After strapping on my Asterisk-Orange Super-Engineer Cape and Goggles, I told 
my Cisco Guy to prepend all the LD traffic with a 3 digit code and send it to 
one of my Cluster Asterisk Servers.  I put in a pattern for matching just this 
customers LD traffic as so:

 

exten => _5551NXXNXX,1,Answer

exten => _5551NXXNXX,2,Set(CDR(userfield)=Company LD)   

exten => _5551NXXNXX,3,Authenticate(/etc/asterisk/companyld.codes|a)

exten => _5551NXXNXX,4,Goto(ccmtrunkld|${EXTEN:3}|1)

 

;Answer because the authenticate cmd sends audio back to the caller

;set the CDR userfield to the company name

;Authenticate with file where the LD codes are stored, 'a' option puts the LD 
code in the CDR accountcode field

;strip the 555 off and pass the LD call outbound

 

So now all the LD traffic from this customer can be authenticated from the 
codes in the file companyld.codes, the CDR is updated properly for parsing the 
LD and generates a nice monthly report for tracking who is using LD for this 
customer.

 

I guess I'm feeling grateful that the Cisco Gateway is passing calls in the 
first place, but it would have been nice for the Cisco CM and the Cisco Gateway 
to play nice together.

 

The real hero here is Asterisk, Digium, and the Community that supports it!

 

Thank you All

 

JR

 

JR Richardson

Engineering for the Masses

 



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Re: [asterisk-users] Fax with T.38

[Ray Jackson]

Could anybody give me an authoritative answer on whether Asterisk can 
support T.38 pass-through when the clients are behind NAT?  We have 
Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) 
and would love to get T.38 going but have had no luck so far.  The 
following case:


http://bugs.digium.com/view.php?id=7844

...suggests that T.38 *does* now work for clients behind NAT but I have 
the latest SVN trunk but still cannot get it to work?  On the other side 
I have seen on this list only 2 weeks or so ago:


http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html

This suggests that T.38 does *NOT* work behind NAT?  So, can anybody 
save me the trouble and tell me how it is.  Am I on a hiding to nothing 
trying to get T.38 going with NAT?  Please put me out of my misery! :)


Cheers,
Ray

PS. Does anybody know whether OpenPBX would support T.38 and NAT 
configurations?  This was my backup plan if I couldn't get it to go in 
Asterisk.


Thomas Deillon wrote:
Yes, the canreinvite means Re invite, but there is a consequence in 
Asterisk configuration.


For sure, all the signalisation traffic will go through the asterisk … 
but for the RTP traffic?


If canreinvite = No, all RTP traffic will go through the Asterisk 
(useful for NATed phoned without ALG/STUN/…)


If canreinvite = Yes, the phones will try to exchange RTP packets directly.

 

Do you thing there is a way to allow Re Invite (because you’re right) 
without the RTP consequence?


 


Thanks a lot for your help,

 


Thomas

 




*De :* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *De la part de* Rajnish 
Jain

*Envoyé :* lundi, 19. février 2007 16:25
*À :* Asterisk Users Mailing List - Non-Commercial Discussion
*Objet :* Re: [asterisk-users] Fax with T.38

 

A T.38 fax call typically begins as a normal voice media call. The 
call then dynamically switches over T.38 image media on detection of fax 
handshake tones.  The dynamic modification of session from audio to 
image is accomplished through SIP RE-INVITE messages. I would imagine 
canreinvite= flag controls if an end-point is allowed to send/recv 
RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 
to work.


 



 

On 2/19/07, *Thomas Deillon* <[EMAIL PROTECTED] 
> wrote:


Hi all,

I make others tests.
Analog Fax 1 -> PATTON M-ATA -> Asterisk -> PATTON M-ATA -> Analog Fax2

It works only if I use canreinvite= yes.
But all my clients are behind a Nat without ALG or stun stuffs...

Do you know if canreinvite= yes it's the only way to make it works??

Thanks a lot for your help,

Thomas



-Message d'origine-
De: [EMAIL PROTECTED] 
 [mailto: 
[EMAIL PROTECTED] 
] De la part de Thomas 
Deillon

Envoyé: jeudi, 15. février 2007 11:26
À: Asterisk Users Mailing List - Non-Commercial Discussion
Objet: [asterisk-users] Fax with T.38

Hi all,

I make mistakes in my explanation, so I will try to re-explain my problem…

I want to send fax with FoIP.
Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA 
←Analog→ Analog Fax 2


In the Patton SN4960 configuration I have :
profile voip FOIP
codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression
codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression
codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression
dtmf-relay signaling
dejitter-max-delay 100
fax transmission 1 relay t38-udp
fax redundancy low-speed 2 high-speed 1
fax detection fax-frames
modem transmission 1 bypass g711alaw64k
modem bypass-method nse

On Patton M-ATA :
1. codec alaw
2. codec ulaw
3. codec g729
No silence suppression on these codecs.
I not use this option "FAX without T.38(Use G.711 fax)"


On asterisk side I have:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0 
srvlookup=yes
disallow=all
allow=alaw
dtmfmode = rfc2833
rtcachefriends=yes
realm=vtxvoip
useragent=VTX SIP
rtupdate=yes
language=en
tos=184
notifyringing=yes
t38pt_udptl=yes

And t38pt_udptl=yes in the 2 PATTONs sip accounts …


Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 ….
I received T.38 packets from the Patton sn4960 but no T.38 packets go 
through the Asterisk …. And on the asterisk I have 3 WARNINGS:


[Feb 15 10:05:24] WARNING[9167]: channel.c:3033 
ast_channel_make_compatible: No path to translate from 
SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8)
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to 
find a codec translation path from alaw to g729
[Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to 
find a codec translation path from alaw to g729



What I really not understand it's why asterisk try to translate from 
ulaw to g729 !!!
I disallow all and allow just the alaw codec … more than this, I remove 
the g729 licence file …


Do you have an id

[asterisk-users] SIP response 603 driving me nuts

[Eric Bishop]

I have one Asterisk box registering to another via SIP and on the registar
console I keep getting:

-- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx

Anyone know how to turn off this "feature"?
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Re: [asterisk-users] Trixbox -- ACPI and IO-APIC?

[Stephen Bosch]

[EMAIL PROTECTED] wrote:
> Stephen Bosch <> wrote on Wednesday, February 21, 2007 12:26 PM:
> 
>> Hi:
>>
>> Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox
>> server isn't seeing the mainboard's APIC.
> 
> TB is really CentOS 4.4, which is really RHEL 4.4.
> 
> Now all you have to do is find out if RHEL supports it. :)

My point is that if it's going to involve rebuilding a kernel to support
IO-APIC, then I'd just as soon build from the ground up.

-Stephen-
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RE: [asterisk-users] Fax with T.38

[Dan Austin]

Ray wrote:
> Could anybody give me an authoritative answer on whether 
> Asterisk can support T.38 pass-through when the clients 
> are behind NAT?  We have Asterisk servicing clients behind
> NAT (with nat=route, canreinvite=no) and would love to get
> T.38 going but have had no luck so far.  The following case:

> http://bugs.digium.com/view.php?id=7844

Authoritative?  Nope.  But I'll try to help anyways...
1.  t38pt_udptl must be set to yes in [general] in sip.conf

> ...suggests that T.38 *does* now work for clients behind NAT 
> but I have the latest SVN trunk but still cannot get it to work?
> On the other side I have seen on this list only 2 weeks or so ago:

>
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.ht
ml

> This suggests that T.38 does *NOT* work behind NAT?  So, can
> anybody save me the trouble and tell me how it is.  Am I on a 
> hiding to nothing trying to get T.38 going with NAT?  Please put 
> me out of my misery! :)
Part of an age old issue that doesn't bear repeating, but is also not
terribly accurate or relevant.

> Cheers,
> Ray

Capture a debug log of a failed T.38 session and post it on Mantis.
Make sure to set:
>core set verbose 4
>core set debug 4
>sip set debug

Testing and (what little) feedback the developers have received indicate
that it SHOULD work with the latest SVN.

> PS. Does anybody know whether OpenPBX would support T.38 and NAT 
> configurations?  This was my backup plan if I couldn't get it to go in

> Asterisk.

No idea.

Dan
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Re: [asterisk-users] Passing a variable from one Asterisk box toanother

[Craig Guy]

Hi Richard,

there was a thread regarding this a while ago on the dev list which resulted 
in a patch being made to allow variable passing via IAX2 channels.  See 
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in 
SVN or anyhow, is not in 1.2


I have recently backported this patch to 1.2 and have a patch which is 
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at least 
1.2.13 and 1.2.14.  The patch introduces a new dialplan function called 
IAXVAR, Email me if interested.


Craig

- Original Message - 
From: "Richard Lyman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box 
toanother




Richard Lyman wrote:

Eric Bishop wrote:

Hi all,

We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.

For example now on box 1 we have:

exten => _23XX,1,SetVar(Foo=1234)
exten => _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

When the call dials into Box 2 the variable Foo does not get passed...

Does anyone have any clever ideas?

as noted in asterisk/docs/README.variables (iirc)

you should see that variable inheritance can occur by prefacing the 
variable with '_' or '__'


also, depending on the age of your asterisk you might want to start using 
'Set' vice 'SetVar'


also, having ${EXTEN:0} , the :0 doesn't do anything, so you should not 
use it and just have ${EXTEN}


i hope this helps



sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or 
doesn't accept them iirc)


this does not happen using friend, but that has its own issues... check 
the wiki for more thoughts about this.




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Re: [asterisk-users] Digium TE110P

[Tzafrir Cohen]

On Thu, Feb 22, 2007 at 11:58:06AM +1100, Paul Hales wrote:
> 
> genzaptel is _not_ your friend when setting up E1.

/usr/local/sbin/genzaptelconf that comes with trixbox: no. It is very
old copy of genzaptelconf. 

try xpp/utils/genzaptelconf for something that has supported E1 for
quite a while (in this case it will at least know that this is an E1,
due to the number of channels: there's no other hint, and this seems to
be Asterisk's method of detection as well).

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Tzafrir Cohen]

On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> 
> On 21 Feb 2007, at 23:06, Axel Thimm wrote:
> 
> >On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> >>I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on  
> >>FC5 x86_64
> >>very good, but since FC keeps updating, I tried to follow newer  
> >>kernel
> >>versions.
> >
> >If you want to save these hassles, why not use the packages bits that
> >are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even
> >packages for the upcoming F7 and RHEL5 available:
> 
> Hi Axel,
> 
> I tried to use the 1.2.x RPMs and they would not work for me  
> attempting to use them with an Eicon Diva Server card and Melware's  
> chan_capi. Only by looking at the SRPM did I notice that they are  
> patched with BRIStuff patches, which I have assume causes  
> incompatibilities. 

Why is the a problem? The bristuff zaptel patch is a really small and
non-intrussive one. The bristuff Asterisk patch, though, includes a 
complete reimplementation of chan_capi (the Junghanns' original
chan_capi), which I heard noone really uses.

> Compiling Asterisk and Zaptel from sources again  
> solved all my problems. It may be helpful to spell out more clearly  
> how severaly patched the Asterisk in those RPMs is.

I wonder if it is just Asterisk or Asterisk and Zaptel.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
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Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

[Charles Wang]

Dear Phil,

The extension 'h' was a great idea although I still got the error
"exited non-zero".

Thank you for your help.

Best regards,
Charles

2007/2/21, Phil Reynolds <[EMAIL PROTECTED]>:


Quoting Charles Wang <[EMAIL PROTECTED]>:

> Dear Phil,
>
> Thank you for your reply.
>
> I have changed by extensions.conf as below.
> And I also put my debug information for your reference.
>
> It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
> If I use my SIP trunking gateway(outside), I got the return value is zero.
>
> ** extensions.conf **
> exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN})
> exten=> _00[1-9].,h,NoOP(ANSWEREDTIME=${ANSWEREDTIME})

Still wrong... exten => h,1,NoOp...

> exten=> _00[1-9].,102,Hangup

This line is superfluous.

--
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95





--

Best Regards
Charles
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Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Tzafrir Cohen]

On Wed, Feb 21, 2007 at 08:35:12PM -0500, Carlos Alperin wrote:
> Axel,
> 
> Thanks for your advice, but as I tried to found the real problem overpass
> the search is just like close my eyes.
> 
> I'm trying to learn in order to not repeat same mistake twice. I don't know
> how the rpm's are build, and I don't think that 
> You can apply on every kind of variation you can find. Also, there are
> additional functions I going to add, that are directly related 
> To the full system, so I cannot drive this on blind packages.

Learn to (re)build packages. I consider packages as reproducable builds.
Even more so when you have more than one system.

> 
> Thanks a lot, but I'm going to search for the solution and find what mistake
> I made.

I'm not sure what it is, but several distributions (Debian for quite a
while, Fedora in recent versions) have handy tools for building kernel
module packages. The packages that are used in trixbox (I forgot the
origin) don't use those tools, but the packages from at-rpms seem to use
them. 

Your problem may be missing proper build reuirements.

That said, I'm still waiting for a Fedora guru to look into that thread.

(As for other distributions: SuSE has had kernel modules packages for
even longer than Debian, but I'm not sure about the tools for building
them with multiple kernels and such. Mandriva seems to be leaning
towards dkms recently. Ubuntu uses Debian's tools, and with anything
related to Asterisk: repackages Debian's packages. I don't know about
gentoo)

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Re: How to separate outgoing extens from the contexts from sip.conf?

[Benny Amorsen]

> "LA" == Larry Alkoff <[EMAIL PROTECTED]> writes:

LA> If it's not a security issue I might as well have all phones with
LA> context=default in sip.conf even though voip-info specifically
LA> warns against that. Wonder why?

Random SIP calls from the internet could end up in context default, if
that is the default context mentioned in sip.conf. Then anyone on the
Internet can use your outgoing lines.


/Benny



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[asterisk-users] Re: Snom 320 password

[Benny Amorsen]

> "MH" == Mike Hammett <[EMAIL PROTECTED]> writes:

MH> A client of mine has a Snom 320. Usually when he comes in each
MH> morning, it is asking him for a password. A power cycle brings it
MH> back to normal operation. How do I troubleshoot this further?

It isn't necessary to power cycle, it's enough to just hit X.

First you send nasty thoughts towards Snom headquarters. After that,
you go into the web interface and turn off Challenge Response on
Phone. *poof*, problem gone.


/Benny


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[asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Axel Thimm]

On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> 
> On 21 Feb 2007, at 23:06, Axel Thimm wrote:
> 
> >On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> >>I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on  
> >>FC5 x86_64
> >>very good, but since FC keeps updating, I tried to follow newer  
> >>kernel
> >>versions.
> >
> >If you want to save these hassles, why not use the packages bits that
> >are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even
> >packages for the upcoming F7 and RHEL5 available:
> 
> Hi Axel,
> 
> I tried to use the 1.2.x RPMs and they would not work for me  
> attempting to use them with an Eicon Diva Server card and Melware's  
> chan_capi. Only by looking at the SRPM did I notice that they are  
> patched with BRIStuff patches, which I have assume causes  
> incompatibilities. Compiling Asterisk and Zaptel from sources again  
> solved all my problems. It may be helpful to spell out more clearly  
> how severaly patched the Asterisk in those RPMs is.

bristuff is the only patch in functionality, and for 1.2.15 I need to
drop it again, because it does not apply and there is no upstream
equivalent (why isn't bristuff merged in?). But I don't think bristuff
should have done you any harm.

But anything that should be tunable is in the sense that if you really
think asterisk should be build w/o bristuff you get the src.rpm and do
a

rpmbuild --rebuild --without bristuff asterisk-xyz.src.rpm

It would be worth while to try to stick to the packages either simply
using them, or building from them, or modify them, as if there
are any issues they can be ironed out, which then benefits more users.
-- 
Axel.Thimm at ATrpms.net


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[asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Axel Thimm]

BTW Carlos, all your posts are with "Importance: High" in this thread.

On Wed, Feb 21, 2007 at 08:35:12PM -0500, Carlos Alperin wrote:
> Axel,
> 
> Thanks for your advice, but as I tried to found the real problem overpass
> the search is just like close my eyes.

Not really what I was suggesting, see below.

> I'm trying to learn in order to not repeat same mistake twice. I don't know
> how the rpm's are build, and I don't think that 
> You can apply on every kind of variation you can find. Also, there are
> additional functions I going to add, that are directly related 
> To the full system, so I cannot drive this on blind packages.
> 
> Thanks a lot, but I'm going to search for the solution and find what mistake
> I made.

Packages, either rpms, debs, ebuilds or any other technology allow you
not only to blindly use them, but in fact you can rebuild them, modify
them source-wise at will or simply be reviewed for seeing how to build
something.

A src.rpm is a collection of source, patches and a specfile which is
more or less a recipe on how to build and install the
sources/patches. So if you want to go the build-from-scratch route,
you can look at how the packages were built, and either manually
repeat the steps or letting rpmbuild do that for you.

And if you need modifications you can do that at package level, too, a
patch usually takes two lines to be added. Finally if your
modifications could be generally useful you can submit the changes
back for the next package update to cover them.

> Carlos Alperin 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Axel Thimm
> Sent: Wednesday, February 21, 2007 5:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64
> 
> On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> > I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 
> > x86_64 very good, but since FC keeps updating, I tried to follow newer 
> > kernel versions.
> 
> If you want to save these hassles, why not use the packages bits that are
> available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even packages
> for the upcoming F7 and RHEL5 available:
> 
> http://atrpms.net/name/asterisk/
> http://atrpms.net/name/zaptel/
> 
> If you add atrpms to your yum config all you have to do is
> 
> yum install asterisk zaptel zaptel-kmdl-`uname -r`
> 
> If you want yum to automatically install new kmdls for new kernels also
> install yum-plugin-kmdl, and then you only need to use yum update and not
> worry again (or worry less ...).
> 
> > I can't pass the zaptel compilation. Everything is OK, but when I 
> > finished, and tried to load it, allways got module not found when I 
> > run modprobe zaptel, and modprobe ztdummy.
> >  
> > I already tried to modify is with the sed 1 option but doesn't work.
> >  
> > I'm running make linux26, & make install. Also, I have the kernel 
> > sources, and a symlink to /lib/modules/
> >  
> > Also, I tried the make install-udev, since there was no zap device on 
> > /dev/zap but nothing.
> >  
> > The error is that when I run modprobe the result is FATAL NO ZAPTEL 
> > MODULE FOUND.
> >  
> > Any clue about this?
> >  
> > Thanks
> >  
> > Carlos Alperin

-- 
Axel.Thimm at ATrpms.net


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Re: [asterisk-users] SIP response 603 driving me nuts

[Davy Chan]

**>I have one Asterisk box registering to another via SIP and on the registar
**>console I keep getting:
**>
**>-- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx
**>
**>Anyone know how to turn off this "feature"?

Look at:

http://lists.digium.com/pipermail/asterisk-users/2007-February/179168.html

The message is popping up because Asterisk's new behavior to
SIP NOTIFY messages carrying Message Waiting Indication (MWI) info.

See ya...

d.c.
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Re: [asterisk-users] Re: FW: zaptel 1.4.0 on Fedora Core 6 x86_64

[Tzafrir Cohen]

On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote:
> On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> > 
> > 
> > On 21 Feb 2007, at 23:06, Axel Thimm wrote:
> > 
> > >On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
> > >>I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on  
> > >>FC5 x86_64
> > >>very good, but since FC keeps updating, I tried to follow newer  
> > >>kernel
> > >>versions.
> > >
> > >If you want to save these hassles, why not use the packages bits that
> > >are available for FC5/FC6/RHEL4/RHEL3 i386/x86_64/ppc? There are even
> > >packages for the upcoming F7 and RHEL5 available:
> > 
> > Hi Axel,
> > 
> > I tried to use the 1.2.x RPMs and they would not work for me  
> > attempting to use them with an Eicon Diva Server card and Melware's  
> > chan_capi. Only by looking at the SRPM did I notice that they are  
> > patched with BRIStuff patches, which I have assume causes  
> > incompatibilities. Compiling Asterisk and Zaptel from sources again  
> > solved all my problems. It may be helpful to spell out more clearly  
> > how severaly patched the Asterisk in those RPMs is.
> 
> bristuff is the only patch in functionality, and for 1.2.15 I need to
> drop it again, because it does not apply 

Gee, it shows you're not on the bristuff list. Up-to-date bristuff
patch for Asterisk:

http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/bristuff.dpatch?op=file&rev=0&sc=0

BTW: the one for Zaptel:

http://svn.debian.org/wsvn/pkg-voip/zaptel/trunk/debian/patches/bristuff.dpatch?op=file&rev=0&sc=0

> and there is no upstream equivalent (why isn't bristuff merged in?). 

Ask Kapejod. 

> But I don't think bristuff should have done you any harm.

bristuff has many changes to the ISDN stack. Some may disapprove of
them. It is considered as better in many accounts.

The chan_capi included in bristuff is probably not as good as
chan_capi-cm from sourceforge. In Debian we simply delete all the capi
modules from the bristuff version.

There are also a number of nice small applications, and a bunch of fixes
(e.g: related to snom phones).

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] SIP response 603 driving me nuts

[Eric Bishop]

Surely there must be a simpler way than patching the Asterisk code? After
all this is Asterisk-to-Asterisk registration not some third party
softswitch idiosyncrasy. Would setting up fake voicemail boxes help?


On 2/22/07, Davy Chan <[EMAIL PROTECTED]> wrote:


**>I have one Asterisk box registering to another via SIP and on the
registar
**>console I keep getting:
**>
**>-- Got SIP response 603 "Declined (no dialog)" back from xxx.xxx.xxx.xx
**>
**>Anyone know how to turn off this "feature"?

Look at:

http://lists.digium.com/pipermail/asterisk-users/2007-February/179168.html

The message is popping up because Asterisk's new behavior to
SIP NOTIFY messages carrying Message Waiting Indication (MWI) info.

See ya...

d.c.

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