[Asterisk-Users] Outbound calling number problem

[Pedro Vela]

Hello,

We have a E400P and when we make a call we can´t change de calling number in
my ISDN primary number range. We have 5 number over them and even have the
first one identity in the called party.

The "pri debug" says:

"> Calling Number (len= 4) [Ext: 0 TON: Unknown Number Type (0) ..."

What is wrong ?

Regards,
Pedro

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AW: [Asterisk-Users] CAPI problems when loading chan_capi.so

[Sascha Knific]

Hi

> capiinfo gives:
> ---
> capi not installed - No such device or address (6)
> ---

It´s not just about installing the apropriate package but you have to load
the capi kernel module for your isdn card.

The module to load on boot time is set in /etc/isdn/capi.conf (on Debian).
You have to check how it´s done on your distro (I presume RedHat or SuSE).

You can load the module manually. For a AVM Fritz!Card PCI you would do:
"modprobe fcpci"


Sascha

---
Sascha Knific   K Systems & Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
[EMAIL PROTECTED] http://www.k-sysdes.net


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[Asterisk-Users] Answering Machine Detection

[Navnit Chachan]

Hi,
How do I detect an Answering Machine in Asterisk.
I saw a post by Francois Lambert on 19 Jan. but am unable to get his email
id.
http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html

Can somebody please help?

Thank you
Navnit

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Re: [Asterisk-Users] CAPI problems when loading chan_capi.so

[Martin Mielke]

Hi there,

Martin List-Petersen wrote:

Hi Martin,

Have you checked the rights of your /dev/capi20* interfaces ? 
 

pbx:~ # ls -l /dev/capi*
crw-rw1 root dialout   68,   0 Sep 23  2003 /dev/capi20
crw-rw1 root dialout   68,   1 Sep 23  2003 /dev/capi20.00
crw-rw1 root dialout   68,   2 Sep 23  2003 /dev/capi20.01
crw-rw1 root dialout   68,   3 Sep 23  2003 /dev/capi20.02
crw-rw1 root dialout   68,   4 Sep 23  2003 /dev/capi20.03
crw-rw1 root dialout   68,   5 Sep 23  2003 /dev/capi20.04
crw-rw1 root dialout   68,   6 Sep 23  2003 /dev/capi20.05
crw-rw1 root dialout   68,   7 Sep 23  2003 /dev/capi20.06
crw-rw1 root dialout   68,   8 Sep 23  2003 /dev/capi20.07
crw-rw1 root dialout   68,   9 Sep 23  2003 /dev/capi20.08
crw-rw1 root dialout   68,  10 Sep 23  2003 /dev/capi20.09
crw-rw1 root dialout   68,  11 Sep 23  2003 /dev/capi20.10
crw-rw1 root dialout   68,  12 Sep 23  2003 /dev/capi20.11
crw-rw1 root dialout   68,  13 Sep 23  2003 /dev/capi20.12
crw-rw1 root dialout   68,  14 Sep 23  2003 /dev/capi20.13
crw-rw1 root dialout   68,  15 Sep 23  2003 /dev/capi20.14
crw-rw1 root dialout   68,  16 Sep 23  2003 /dev/capi20.15
crw-rw1 root dialout   68,  17 Sep 23  2003 /dev/capi20.16
crw-rw1 root dialout   68,  18 Sep 23  2003 /dev/capi20.17
crw-rw1 root dialout   68,  19 Sep 23  2003 /dev/capi20.18
crw-rw1 root dialout   68,  20 Sep 23  2003 /dev/capi20.19
/dev/capi:
total 114
drwxr-xr-x2 root root0 Mar 30 19:19 .
drwxr-xr-x   32 root root   116416 Mar 30 19:17 ..

Do you run asterisk as a user or root ?
 

It's running as root.

Either capi is not installed correctly (check with capiinfo) or you have not
given the user asterisk is using rights to access the capi devices.
 

capiinfo gives:
---
capi not installed - No such device or address (6)
---
How does it come? The capi-packages are installed, as I showed yesterday:
---
pbx:~ # rpm -qa | grep capi
avmfritzcapi-1.0-194
capisuite-0.4.3-52
capi4linux-2003.9.17-7
---
Do I need the -devel ones? :-/

Martin

[ snip ]

PS: please don't CC me your replies :-)



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[Asterisk-Users] Asterisk and picoCell GSM Base Stations

[Simon Anderson]

picoCell/microCell/nanoCell Base stations are low powered GSM nodes
designed for indoor or small area GSM coverage. Their transmission power
is low enough to meet the legal restrictions of many countries.

Here are two examples; 

http://www.rivanetworks.com/nano/nano.htm
http://www.ipaccess.com/ipaccess_pages2/bts2.html

Both of these Base Stations have an RJ-45 in order to do Ethernet on the
back end; GSM in, IP out.

I wonder if anyone has experience interfacing one of these or a similar
product with Asterisk?

I also wonder if Digium has any plans to supply PCI cards which provide
similar functionality?

The prospect of recycling old cell phones for use as Asterisk extensions
is extremely attractive.

Cellular phone ---GSM---> picoCell ---IP---> Asterisk

Regards,
Simon.

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RE: [Asterisk-Users] Exception flag set - snom200

[jc]

Asterisk CVS-03/11/04
18:18:12 

snom200-SIP 2.03o

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
Sent: Wednesday, March 31, 2004
1:01 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Exception flag set - snom200

 

What version of asterisk
are you using, and what version of the SNOM firmware?

 

--Ernest



 







From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jc
Sent: Tuesday, March 30, 2004
10:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Exception flag set - snom200

Sorry I forgot the
subject in the last post.

 

When my snom200 receives
an inbound SIP external sip call, it somehow rejects the call and with a busy
tone.  The debug shows the following error:

 

channel.c:1142 ast_read:
Exception flag set on 'SIP/sipphone-7796', but no exception handler

 

 

what does this mean and
how can I debug it further??

 

Thanks 

JC

 

Re: [Asterisk-Users] Caller entered digits ignored during wait....

[Stig Andersson]

Asterisk doesn't accept keys during wait, use Background 
and play 1 sec silence instead.

/Stig

At 23:46 2004-03-30 -0600, you wrote:
>
>On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote:
>
>> How would you use the t extension to accomplish this?
>
>exten => s,1,Wait(1)
>exten => s,2,Answer
>exten => s,3,SetVar(loopCnt=0)
>exten => s,4,Background(welcome)
>exten => s,5,Background(parties)
>
>exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1])
>exten => t,2,GotoIf($[${loopCnt} < 3]?s|4)
>exten => t,3,Background(vm-goodbye)
>exten => t,4,Hangup
>
>-Tilghman
>
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[Asterisk-Users] Can't talk on Cisco VIP 30 using Chan Skinny

[Dean]

I have gotten some cisco VIP 12 and VIP 30 IP phones that I would like
to use with asterisk, I have set them up using chan_skinny. The phones
work well, except the only problem is that it is like the cisco phones
are muted. When I talk on the cisco phones I can hear my self through
the ear peice, but the person who I am calling can not hear me at all. I
have tried various cisco phones from various sources on 2 different
linux computers (one running redhat 7.3 and one running redhat 9) I have
tried using the 0.7.2 code and the latest development code from the CVS
and I still get the same results. All help will be greatly appreciated.
Below is the error log and my skinny.conf

Thanks,

Dean

Error Log:

Mar 31 00:09:29 WARNING[1024]: Ignoring port for now
Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource
temporarily unavailable
Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink
Mar 31 00:16:48 WARNING[1024]: Ignoring port for now
Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource
temporarily unavailable
Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink
Mar 31 00:22:30 WARNING[16401]: No audio available on
Skinny/[EMAIL PROTECTED]


Console:

skinny_answer(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED]
Recieved Open Recieve Channel Ack
us port: 17874
sin port: 53316
-- Playing 'voicemail/default/1234/unavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing: 
/var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav49,
0x80dc958
-- x=1, open writing: 
/var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: gsm,
0x811a3f8
-- x=2, open writing: 
/var/spool/asterisk/voicemail/default/1234/INBOX/msg0004 format: wav,
0x811a570
Mar 31 00:22:30 WARNING[16401]: app_voicemail.c:1222 play_and_record: No
audio available on Skinny/[EMAIL PROTECTED]
-- User hung up
  == Spawn extension (demo, 1235, 1) exited non-zero on
'Skinny/[EMAIL PROTECTED]'
skinny_hangup(Skinny/[EMAIL PROTECTED]) on [EMAIL PROTECTED]

Error Log

Mar 31 00:09:29 WARNING[1024]: Ignoring port for now
Mar 31 00:09:29 WARNING[10251]: Read error on sound device: Resource
temporarily unavailable
Mar 31 00:09:29 WARNING[1024]: Ignoring rxwink
Mar 31 00:16:48 WARNING[1024]: Ignoring port for now
Mar 31 00:16:48 WARNING[10251]: Read error on sound device: Resource
temporarily unavailable
Mar 31 00:16:48 WARNING[1024]: Ignoring rxwink
Mar 31 00:22:30 WARNING[16401]: No audio available on
Skinny/[EMAIL PROTECTED]

;
; Skinny Configuration for Asterisk
;
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 0.0.0.0  ; Address to bind to
dateFormat = M-D-Y  ; M,D,Y in any order (5 chars max)
keepAlive = 120 

;allow = all
;disallow = 


; Typical config for 12SP+
[florian]
device=SEP
version=P002G204; Thanks critch
context=demo
line => 120 ; Dial(Skinny/[EMAIL PROTECTED])

; Typical config for 12SP+
[florianx]
device=SEP
version=P0020301003; Thanks critch
context=default
line => 122 ; Dial(Skinny/[EMAIL PROTECTED])

; Typical config for 12SP+
[flex]
device=SEP
version=P002F202
context=demo
line => 133


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[Asterisk-Users] Register vith SIP provider from behind NAT

[Simon Brown]

I cannot successfully register with, or even make calls to, a SIP provider
(such as FWD) with my * server sitting behind a NAT.  The firewall is a Cisco
827 router running 12.3 IOS.

Has anyone successfully got their server behind NAT to register or make a
call to a SIP provider?

TIA 

Simon

-
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Re: [Asterisk-Users] IAX2 trunk mode over satellite

[clive18]

Hi

I have even used H323 over satelite, and beside the lagg,
no trouble.  My only issue is the jitter buffer on IAX2
seems to be broken. On a very jittery connection, I can
hardly make a decent call on IAX2.

Good luck!
regards
Clive


On Tue, 30 Mar 2004 10:15:52 -0800
 John Todd <[EMAIL PROTECTED]> wrote:
> 
> Today has been the day for satellite questions,
> apparently, so I'll proxy one out to the rest of the
> community...  I asked this tangentially a month or two
> ago, but I'll put it in a more blunt way:
> 
> If you have IAX2 trunking mode experience over satellite,
> please let us know your experiences with that
> protocol/transport combination.
> 
> I've got several people asking about IAX2 and trunk mode
> over satellite.  I have not experimented with IAX2 over
> satellite (though I have used IAX1 over satellite) and
> I'm wondering if anyone has direct experiences with
> IAX2's jitter buffer control over such long-latency
> connections.
> 
> I've had SIP working very well over satellite (despite
> what some people have found to the contrary on this list)
> and other than the lag there have been no issues that
> have come up on a reasonably-managed satellite segment.
>  However, the IP overhead really starts to cost
> significant amounts of pennies when you add it up on
> multiple SIP RTP sessions over the same link.  Plus,
> packet contention and buffering may (_may_) be an issue
> when pushing multiple simultaneous streams out the same
> transponder.
> 
> It would seem to me that IAX2 in trunk mode would be
> optimal for people on very expensive satellite bandwidth,
> as a G.729 9.6kbps channel starts to actually look like
> 9.6kbps instead of 24kbps. However, I have had mixed
> success with IAX2 in certain circumstances. Before I
> start to ask for favors and get satellite time for
> testing, I'd like to see if anyone else has performed
> this experiment.  If you'd wish to remain anonymous,
> please mail me directly and I'll appropriately trim
> identity information and re-distribute, or re-write as
> appropriate.
> 
> Other hints I have heard/used on VoIP over satellite:
>- use small transmit cell (packet) sizes on your
> satellite gear
>- turn off error correction (why use it for VoIP?)
>- turn off compression (G.729 is already compressed;
> you ARE using 
> G.729, right?)
>- ensure minimal latency on the terrestrial portions
> of the call
>- tell your users to suck it up and deal with the
> half-second lag
> 
> JT
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Re: [Asterisk-Users] Sipcall.co.uk & [*]

[Dave Cotton]

On Wed, 2004-03-31 at 01:33, Matt wrote:
> Hello all.
> 
> Has anyone managed to get SIPCALL.co.uk's service working with the [*] box?
> 
> I've managed to register with other SIP providers but not SIPcall.
> 
I spent a lot of time trying to get * to connect to SIPcall, I even got
directly in contact with the support depart of the supplier of the
hardware, who informed me that it is because * does not handle SIP
correctly, as I had no trouble connecting to SIPPhone, Nikotel, VoIPTalk
etc I decided to drop it. 
 YMMV

-- 
Dave Cotton <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Caller entered digits ignored during wait....

[Tilghman Lesher]

On 2004 Mar 30, at 20:56, Gene Kochanowsky wrote:

How would you use the t extension to accomplish this?
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,SetVar(loopCnt=0)
exten => s,4,Background(welcome)
exten => s,5,Background(parties)
exten => t,1,SetVar(loopCnt=$[${loopCnt} + 1])
exten => t,2,GotoIf($[${loopCnt} < 3]?s|4)
exten => t,3,Background(vm-goodbye)
exten => t,4,Hangup
-Tilghman

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[Asterisk-Users] DTMF Detection Problem

[Ron McMillin]

Hi,
  My set up is like this
Asterisk--->SipuraATA->AnalogPhone
When I'm calling into asterisk from a cell phone, there's no dtmf detection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain, etc. on the ATA and it helps a little bit, but not much. And this seems to be a problem only if I call in from a cell phone. If I were to use a SIP phone to call in, it works much better.
 
Is there a way to make Asterisk to regenerate the DTMF tones to improve the DTMF tones? Such as making it interpret the DTMF tones and regenerate it w/ a certain length regardless of original signal length. The reason I want to DTMF comes to AnalogPhone clearly is because I want to ultimately connect it to a FXSFXO converter and go back out to PSTN line.
 
Thank you
Ron

Re: [Asterisk-Users] Asterisk Security Audit?

[Steven Critchfield]

On Tue, 2004-03-30 at 16:53, Jim Rosenberg wrote:
> Has Asterisk ever been audited for common security holes, such as buffer
> overruns?
> 
> A quick grep through the source for routines that should never be used,
> like strcpy, strcat, etc., reveals a lot of it. I fear I fear.

These functions aren't as bad as you make out. They are only dangerous
when used with unchecked buffers that where accepted from outside
sources. There are quite a few instances of strcpy and strcat that are
using string constants and therefore are safe. 

Don't take that as an argument against checking other possible security
concerns. Just as a reminder that the mere existence of certain
functions doesn't mean it is unsafe.

Also this discussion is probably better dealt with on the -dev list
where the noise level is better suited for the developers you need to
target to actually see this message. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes

[Sam Bingner]

* listens for fax tones as soon as you "Answer()" the line.  If you Answer
the line before ringing the local lines, it will actually detect fax tones
while in the Dial statement.

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird
Sent: Sunday, March 28, 2004 5:52 PM
To: Martin List-Petersen
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes



On Mar 28, 2004, at 7:40 PM, Martin List-Petersen wrote:
>
>> ; I'm using a shared analog line for testing this, so I'm using the
>> fax
>> ; autodetection code to yank faxes out of my IVR and into the 'fax'
>> ; pseudo-extension
>> [outside]
>>...
>>exten => fax,1,Goto(fax,2201,1)
>
> I would be interested in how you do fax autodetection.

I don't do anything particularly special, Asterisk just makes it work.
This is using a bog-standard POTS line at home.  Here's the relevant
part of my config:

[macro-outsideline]
   exten => s,1,LookupCIDName
   exten => s,2,SetMusicOnHold(random)
   exten => s,3,Dial(${PHONES},13,Ttm)
   exten => s,4,Answer
   exten => s,5,Goto(outside-ivr,s,1)

[outside-ivr]
   ; This is the outside IVR
   ; Playback a "We're not home message"
   ; To leave a message for Scott, press 1
   ; To leave a message for C, press 2
   ; Otherwise stay on the line.
   ;
   ; Also, 3 => main voicemail
   ;   4 => check voicemail (main)
   ;   5 => check voicemail
   ;   6 => DISA (with password)
   ;
   ; Check for fax, too

   exten => s,1,NoOp
   exten => s,2,DigitTimeout(5)
   exten => s,3,ResponseTimeout(2)
   exten => s,4,Wait(1)
   exten => s,5,Background(laird/ivr-greeting)

   exten => t,1,VoiceMail(s2201)
   exten => t,2,Hangup

   ; other stuff goes here, but it's not really important

   exten => fax,1,Answer
   exten => fax,2,Goto(fax,2201,1)

[outside]
   exten => s,1,Macro(outsideline)
   exten => fax,1,Goto(fax,2201,1)


95% of this isn't important for faxing, but I included it for context.
The big issue is the IVR stuff and the 'fax' extension.  Once we get to
the IVR, asterisk is listening for DTMF tones and apparently also fax
tones.  If it hears a fax, then it goes to the 'fax' extension.  That's
it.


Scott


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smime.p7s
Description: S/MIME cryptographic signature

Re: [Asterisk-Users] Voicemail retrieval from Cisco 7960

[Heison Chak]

You need to setup the 7960 message button with an uri.

SIP.cnf:
messages_uri: "8599"

extensions.conf:
exten => 8599,1,VoicemailMain2(${CALLERIDNUM})
exten => 8599,2,Hangup

-Heison



On Wed, Mar 31, 2004 at 12:38:33PM +0800, Radius wrote:
> Hi all,
> 
> I installed a new Cisco 7960 running SIP. I can make/receive calls to/from other 
> extensions and leave voicemails. The LED on the handset turned RED, indicating 
> voicemail for 7960. When I pressed the message button, 7960 gave me a dial tone 
> only. What should I do to configure 7960 for voicemail retrieval??
> 
> Thanks.
> 
> Ben
> 
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[Asterisk-Users] Voicemail retrieval from Cisco 7960

[Radius]

Hi all,
 
I installed a new Cisco 7960 running SIP. 
I can make/receive calls to/from other extensions and leave voicemails. The LED 
on the handset turned RED, indicating voicemail for 7960. When I pressed 
the message button, 7960 gave me a dial tone only. What should I do to 
configure 7960 for voicemail retrieval??
 
Thanks.
 
Ben
 

RE: [Asterisk-Users] Programming an unlocked ADSI Astra 390 phone?

[Gene Kochanowsky]

I have managed to program a phone however the voicemail option doesn't
appear to work correctly. When I press the VMail key it dials the number
to get into voice mail by the screen shows: 

Comedian Mail 
download refused 
Services is full

What gives? Also is there any way to get the programming specs for ADSI.
It would be great to get these phones to do more tricks but other than
what little I have managed to do, I am completely lost.

Gene

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gene
Kochanowsky
Sent: Sunday, March 28, 2004 11:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Programming an unlocked ADSI Astra 390 phone?

Greetings, 

I have just purchased several Astra 390 phones ready for asterisk. I
have placed a line with 

adsi=yes 

in the Zapata.conf file just before 

channel => 13

I have also added an extension

exten => 6199,1,ADSIProg(asterisk.adsi)  
exten => 6199,2,Hangup

in the extensions.conf file.

When I try to program the phone I get the following:

Asterisk CVS-03/28/04-12:02:10, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

=
Connected to Asterisk CVS-03/28/04-12:02:10 currently running on
asterisk (pid = 3328)
-- Remote UNIX connection
-- Starting simple switch on 'Zap/13-1'
-- Executing ADSIProg("Zap/13-1", "asterisk.adsi") in new stack
-- ADSI Unavailable on CPE.  Not bothering to try.
-- Executing Hangup("Zap/13-1", "") in new stack
  == Spawn extension (local, 6199, 2) exited non-zero on 'Zap/13-1'
-- Hungup 'Zap/13-1'
asterisk*CLI>


I am using a Zhone ZPlex 10B channel bank. Why is it telling me that
ADSI unavailable on CPE? What do I have to do to get this to work? Also
the ADSI documentation is very spotty.

Gene Kochanowsky

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Re: [Asterisk-Users] H323 in Asterisk

[Adam Hart]

Read README under channels/h323, it should point you in the right direction

Terence Parker wrote:

I have posted before but didn't get any replies so i'll ask again in a 
more simple way :

Does H323 work on asterisk out of the box? I notice there is already a 
channels/chan_h323.c file, but creating an h323.conf file I can't seem 
to get H323 working.

Do I have to compile an additional package first or something?

I tried the asterisk-oh323 thing, but can't get it to compile.

Terence

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RE: [Asterisk-Users] Asterisk Security Audit?

[John Vogel]

If you ever get an answer to this please let me know off-line,
[EMAIL PROTECTED]

I have a security expert friend using Asterisk who is interested in running
a whole set of such tests on it. My theory is it is security swiss cheese.

Thanks, John V. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Rosenberg
Sent: Tuesday, March 30, 2004 2:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Security Audit?

Has Asterisk ever been audited for common security holes, such as buffer
overruns?

A quick grep through the source for routines that should never be used, like
strcpy, strcat, etc., reveals a lot of it. I fear I fear.

Has anyone flung pathology at IAX2 to see if it stands up to malformed
packets? (This is always an issue when you have a protocol that only a small
number of programs use ...)

I hope I'm wrong, but I have a very queasy feeling ...

[We already know that H.323 is not being looked after, security-wise ...]
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RE: [Asterisk-Users] H323 in Asterisk

[Keith D'Atrio]

Out of the box it is not supported exactly. You do have to compile
openh-323 and then there is a openh323-driver for asterisk that must be
compiled. I have the steps at work and if you need any more help let me
know. If you read the Readme files in the tarballs you will see how to
do it. I know htat sounds like a blow off or a rtfm but really this is
how I got it working with some help from the irc channel. They pointed
me in the right direction. 

Keith D'Atrio
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terence
Parker
Sent: Tuesday, March 30, 2004 10:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] H323 in Asterisk

I have posted before but didn't get any replies so i'll ask again in a
more simple way :

Does H323 work on asterisk out of the box? I notice there is already a
channels/chan_h323.c file, but creating an h323.conf file I can't seem
to get H323 working.

Do I have to compile an additional package first or something?

I tried the asterisk-oh323 thing, but can't get it to compile.

Terence

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RE: [Asterisk-Users] setting up 7940

[Simon Brown]

Make sure that you have the following (or equivalent in the
SIP.conf
# Line 1 appearance
line1_name: 202
# Line 1 Registration Authentication 
line1_authname: "202"
# Line 1 Registration Password
line1_password: "202" 

and this in the SIPDefault.conf
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

Simon

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
Sent: Wednesday, 31 March 2004 13:07
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] setting up 7940

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3.  
I've downloaded/installed asterisk via cvs.

I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks
box & phone are on the same network.  I've gone through and setup a test
account per the instructions @

http://voip-info.org/wiki-Asterisk+phone+cisco+79xx

but time I do a

sip show peers

*CLI> sip show peers
Name/usernameHost Mask Port Status
3014/3014(Unspecified)   (D)  255.255.255.255  0Unmonitored

I get this.  Also the cisco phone says "phone unprovisioned".  I've tried
telneting to the phone, but it looks like I'm in user mode, can't setup a
username/password.  So far I've just be able to log in using the default
username/password

I've put the following

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level : 2 ; 0-Disabled (default), 1-Enabled, 
2-Privileged

in the SIP.cnf on the tftp server thinking that this would help but
I still can't setup a username/password combo.

The quick-start guide

http://www.voip-info.org/wiki-Asterisk+quickstart

I got lost between creating a sip account and the demo.

Just need some help getting off the ground w/ my 7940
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-
This mail was content checked for malicious code and viruses
by GFI MailSecurity.

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[Asterisk-Users] setting up 7940

[Roger]

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3.  
I've downloaded/installed asterisk via cvs.

I've set the phone up to get its info via dhcp - the dhcp, tftp, 
astericks box & phone are on the same network.  I've gone through and 
setup a test account per the instructions @

http://voip-info.org/wiki-Asterisk+phone+cisco+79xx

but time I do a

sip show peers

*CLI> sip show peers
Name/usernameHost Mask Port Status
3014/3014(Unspecified)   (D)  255.255.255.255  0Unmonitored
I get this.  Also the cisco phone says "phone unprovisioned".  I've 
tried telneting to the phone, but it looks like I'm in user mode, can't 
setup a username/password.  So far I've just be able to log in using the 
default username/password

I've put the following

# Telnet Level (enable or disable the ability to telnet into the phone)
telnet_level : 2 ; 0-Disabled (default), 1-Enabled, 
2-Privileged

in the SIP.cnf on the tftp server thinking that this would 
help but I still can't setup a username/password combo.

The quick-start guide

http://www.voip-info.org/wiki-Asterisk+quickstart

I got lost between creating a sip account and the demo.

Just need some help getting off the ground w/ my 7940
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Re: [Asterisk-Users] CAPI problems when loading chan_capi.so

[Martin List-Petersen]

Hi Martin,

Have you checked the rights of your /dev/capi20* interfaces ? 
Do you run asterisk as a user or root ?

Either capi is not installed correctly (check with capiinfo) or you have not
given the user asterisk is using rights to access the capi devices.

/Martin

Citat Martin Mielke <[EMAIL PROTECTED]>:

> Hi all,
> 
> I compiled/installed chan_capi.so without problems. When I launch 
> Asterisk, I get the following error:
> 
> ---
> 
>  [chan_capi.so] => (Common ISDN API for Asterisk)
>   == Parsing '/etc/asterisk/capi.conf': Found
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: 
> ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: 
> ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2675 load_module: CAPI not 
> installed!
> Mar 30 19:47:52 WARNING[16384]: loader.c:312 ast_load_resource: 
> chan_capi.so: load_module failed, returning -1
> Mar 30 19:47:52 WARNING[16384]: chan_capi.c:2762 unload_module: Unable 
> to unregister from CAPI!
>   == Unregistered channel type 'CAPI'
> Mar 30 19:47:52 WARNING[16384]: loader.c:358 load_modules: Loading 
> module chan_capi.so failed!
> 
> ---
> 
> To test, I just modified the default MSN (50) to a real one (91xx 
> <-- faked here).
> 
> My capi.conf:
> ---
> pbx:/etc/asterisk # cat capi.conf
> ;
> ; CAPI config
> ;
> ;
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> 
> [interfaces]
> 
> msn=91xxx
> incomingmsn=*
> controller=1
> softdtmf=1
> accountcode=
> context=pstn
> ;echosquelch=1
> ;echocancel=yes
> ;echotail=64
> callgroup=1
> deflect=91xxx
> devices=2
> 
> 
> ;PointToPoint (55512-0)
> ;for outgoing calls use example 5551212
> ;and in dialplan you can use callerid like
> ;exten => _0XXX.,1,StripMSD,1
> ;exten => _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
> ;
> mode=immediate
> ;isdnmode=ptp
> ;msn=55512
> ;controller=2
> ;devices=30
> 
> ---
> 
> 
> The messege "CAPI not installed" is weird because CAPI *is* installed:
> ---
> pbx:~ # rpm -qa | grep capi
> avmfritzcapi-1.0-194
> capisuite-0.4.3-52
> capi4linux-2003.9.17-7
> ---
> 
> In this sense: do I need any other special package?
> 
> 
> TIA,
> Martin
> 
> 
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> 


-- 
Training is everything.  The peach was once a bitter almond; cauliflower is
nothing but cabbage with a college education.
-- Mark Twain, "Pudd'nhead Wilson's Calendar"

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[Asterisk-Users] H323 in Asterisk

[Terence Parker]

I have posted before but didn't get any replies so i'll ask again in a 
more simple way :

Does H323 work on asterisk out of the box? I notice there is already a 
channels/chan_h323.c file, but creating an h323.conf file I can't seem 
to get H323 working.

Do I have to compile an additional package first or something?

I tried the asterisk-oh323 thing, but can't get it to compile.

Terence

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Re: [Asterisk-Users] Caller entered digits ignored during wait....

[willy]

That's what the 'silence' files were invented for.
See loligo.com (forgot the exact reference, but do a wiki
for J Todd's sound files).
Yes, it's a hack, but it works.
Cheers,
Willy
- Original Message Follows -
> Greetings,
> 
>  
> 
> Below is part of the contents of my extensions.conf file. 
> 
>  
> 
> exten => s,1,Wait,1   ; Wait a
> second before answering.
> 
> exten => s,2,Answer
> 
> exten => s,3,ResponseTimeout,10  ; Set
> the amount of time the user
> 
>   
>  ; has to make a selection.
> 
> exten => s,4,DigitTimeout,5; Set the
> amount of time user has
> 
>   
>  ; between each number entry when
> 
>   
>  ; dialing an extension.
> 
> exten => s,5,Background(welcome)
> 
> exten => s,6,Background(parties)
> 
> exten => s,7,Wait(10)
> 
> exten => s,8,Background(parties)
> 
> exten => s,9,Wait(10)
> 
> exten => s,10,Background(vm-goodbye)
> 
> exten => s,11,Hangup
> 
>  
> 
> I can make a menu selection as long as Background is
> running however during Wait(10) DTMF digits are ignored.
> How can I wait for a response and register the response at
> the same time? I supposed I could create a sound file of
> 10 second duration and play this but that seems kinda like
> a hack to me.
> 
>  
> 
> Gene Kochanowky
> 
> 

Willy Wouters
ypOne Publishing

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RE: [Asterisk-Users] Caller entered digits ignored during wait....

[Gene Kochanowsky]

Title: Message








How would you use the t extension to
accomplish this?

 

Gene

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
Sent: Tuesday, March 30, 2004 9:03
PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Caller
entered digits ignored during wait



 



You could use the t extension to
accomplish this.  But if you're happy with your way...  :-)






Sean





-Original Message-
From: Gene Kochanowsky
[mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 30, 2004 8:53
PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Caller entered digits ignored during wait

I figured it out… This works  -

 

exten => s,1,Wait,1  
; Wait a second before answering.

exten => s,2,Answer

exten =>
s,3,SetVar,"loopCnt=0" 

exten => s,4,Background(welcome)

exten =>
s,5,SetVar,"loopCnt=$[${loopCnt} + 1]"

exten => s,6,gotoif,"$[${loopCnt}
>= 3]?s|7:s|9"

exten => s,7,Background(vm-goodbye)

exten => s,8,Hangup

exten => s,9,Background(parties)

exten => s,10,ResponseTimeout(10)

exten => t,1,Goto,s|5

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gene Kochanowsky
Sent: Tuesday, March 30, 2004 7:46
PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Caller
entered digits ignored during wait



 

Greetings,

 

Below is part of the contents of my extensions.conf file. 

 

exten => s,1,Wait,1  
; Wait a second before answering.

exten => s,2,Answer

exten =>
s,3,ResponseTimeout,10 
; Set the amount of time the user

   
; has to make a selection.

exten =>
s,4,DigitTimeout,5   
; Set the amount of time user has

   
; between each number entry when

   
; dialing an extension.

exten => s,5,Background(welcome)

exten => s,6,Background(parties)

exten => s,7,Wait(10)

exten => s,8,Background(parties)

exten => s,9,Wait(10)

exten => s,10,Background(vm-goodbye)

exten => s,11,Hangup

 

I can make a menu selection as long as Background is running
however during Wait(10) DTMF digits are ignored. How can I wait for a response
and register the response at the same time? I supposed I could create a sound
file of 10 second duration and play this but that seems kinda like a hack to
me.

 

Gene Kochanowky

RE: [Asterisk-Users] Caller entered digits ignored during wait....

[Sean Cheesman]

Title: Message



You 
could use the t extension to accomplish this.  But if you're happy with 
your way...  :-)
Sean

  
  -Original Message-From: Gene Kochanowsky 
  [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 30, 2004 
  8:53 PMTo: [EMAIL PROTECTED]Subject: RE: 
  [Asterisk-Users] Caller entered digits ignored during 
  wait
  
  I figured it out… 
  This works  -
   
  exten => 
  s,1,Wait,1   
  ; Wait a second before answering.
  exten => 
  s,2,Answer
  exten => 
  s,3,SetVar,"loopCnt=0" 
  exten => 
  s,4,Background(welcome)
  exten => 
  s,5,SetVar,"loopCnt=$[${loopCnt} + 1]"
  exten => 
  s,6,gotoif,"$[${loopCnt} >= 3]?s|7:s|9"
  exten => 
  s,7,Background(vm-goodbye)
  exten => 
  s,8,Hangup
  exten => 
  s,9,Background(parties)
  exten => 
  s,10,ResponseTimeout(10)
  exten => 
  t,1,Goto,s|5
   
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gene KochanowskySent: Tuesday, March 30, 2004 7:46 
  PMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Caller entered 
  digits ignored during wait
   
  Greetings,
   
  Below is part of the contents of 
  my extensions.conf file. 
   
  exten => 
  s,1,Wait,1   
  ; Wait a second before answering.
  exten => 
  s,2,Answer
  exten => 
  s,3,ResponseTimeout,10  
  ; Set the amount of time the user
      
  ; has to make a selection.
  exten => 
  s,4,DigitTimeout,5    
  ; Set the amount of time user has
      
  ; between each number entry when
      
  ; dialing an extension.
  exten => 
  s,5,Background(welcome)
  exten => 
  s,6,Background(parties)
  exten => 
  s,7,Wait(10)
  exten => 
  s,8,Background(parties)
  exten => 
  s,9,Wait(10)
  exten => 
  s,10,Background(vm-goodbye)
  exten => 
  s,11,Hangup
   
  I can make a menu selection as 
  long as Background is running however during Wait(10) DTMF digits are ignored. 
  How can I wait for a response and register the response at the same time? I 
  supposed I could create a sound file of 10 second duration and play this but 
  that seems kinda like a hack to me.
   
  Gene 
  Kochanowky

RE: [Asterisk-Users] Caller entered digits ignored during wait....

[Gene Kochanowsky]

I figured it out… This works  -

 

exten => s,1,Wait,1   ;
Wait a second before answering.

exten => s,2,Answer

exten =>
s,3,SetVar,"loopCnt=0" 

exten => s,4,Background(welcome)

exten =>
s,5,SetVar,"loopCnt=$[${loopCnt} + 1]"

exten => s,6,gotoif,"$[${loopCnt}
>= 3]?s|7:s|9"

exten => s,7,Background(vm-goodbye)

exten => s,8,Hangup

exten => s,9,Background(parties)

exten => s,10,ResponseTimeout(10)

exten => t,1,Goto,s|5

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gene Kochanowsky
Sent: Tuesday, March 30, 2004 7:46
PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Caller
entered digits ignored during wait



 

Greetings,

 

Below is part of the contents of my extensions.conf file. 

 

exten =>
s,1,Wait,1  
; Wait a second before answering.

exten => s,2,Answer

exten =>
s,3,ResponseTimeout,10 
; Set the amount of time the user

   
; has to make a selection.

exten =>
s,4,DigitTimeout,5   
; Set the amount of time user has

   
; between each number entry when

   
; dialing an extension.

exten => s,5,Background(welcome)

exten => s,6,Background(parties)

exten => s,7,Wait(10)

exten => s,8,Background(parties)

exten => s,9,Wait(10)

exten => s,10,Background(vm-goodbye)

exten => s,11,Hangup

 

I can make a menu selection as long as Background is running
however during Wait(10) DTMF digits are ignored. How can I wait for a response
and register the response at the same time? I supposed I could create a sound
file of 10 second duration and play this but that seems kinda like a hack to
me.

 

Gene Kochanowky

Re: [Asterisk-Users] Cisco 7960 tftp question

[Roger]

Oliver Kaven wrote:

Hello, and thank you for your time answering my question.

I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to 
upload
SIP configurations via *.cnf file from my tftp server, do I need to
include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root
directory to be uploaded every time the phone reboots?

You might try to put phone specific information in

SIP.cnf

Upon boot the phone checks for 

OS79XX.txt
SIPDefault.cnf
SIP.cnf
If you don't have 

OS79XX.txt the phone will boot as normal.

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Re: [Asterisk-Users] (no subject)

[Eric Wieling]

Search the archives.

On Tue, 2004-03-30 at 19:00, Peter Mitchell wrote:
> Has anyone had any luck using a 7910 with SIP image.  
> 
>  
> 
> Some information I found says 7910 is skinny only, other info suggests
> the 7910 may take the 7960 sip image.
> 
>  
> 
> Can anyone offer their experience ?
> 
>  
> 
> Cheers
> 
> Peter
> 
>  
-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

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[Asterisk-Users] (no subject)

[Peter Mitchell]

Has
anyone had any luck using a 7910 with SIP image.  

 

Some
information I found says 7910 is skinny only, other info suggests the 7910 may
take the 7960 sip image.

 

Can
anyone offer their experience ?

 

Cheers

Peter

 

[Asterisk-Users] Manager Interface "Action: Originate" changed

[Tony Wasson]

I have recently noticed that the "Action: Originate" options in asterisk 
   1.0 CVS has changed sometime between 2/23 and 3/18.

I have a 2/23/04 CVS installation (cvs checkout -r v1-0_stable asterisk 
) that allows me to make calls like this using the Manager Interface on 
port 5038.

  action: login
  login: admin
  secret: mypass
  action: originate
  exten: 200
  context: stations
  channel: SIP/agent007
I have a 3/18/04 CVS installation that does NOT work the same way. 
Entering the same information in spits out

  Response: Error
  Message: Originate with 'Exten' requires 'Context' and 'Priority'
So I've tried adding a priority of 1, like this:

  action: originate
  exten: 200
  context: stations
  channel: SIP/agent007
  priority: 1
I simply get:

  Response: Error
  Message: Originate failed
Obviously, something in the Manager code has changed. With the newer 
code I am unable to originate calls. Can anyone shed additional light on 
how to originate calls under the new 1.0 style Manager Interface?

Tony Wasson
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[Asterisk-Users] Caller entered digits ignored during wait....

[Gene Kochanowsky]

Greetings,

 

Below is part of the contents of my extensions.conf file. 

 

exten => s,1,Wait,1   ; Wait
a second before answering.

exten => s,2,Answer

exten => s,3,ResponseTimeout,10  ;
Set the amount of time the user

    ;
has to make a selection.

exten => s,4,DigitTimeout,5    ; Set the
amount of time user has

    ;
between each number entry when

    ;
dialing an extension.

exten => s,5,Background(welcome)

exten => s,6,Background(parties)

exten => s,7,Wait(10)

exten => s,8,Background(parties)

exten => s,9,Wait(10)

exten => s,10,Background(vm-goodbye)

exten => s,11,Hangup

 

I can make a menu selection as long as Background is running
however during Wait(10) DTMF digits are ignored. How can I wait for a response
and register the response at the same time? I supposed I could create a sound
file of 10 second duration and play this but that seems kinda like a hack to
me.

 

Gene Kochanowky

RE: [Asterisk-Users] SoftFAX/spandsp - txfax

[Alex Zarubin]

Title: RE: [Asterisk-Users] SoftFAX/spandsp - txfax





Hi Steve and all,


1. Faxing from asterisk back to the same asterisk (from one Zap channel to another)
    doesn't work for us. Txfax called with the 'caller' parameter issues CED, while the
    receiving side needs CNG in order to switch to fax extension with rxfax.


2. This is probably the reason why J2 and our UC don't recognize incoming fax.


Thank you.


Alex Zarubin
Webley Systems


 

[Asterisk-Users] Carrier Access CMG/FXS MGCP to Asterisk, Works Fine

[JR Richardson]

FYI,

 

Follow the Quick Start Guide from Carrier Access to setup
the CMG (Customer Media Gateway) Router card.  Follow the Asterisk
mgcp.conf wiki page setup.  The only issue I had was with the CAC CMG
card, it defaults to strict policy message exchange and dial-tone will not come
across when you go off-hook on a FXS port.  To resolve this, get back into
the CAC CMG configuration, goto:

-profile directory – router card setup – MGCP Configure
– MGCP Interoperability Settings – (set) Parse Mode: Lenient

 

Either Carrier Access or Asterisk is not completely
compatible with the RFC, but this seems to get the two talking together.

 

Hope this helps.

 

JR

RE: [Asterisk-Users] Exception flag set - snom200

[Ernest W. Lessenger]

What version of asterisk are you using, and what version of 
the SNOM firmware?
 
--Ernest

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  jcSent: Tuesday, March 30, 2004 10:20 AMTo: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] Exception 
  flag set - snom200
  
  
  Sorry I forgot the 
  subject in the last post.
   
  When my snom200 
  receives an inbound SIP external sip call, it somehow rejects the call and 
  with a busy tone.  The debug shows the following error:
   
  channel.c:1142 
  ast_read: Exception flag set on 'SIP/sipphone-7796', but no exception 
  handler
   
   
  what does this mean 
  and how can I debug it further??
   
  Thanks 
  
  JC
   

Re: [Asterisk-Users] Cisco 7960 tftp question

[kwijibo]

Depends if what you have in OS79XX.TXT is different then
what is running on your phone.  If it isn't it won't bother to touch
the image files, if it is then it will try to load whatever image
you have specified in OS79XX.TXT.
So far I have been unable to tell the phones to boot out of anything
but the tftp root directory.  They seem to ignore the filename directive
from dhcp.  Once the phones get the SIPDefault.cnf file from the root
directory you can change the directory where the phones get their
specific configurations/ringers/dialplans.
Steve

Oliver Kaven wrote:

Hello, and thank you for your time answering my question.

I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to upload
SIP configurations via *.cnf file from my tftp server, do I need to
include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root
directory to be uploaded every time the phone reboots?
Thanks again for your help.

Oliver



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[Asterisk-Users] microsoft messenger with sip debug

[Shawn]

Sip read:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:12250
From: ;tag=1e263406-3e84-45fb-a971-6f08bf684275
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Contact: ;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"
User-Agent: Windows RTC/1.0
Expires: 1200
Event: registration
Allow-Events: presence
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.100 : 12250 (non-NAT)
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.100:12250
From: ;tag=1e263406-3e84-45fb-a971-6f08bf684275
To: ;tag=as3f2ea3e5
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


 to 192.168.1.100:12250
Mar 30 18:35:28 NOTICE[1125329600]: chan_sip.c:5609 handle_request:
Registration from '' failed for '192.168.1.100'


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[Asterisk-Users] Sipcall.co.uk & [*]

[Matt]

Hello all.

Has anyone managed to get SIPCALL.co.uk's service working with the [*] box?

I've managed to register with other SIP providers but not SIPcall.

The debug just show's [*] attempting to register.
But receiving a 401 error everytime.

Cheers

Matt

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[Asterisk-Users] error with microsoft messenger

[Shawn]

NOTICE[1125329600]: chan_sip.c:5609 handle_request: Registration from
'' failed for '192.168.1.100'


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RE: [Asterisk-Users] RE: mysql or postgresql?

[Sean Cheesman]

have you installed the mysql-devel package?  

-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 30, 2004 6:11 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: mysql or postgresql?


thanks for awnser, I've already download from CVS the asterisk-addons,
now get this error when compiling

# make install
./mkdep -fPIC -I../asterisk `ls *.c`
cdr_addon_mysql.c:33:19: mysql.h: No such file or directory
cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory for x in  ;
do install -m 755 $x /usr/lib/asterisk/modules ; done
/bin/sh: -c: line 1: syntax error near unexpected token `;'
/bin/sh: -c: line 1: `for x in  ; do install -m 755 $x
/usr/lib/asterisk/modules ; done'
make: *** [install] Error 2

Any idea?

Thanks again.

(o_
//\
V_/_
"hackers build things, crackers break them." http://kokey.gluch.org.mx

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[Asterisk-Users] RE: mysql or postgresql?

[Jorge de J. Ramirez S.]

thanks for awnser, I've already download from CVS the asterisk-addons, now
get this error when compiling

# make install
./mkdep -fPIC -I../asterisk `ls *.c`
cdr_addon_mysql.c:33:19: mysql.h: No such file or directory
cdr_addon_mysql.c:34:20: errmsg.h: No such file or directory
for x in  ; do install -m 755 $x /usr/lib/asterisk/modules ; done
/bin/sh: -c: line 1: syntax error near unexpected token `;'
/bin/sh: -c: line 1: `for x in  ; do install -m 755 $x
/usr/lib/asterisk/modules ; done'
make: *** [install] Error 2

Any idea?

Thanks again.

(o_
//\
V_/_
"hackers build things, crackers break them."
http://kokey.gluch.org.mx

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Re: [Asterisk-Users] Asterisk server lockup

[Steven Critchfield]

On Tue, 2004-03-30 at 07:34, Gary Franczyk wrote:
> Hello,
> 
> We are trying to deploy a new asterisk server with a Wildcard T400P (quad
> T1) card.  It uses a custom voice recording app written in the perl AGI.
> 
> Now that the machine has been in production, it seems to lock up within 24
> hours of reboot!  When it locks, we can ping the machine, but we cannot log
> in using telnet or ssh.  Asterisk stops answering the phone and our Big
> Brother monitoring scripts stop sending data.  Nothing is shown in the
> messages log.  The script we are using seems to be ok, since it exits after
> every
> 
> My first guess is that it could be running out of memory (since I can still
> ping it, the kernel must be working to some extent).   I don't have a
> console attached to this machine yet, so I can't tell you what is displayed
> on the screen yet.  (I will get one soon) Has anyone seen any asterisk lock
> ups like this before?

pinging a machine doesn't mean any user level applications are even
running. Look around and you should be able to find a router project
that you set your rules up then halt the kernel. At that point the
kernel continues to pass traffic, but there is nothing on the machine
that can be run until it is rebooted.

I also have seen that on newer checkouts though not based a specific
amount of time or number of calls. We have had great experience out of
older asterisk deployments. So much so that they stagnated due to no
need to upgrade to anything newer.   
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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[Asterisk-Users] Asterisk Security Audit?

[Jim Rosenberg]

Has Asterisk ever been audited for common security holes, such as buffer
overruns?

A quick grep through the source for routines that should never be used,
like strcpy, strcat, etc., reveals a lot of it. I fear I fear.

Has anyone flung pathology at IAX2 to see if it stands up to malformed
packets? (This is always an issue when you have a protocol that only a
small number of programs use ...)

I hope I'm wrong, but I have a very queasy feeling ...

[We already know that H.323 is not being looked after, security-wise ...]
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[Asterisk-Users] G.729 and h323.conf

[Jim Rosenberg]

What should my allow= line look like in h323.conf for G.729?

I've tried

allow=G729A

but this doesn't seem to be right. These "codec indentifiers" sure are
mysterious. Take g711alaw. To allow it you seem to have to use allow=ALAW.
Even though "ALAW" does not show anywhere as an identifier when you say
show codecs.
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Re: [Asterisk-Users] Asterisk + GrandStream SIP phones

[pesb]

Hi,
Thanks for the help. You were correct. There was some data missing in the 
extension.conf file
I was able to call one SIP phone from the other. I was even able to call an 
H323 IP phone registered to the gnugk GK (It has Asterisk registered to him 
as a GW).
But, I have another problem rigth now.
All the RTP Data Flow is passing through the Asterisk Proxy, which is a bad 
thing if I want to have many SIP phones in my system.
How can I configure the SIP phone in order to make all RTP data flow directly 
from one SIP phone to the other? 
And, how can I configure it in order to make all RTP data flow directly from 
one SIP phone to the H323 IP phone (the one registered to my gnugk GK)?
I would also like to be able to make calls from a SIP phone to the other SIP 
phone, but instead of having the ASTERISK PBX authorizing the calls, it would 
be the H323 GK the one that would authorize calls. How can I do this?

Thanks again

On Monday 29 March 2004 15:58, David J Carter wrote:
> Try this small extensions.conf
>
> Don't think I have missed owt.
>
> My config files are here, you just need to add your own extension numbers.
>
> http://www.codepipe.com/id25.htm
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of pesb
> Sent: 29 March 2004 19:26
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones
>
>
> -This is my 'sip.conf' file:
>
> ;*
> ;
> ; SIP Configuration for Asterisk
> ;
> [general]
> port = 5060 ; Port to bind to
> bindaddr = 0.0.0.0  ; Address to bind to
> context = default   ; Default for incoming calls
> tos=184
> maxexpirey=3600 ; Max length of incoming registration we allow
> defaultexpirey=120  ; Default length of incoming/outoing
> registration
> disallow=all; Disallow all codecs
> allow=ulaw  ; Allow codecs in order of preference
> allow=alaw
>
>
> [1004]
> type=friend
> username=1004
> secret=
> reinvite=no
> canreinvite=no
> host=dynamic
> dtmfmode=inband
> mailbox=1004
> nat=1
> disallow=all
> allow=ulaw
> allow=alaw
>
> [1005]
> type=friend
> username=1005
> secret=
> reinvite=no
> canreinvite=no
> host=dynamic
> dtmfmode=inband
> mailbox=1005
> nat=1
> disallow=all
> allow=ulaw
> allow=alaw
>
> ;***
>
>
> -And this is the basic seting of my two GrandStream SIP phones:
>
> ***[1005]
> IP Address:192.168.0.105
> Subnet Mask:255.255.255.0
> SIP Server: 192.168.0.103
> Outbound Proxy:
> SIP User ID:1005
> Authenticate ID:1005
> Authenticate Password:123
> Name:1005
>
> Preferred Vocoder:
> choice 1: PCMU
> choice 2: PCMA
> choice 3: G723
> choice 4: G729
> choice 5: G726-32
> choice 6: G728
>
> G723 rate:  6.3kbps
> Silence Suppression:No
> Send DTMF:in-audio
>
> ***[1004]
> IP Address:192.168.0.104
> Subnet Mask:255.255.255.0
> SIP Server: 192.168.0.103
> Outbound Proxy:
> SIP User ID:1004
> Authenticate ID:1004
> Authenticate Password:123
> Name:1004
>
> Preferred Vocoder:
> choice 1: PCMU
> choice 2: PCMA
> choice 3: G723
> choice 4: G729
> choice 5: G726-32
> choice 6: G728
>
> G723 rate:  6.3kbps
> Silence Suppression:No
> Send DTMF:in-audio
>
> **
>
> I have 2 SIP GrandStream phones, both phones are correctly registered to
> the Asterisk server. But, when I try to make a call from registered phone
> '1005' to registered phone '1004', dialing 1004, Asterisk responds with the
> 'Status:
> 404 Not Found' message.
> How do I have to dial? What else do I need to set?
> Find attached my traffic captured on ethereal.

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[Asterisk-Users] Cisco 7960 tftp question

[Oliver Kaven]

Hello, and thank you for your time answering my question.

I have a Cisco 7960 running SIP firmware POS3-06-X-XX. If I want to upload
SIP configurations via *.cnf file from my tftp server, do I need to
include the OS79XX.TXT and P0S3-06-X-XX.bin to the tftp server root
directory to be uploaded every time the phone reboots?

Thanks again for your help.

Oliver



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RE: [Asterisk-Users] asterisk @ home ?

[Dave Weis]

On Tue, 30 Mar 2004, calvis wrote:
> What are reciprocal comp minutes?  Please explain.

In some states, the competitive and incumbent phone carriers bill each 
other for calls that they terminate from the other. If a Qwest customer 
calls a Dave's Phone Company customer, I will get a small amount of money 
per minute for completing the call. Some states are bill and keep, meaning 
no one gets paid. ISP's loved it when they could set up as a CLEC and put 
their modem banks on their own switch, because they are terminating all of 
these calls and getting paid per minute from the ILEC to do so. The FCC 
let the ILEC's not pay reciprocal comp for those calls.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
> Sent: Tuesday, March 30, 2004 1:12 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] asterisk @ home ?
> 
> 
> Must be a CLEC trying to build up reciprocal comp minutes ;-)
> 
> 

-- 
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent 
  and sudden usurpations."- James Madison

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RE: [Asterisk-Users] asterisk @ home ?

[calvis]

What are reciprocal comp minutes?  Please explain.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis
Sent: Tuesday, March 30, 2004 1:12 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] asterisk @ home ?


Must be a CLEC trying to build up reciprocal comp minutes ;-)

-- 
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent 
  and sudden usurpations."- James Madison

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RE: [Asterisk-Users] asterisk @ home ?

[Dave Weis]

On Tue, 30 Mar 2004, Kevin Walsh wrote:
> > I'm curious as to what a "Washington State free phone number" is?  I live
> > in Washington State(Spokane) and we get our PSTN service from Qwest which
> > is certainly not free! 
> > 
> The poster was probably referring to IPKall (http://www.ipkall.com/).
> IPKall will assign you a Washington State (USA) phone number for free
> and have incoming calls routed to the SIP address of your choice.  I
> have one and am using it to route calls to my office in England.

Must be a CLEC trying to build up reciprocal comp minutes ;-)

-- 
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent 
  and sudden usurpations."- James Madison

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RE: [Asterisk-Users] asterisk @ home ?

[Kevin Walsh]

> I'm curious as to what a "Washington State free phone number" is?  I live
> in Washington State(Spokane) and we get our PSTN service from Qwest which
> is certainly not free! 
> 
The poster was probably referring to IPKall (http://www.ipkall.com/).

IPKall will assign you a Washington State (USA) phone number for free
and have incoming calls routed to the SIP address of your choice.  I
have one and am using it to route calls to my office in England.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
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_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] G726 not working ?

[Bill Hamel]

Hi,

I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of
this morning 3/30/04 of asterisk, zap and libpri.

The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced".

When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I
can see:

[format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data)
  == Registered file format g726-40, extension(s) g726-40
  == Registered file format g726-32, extension(s) g726-32
  == Registered file format g726-24, extension(s) g726-24
  == Registered file format g726-16, extension(s) g726-16

I 'Ass'ume this indicates that g726 is installed..

So in my sip.conf I put many variations of what I thought should go in there,
finally includeing (to no avail):

disallow=all
allow=g726-40
allow=g726-32
allow=g726-24
allow=g726-16
allow=g726
allow=ima-adpcm

(Also tried G.726-xx etc... ) And none seem to work because when I dial out I
get

Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible
codecs!

I must not be putting the correct "allow=" value in sip.conf or possibly missing
something. 

Can anyone point me in the right direction ?

Thanks
-bh


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Re: [Asterisk-Users] Watchguard Firebox 1000 and Asterisk

[Bill Hamel]

The firebox has the UDP timeout set pretty low by default, this is a good thing
to help prevent DOS attacks, but isn't a really good thing for a SIP device.

There is no option in the GUI to set this.

However you can go into the config file itself and modify the following:

options.masquerade.udp.timeout: 30
options.services.dynamic.timeout.udp: 25

Set them higher than your "register timeout" on your 7960.

Then save the config file and upload to the firebox.

HTH
-bh




Quoting Glenn Dalgliesh <[EMAIL PROTECTED]>:

> Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I
> have asterisk on public side and phones on the private side. I am able to
> get the phones to register and make outbound calls but the inbound calls are
> intermittent. I have NAT enable in asterisk and on the Cisco 7960.
> 
> Any insight would be appreciated.
> 
> Thanks
> 
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RE: [Asterisk-Users] FreeBSD

[Joe Phillips]

> On Mon, 2004-03-29 at 20:25, Steven M. Sokol wrote:
> > Not currently.  There is a bounty for the development of working Wildcard
> > drivers for Free/Net/Open BSD.  Care to write them?

On Mon, 2004-03-29 at 20:33, James Moran wrote: 
> Dam wish I was that good to do that.

You can pitch into the bounty to sweeten the pot for someone who is good
enough. 

I know some guys who are capable and I've forwarded on the bounty notice
but I'm guessing it's not high enough to make it worth their while
. 

-joe 
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RE: [Asterisk-Users] problem with configuration.

[Sean Cheesman]

The answer is in the error  use FXS signalling.  replace fxo_ks with
fxs_ks.  

Sean

-Original Message-
From: vozip [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 30, 2004 2:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problem with configuration.
Importance: High


Hi, 

When i run the asterisk with my FXO x100p and configure:
vi /etc/zaptel.conf

fxoks=1
loadzone=us
defaultzone=us

# vi /etc/asterisk/zapata.conf

[channels]
 
busydetect=1
busycount=7
 
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
 
usecallerid=yes
 
echocancel=yes
echocancelwhenbridged=yes
 
rxgain=0.0
txgain=0.0
 
group=1
pickupgroup=1-4
 
immediate=no
 
context=bell
 
callerid=asreceived
channel=1
 
context=home
 
group=1
signalling=fxo_ks
mailbox=2468
callerid="Phone 1" <2468>
channel=1


AND RUN ZTCFG.


[EMAIL PROTECTED]:/etc/asterisk# ztcfg -vv

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?



ANY IDEAS.!

CHEERS.!

VOZIP


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RE: [Asterisk-Users] SoftFAX/spandsp - release 0.0.1i - txfax fin dings

[Alex Zarubin]

Title: RE: [Asterisk-Users] SoftFAX/spandsp - release 0.0.1i - txfax findings





Hi,


We have no problems sending to HP and Panasonic fax machines in the office.
We do have problems when we try to send faxes to services supporting
fax, i.e. J2 or our UC platform. The receiving side doesn't recognize fax.


To send a fax we drop into /var/spool/asterisk/outgoing:
Channel: Zap/g1/
MaxRetries: 0
WaitTime: 20
Context: webley_txfax
Extension: txfax_ext
Priority: 1
SetVar: TXFAX_NAME=


and extensions.conf contains:
[webley_txfax]
exten => txfax_ext,1,txfax(${TXFAX_NAME}|caller)
exten => txfax_ext,2,Hangup


Two questions:
1. Sounds like txfax sends just one CNG tone. Can we have a parameter making
    txfax issue periodic CNGs until it gets CED back? This might resolve the
    problem with the receiver missing the first (and only) tone and therefore
    working as if this is not a fax?
2. For Zap channels - don't we need to take care of 'dataquality', 'ignoredtmf'
    and/or 'faxhandled'? In several cases J2 would send CED, so asterisk
    switched to a fax extension creating a complete mess.


Thank you.


Alex Zarubin
Webley Systems

[Asterisk-Users] Re: What failed here?

[John Chambers]

James Golovich wrote:

The mkdep simply builds .depend files in each directory of the source
tree.  make uses this to determine what needs to be rebuilt if one of the
header files has changed.  There is nothing to worry about at all with
that part.
OK; I'll ignore it.  It can be confusing when a make produces a lot of
things that look like error messages but aren't. ;-)
It looks like either your CC line in the Makefile has been changed, or
perhaps your overriding it somehow.  Make sure your using unmodified code.
The code was just downloaded from cvs, the current "stable" version.
I didn't have a CC in the environment.  I added CC=/usr/bin/gcc (which
is the same as /usr/bin/cc), but that had no effect.
/var/spool/asterisk is a directory not an executable, and make should be
calling gcc instead of /var/spool/asterisk
Yeah; I sorta noticed that, and the multiple spaces in the command.
It looks like a bunch of Makefile macros came up undefined.  Maybe
I'll have to dig into the Makefile and see what it's trying to do.
I was sorta hoping that, since it's the current "stable" release,
someone else had seen the same symptoms and could say "Oh, yeah;
you just "
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Re: [Asterisk-Users] problem with configuration.

[John Fraizer]

vozip wrote:
group=1
signalling=fxo_ks
mailbox=2468
callerid="Phone 1" <2468>
channel=1
ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?


ANY IDEAS.!

CHEERS.!

VOZIP

The error message is fairly descriptive.  It's telling you what the problem 
is.  You need to configure it with FXS signaling.  You have it configured to 
use FXO signaling.

John

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Re: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR) (update)

[Jeb Campbell]

On Mar 30, 2004, at 1:19 PM, James Golovich wrote:

Another possibility is that the interface is looped back so the NET
packets it sees are the ones asterisk sends out and gets looped back
James
How can I see or test this?

zttool has a loop feature, but I have no idea how to use it.

Jeb Campbell
[EMAIL PROTECTED]
Cell: 865-385-1437
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[Asterisk-Users] Problems with stuck PRI channel

[Korey Chapman]

I haven't seen anything about this is the archives, so here we go. 
Sorry, its a long one.

My setup:
Dual Xeon 2.4 GHz.
One TE410P card.
Span 1 populated with a PRI.
Span's 2-4 empty.

zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us


I'm currently having a problem where one of our PRI channels seems to
keep a call even after it is hung up.  We have only seen it happen with
one channel at a time, but its not the same channel every time.  The
channel continues to receive calls from the telco (we are in Qwest land
here) and Asterisk is dumping them.

Here is the message from asterisk when a new call comes in on the busy
channel:
"Ring requested on channel 13 already in use on span 1.  Hanging up
owner."

The state of the channel stays the same after this error occurs.

The channel data for that zap channel shows the channel still connected
to a SIP phone.  The phone is not in use and does not have anything on
hold.

I have attempted to kill the channel with "zap destroy channel", but it
does not appear to work.  Doing some operations on the console appears
to cause asterisk to die.  When asterisk returns to normal operations
the channel is clear (of course all of our calls are gone too).

I looked at zttool, it doesn't show any IRQ misses or alarms.

This has been happening once a week, but suddenly has become once a day.
I'm at a loss.

We just did a full update of asterisk, libpri, zaptel, and zapata within
the last week.  The problem is still occurring after the update.


Here is a pri debug from during an incident, phone number obscured to
protect the innocent :)

< Protocol Discriminator: Q.931 (8)  len=35
< Call Ref: len= 2 (reference 5160/0x1428) (Originator)
< Message type: SETUP (5)
< Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech 
(0)
<  Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
<  Ext: 1  User information layer 1: u-Law (34)
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan:0
 Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 37928/0x9428) (Terminator)
> Message type: RELEASE COMPLETE (90)
> Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private 
> network serving the local user (1)
>  Ext: 1  Cause: Requested channel not available (44), class =Network 
> Congestion (2) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null


Any help is appreciated.

Sincerely,
Korey Chapman


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[Asterisk-Users] problem with configuration.

[vozip]

Hi, 

When i run the asterisk with my FXO x100p and configure:
vi /etc/zaptel.conf

fxoks=1
loadzone=us
defaultzone=us

# vi /etc/asterisk/zapata.conf

[channels]
 
busydetect=1
busycount=7
 
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
 
usecallerid=yes
 
echocancel=yes
echocancelwhenbridged=yes
 
rxgain=0.0
txgain=0.0
 
group=1
pickupgroup=1-4
 
immediate=no
 
context=bell
 
callerid=asreceived
channel=1
 
context=home
 
group=1
signalling=fxo_ks
mailbox=2468
callerid="Phone 1" <2468>
channel=1


AND RUN ZTCFG.


[EMAIL PROTECTED]:/etc/asterisk# ztcfg -vv

Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?



ANY IDEAS.!

CHEERS.!

VOZIP


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AW: [Asterisk-Users] CAPI problems when loading chan_capi.so

[Sascha Knific]

Hi Martin

>  [chan_capi.so] => (Common ISDN API for Asterisk)
>   == Parsing '/etc/asterisk/capi.conf': Found
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
> ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif:
> ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
> Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2675 load_module: CAPI not
> installed!

What is the output of "capiinfo"?


Sascha

---
Sascha Knific   K Systems & Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319 Starnberg, Germany
[EMAIL PROTECTED] http://www.k-sysdes.net




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RE: [Asterisk-Users] mysql or postgresql?

[Sean Cheesman]

it is not included with the asterisk distribution.  you must download it
separately.  asterisk_addons.

-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, March 30, 2004 2:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] mysql or postgresql?


Hi,
there are something that is using mysql instead postresql?

If I modify the modules.conf, and write load => cdr_mysql.so get this
error:

loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/cdr_mysql.so:
cannot open shared object file: No such file or directory loader.c:359
load_modules: Loading module cdr_mysql.so failed!

Ok, there is no cdr_mysql.so, anybody it's using * with mysql?

Thanks!

(o_
//\
V_/_
"hackers build things, crackers break them." http://kokey.gluch.org.mx

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RE: [Asterisk-Users] Queue feature

[ml]

> What I'm looking for is the ability to determine whether or not a queue
> has 
> any queue handlers (active agents), and if it does not, bypass sending
> the 
> caller to the queue and pass them on to a message or IVR system.
> 
> -Chris

http://bugs.digium.com/bug_view_page.php?bug_id=214

This is not exactly what you are looking for, but you can set a queue timeout.  Post a 
request to that bug for
the feature.

Kevin
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[Asterisk-Users] is Asterisk capable for SIP-H323 translation?

[Konstantin Kropivny]

Title: is Asterisk capable for SIP-H323 translation?






Hi


is Asterisk capable for SIP-H323 translation? Any manual how to do this?


Thank you


Konstantin 

Re: [Asterisk-Users] Cisco 7960 SIP Images

[Jon Lawrence]

On Tuesday 30 March 2004 19:01, Brian Cuthie wrote:

>  My beef with Cisco is that the software license doesn't travel with the
> device. Without the license you can't buy an upgrade even if you want to.
>
Indeed that bit is a complete joke. I can't think of anything that could be 
done about it though.

Jon

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[Asterisk-Users] mysql or postgresql?

[Jorge de J. Ramirez S.]

Hi,
there are something that is using mysql instead postresql?

If I modify the modules.conf, and write load => cdr_mysql.so get this error:

loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/cdr_mysql.so:
cannot open shared object file: No such file or directory
loader.c:359 load_modules: Loading module cdr_mysql.so failed!

Ok, there is no cdr_mysql.so, anybody it's using * with mysql?

Thanks!

(o_
//\
V_/_
"hackers build things, crackers break them."
http://kokey.gluch.org.mx

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Re: [Asterisk-Users] VON Update - Greetings from Infomercial Central

[Rainer Jochem]

>   This is not all bad.  It is good research into the strategies the
> big players are using.  It looks like the big money players (AT&T, Nortel,
> Siemens, Cisco, etc.) are really trying to push into VoIP in a big way.  The
> other big positive is the fact that people are actually, well, positive.
> The buzz and the money seem to be back in the VoIP market.  I just hope that
> the Asterisk community can take advantage of the change.

Cool. I can affirm this from  CeBIT this month where we were also
showing VoIP with */SER at the booth of our University.
Also there was almost every company at least doing something with VoIP
and there were several also big companies interested in OpenSource
VoIP.


> Asterisk Team Arrives
> 
>   Mark and Greg, plus Jeremy from NuFone and Don Witt from Cylogistics
> are here and manning the booth.  A good number of people wandered by to see
> the display and to ask questions about Asterisk.  Lots of interest!  I
> actually have seen an IAXy, so I can attest to the fact that they actually
> exist.  Rumors have it that they will be available in the near future.
> Similar rumors surround the FXO modules for the TDM4XX platform.

Where are the pics and the livecam??? :)



Have a nice journey there and greet Mark,

 Rainer


-- 
http://graphics.cs.uni-sb.de/VoIP/

pgp0.pgp
Description: PGP signature

[Asterisk-Users] Exception flag set - snom200

[jc]

Sorry I forgot the subject in the last
post.

 

When my snom200 receives an inbound SIP
external sip call, it somehow rejects the call and with a busy tone.  The
debug shows the following error:

 

channel.c:1142 ast_read: Exception flag
set on 'SIP/sipphone-7796', but no exception handler

 

 

what does this mean and how can I debug it
further??

 

Thanks 

JC

 

[Asterisk-Users] IAX2 trunk mode over satellite

[John Todd]

Today has been the day for satellite questions, apparently, so I'll 
proxy one out to the rest of the community...  I asked this 
tangentially a month or two ago, but I'll put it in a more blunt way:

If you have IAX2 trunking mode experience over satellite, please let 
us know your experiences with that protocol/transport combination.

I've got several people asking about IAX2 and trunk mode over 
satellite.  I have not experimented with IAX2 over satellite (though 
I have used IAX1 over satellite) and I'm wondering if anyone has 
direct experiences with IAX2's jitter buffer control over such 
long-latency connections.

I've had SIP working very well over satellite (despite what some 
people have found to the contrary on this list) and other than the 
lag there have been no issues that have come up on a 
reasonably-managed satellite segment.  However, the IP overhead 
really starts to cost significant amounts of pennies when you add it 
up on multiple SIP RTP sessions over the same link.  Plus, packet 
contention and buffering may (_may_) be an issue when pushing 
multiple simultaneous streams out the same transponder.

It would seem to me that IAX2 in trunk mode would be optimal for 
people on very expensive satellite bandwidth, as a G.729 9.6kbps 
channel starts to actually look like 9.6kbps instead of 24kbps. 
However, I have had mixed success with IAX2 in certain circumstances. 
Before I start to ask for favors and get satellite time for testing, 
I'd like to see if anyone else has performed this experiment.  If 
you'd wish to remain anonymous, please mail me directly and I'll 
appropriately trim identity information and re-distribute, or 
re-write as appropriate.

Other hints I have heard/used on VoIP over satellite:
  - use small transmit cell (packet) sizes on your satellite gear
  - turn off error correction (why use it for VoIP?)
  - turn off compression (G.729 is already compressed; you ARE using 
G.729, right?)
  - ensure minimal latency on the terrestrial portions of the call
  - tell your users to suck it up and deal with the half-second lag

JT
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Re: [Asterisk-Users] T100P T1 problem (Avaya -> asterisk IVR) (update)

[James Golovich]

On Mon, 29 Mar 2004, Eric Wieling wrote:

> Jeb Campbell wrote:
> 
> > Anyway, the only stuff off list was trying to debug the connection.
> > 1. With a crossover there is no sync (YELLOW and RED alarms)
> > 2. With standard cable I get a pri error that they think they are the 
> > NET, but we are the NET.
> > (This is asterisk 1.0 stable and the directions from voip-info)
> 
> If they think they are NET then make Asterisk CPE, if they think they 
> are CPE then make Asterisk NET.

Another possibility is that the interface is looped back so the NET
packets it sees are the ones asterisk sends out and gets looped back

James

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[Asterisk-Users] Is Wildcard TDM400P capable of sending DTMF callerid?

[Stig Andersson]

Hi,

Is Wildcard TDM400P capable of sending DTMF callerid?
Does asterisk support it?

I know X100P does not, but I have found no info as to TDM400P...

/Stig
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RE: [Asterisk-Users] Cisco 7960 SIP Images

[Dean Collins]

Yep, that would be my guess



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Tuesday, 30 March 2004 6:47 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images

Rich Adamson wrote:

> 
> Wanta take a guess what would happen if Cisco decide to really enforce
> the legal rules?
> 

I'll bite:

Their market share would plummet in all their markets, and then smaller,

more innovative companies would become more able to compete with them, 
and the overall marketplace would be vastly improved because of more 
participants and more choices?

B.
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Re: [Asterisk-Users] Re: What failed here?

[James Golovich]

On Tue, 30 Mar 2004, John Chambers wrote:

> Another worrying thing that I've noticed:  The stuff at the start
> of the make (that scrolls off the top too fast to read ;-) first does
> a mkdep, and then these messages appear:
> 
> cli.c:31:19: build.h: No such file or directory
> dlfcn.c:40:25: mach-o/dyld.h: No such file or directory
> dlfcn.c:41:26: mach-o/nlist.h: No such file or directory
> dlfcn.c:42:28: mach-o/getsect.h: No such file or directory
> 
> Sure enough, those files don't exist. Some time later, build.h does
> appear, when the Makefile runs make_build.h.  It seems a bit odd that
> the Makefile would attempt to use build.h before creating it.  This
> looks like a sign of something wrong, but I can't tell what. Any
> idea how to fix this?  Or is it actually a problem?

The mkdep simply builds .depend files in each directory of the source
tree.  make uses this to determine what needs to be rebuilt if one of the
header files has changed.  There is nothing to worry about at all with
that part.

It looks like either your CC line in the Makefile has been changed, or
perhaps your overriding it somehow.  Make sure your using unmodified code.

/var/spool/asterisk is a directory not an executable, and make should be
calling gcc instead of /var/spool/asterisk

James


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RE: [Asterisk-Users] Cisco 7960 SIP Images

[Brian Cuthie]

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jon Lawrence
> Sent: Tuesday, March 30, 2004 12:50 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images
...
> I have no problem with the idea of paying cisco for software 
> that they write.
> In fact I have no problem with with paying for software full 
> stop. But I'd love to have enough money to sue them if that 
> software proved to have security issues or proved to be not 
> fit for purpose - eg if a phone had a bug in its 
> implementation of SIP.
> If people/companies want to charge for software fine (after 
> all it takes time/money to develop) but they should be 
> willing to take the responsibility that goes with it. Most 
> companies don't - at least if you cantact cisco with a 
> problem then they'll do their best to fix it or at least come 
> up with a work-around, which is more than a certain other 
> companies do.
> 
> Jon
> 

I don't have a problem paying for updates, even if they include bug fixes. I
write software for a living, and it's an imperfect art. My beef with Cisco
is that the software license doesn't travel with the device. Without the
license you can't buy an upgrade even if you want to.

-brian 

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[Asterisk-Users] (no subject)

[jc]

When my snom200 receives an inbound SIP external sip call,
it somehow rejects the call and with a busy tone.  The debug shows the
following error:

 

channel.c:1142 ast_read: Exception flag set on
'SIP/sipphone-7796', but no exception handler

 

 

what does this mean and how can I debug it further??

 

Thanks 

JC

 

Re: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group

[Ryan Thrash]

Actually, ignore that... forgot to take the check the calendar pill 
this AM. Doh!

rt

On Mar 30, 2004, at 11:46 AM, Ryan Thrash wrote:

How did the launch meeting go?

rt
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[Asterisk-Users] Sacking calls to extension to voicemail

[Kurt Pasewaldt]

A SIP call comes into the * server on a number that I
want to immediately sack to vovoicemail. How would
this be achieved

Kurt

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[Asterisk-Users] CAPI problems when loading chan_capi.so

[Martin Mielke]

Hi all,

I compiled/installed chan_capi.so without problems. When I launch 
Asterisk, I get the following error:

---

[chan_capi.so] => (Common ISDN API for Asterisk)
 == Parsing '/etc/asterisk/capi.conf': Found
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: 
ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: 
ast_capi_pvt(91xx,*,pstn,0x2,2) (1,2,64) (0)(0.80/0.80) 0
Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2675 load_module: CAPI not 
installed!
Mar 30 19:47:52 WARNING[16384]: loader.c:312 ast_load_resource: 
chan_capi.so: load_module failed, returning -1
Mar 30 19:47:52 WARNING[16384]: chan_capi.c:2762 unload_module: Unable 
to unregister from CAPI!
 == Unregistered channel type 'CAPI'
Mar 30 19:47:52 WARNING[16384]: loader.c:358 load_modules: Loading 
module chan_capi.so failed!

---

To test, I just modified the default MSN (50) to a real one (91xx 
<-- faked here).

My capi.conf:
---
pbx:/etc/asterisk # cat capi.conf
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]

msn=91xxx
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=pstn
;echosquelch=1
;echocancel=yes
;echotail=64
callgroup=1
deflect=91xxx
devices=2
;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten => _0XXX.,1,StripMSD,1
;exten => _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;
mode=immediate
;isdnmode=ptp
;msn=55512
;controller=2
;devices=30
---

The messege "CAPI not installed" is weird because CAPI *is* installed:
---
pbx:~ # rpm -qa | grep capi
avmfritzcapi-1.0-194
capisuite-0.4.3-52
capi4linux-2003.9.17-7
---
In this sense: do I need any other special package?

TIA,
Martin
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[Asterisk-Users] VON Update - Greetings from Infomercial Central

[Steven M. Sokol]

The Hype Is Back

Ok.  Everybody who has ever been to a trade show knows that the
majority of what you hear is marketing hype.  That's to be expected.  But
usually you expect to see that on the exhibit floor or in companies'
hospitality rooms.  Unfortunately, the VON show seems to have decided to
extend that to the Keynote Addresses and Industry Perspective sessions.  By
and large it looks like I have paid 2500 USD to listen to a series of the
world's longest AT&T infommercials.  It's hard to think of AT&T as a driving
force in disruptive technology.

This is not all bad.  It is good research into the strategies the
big players are using.  It looks like the big money players (AT&T, Nortel,
Siemens, Cisco, etc.) are really trying to push into VoIP in a big way.  The
other big positive is the fact that people are actually, well, positive.
The buzz and the money seem to be back in the VoIP market.  I just hope that
the Asterisk community can take advantage of the change.

Asterisk Team Arrives

Mark and Greg, plus Jeremy from NuFone and Don Witt from Cylogistics
are here and manning the booth.  A good number of people wandered by to see
the display and to ask questions about Asterisk.  Lots of interest!  I
actually have seen an IAXy, so I can attest to the fact that they actually
exist.  Rumors have it that they will be available in the near future.
Similar rumors surround the FXO modules for the TDM4XX platform.

Astricon News

Several more people have expressed interest in joining the Astricon tomorrow
(Weds) evening.  If we get more than 18 people we will change our location,
so please show up in front of the conference center at 6:00 if you want to
join us.  You can also call me on my cell if you have questions:
816.806.8844.

More as it happens.  Live on the scene at VON.

Steve 


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[Asterisk-Users] ammount of packages

[Joao Carlos Moura]

Making use of Asterisk´s resources I can see that when 2 connections between
users is active, this activity
generates a huge ammount of packages on server interface where Asterisk is
running. So, I can see that Asterisk controls the calls system usage.
Is there a way to set up Asterisk to avoid this ammont of traffic control?



Thank You

João Carlos Moura

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Re: [Asterisk-Users] Cisco 7960 SIP Images

[Jon Lawrence]

On Tuesday 30 March 2004 12:34, Terence Parker wrote:
> > > Wanta take a guess what would happen if Cisco decide to really enforce
> > > the legal rules?
> >
> > I'll bite:
> >
> > Their market share would plummet in all their markets, and then smaller,
> > more innovative companies would become more able to compete with them,
> > and the overall marketplace would be vastly improved because of more
> > participants and more choices?
> >
> > B.
>
> I can't wait for that day.
>
> I don't deny that cisco make some nice products, but I don't like companies
> who have the attitude that since they're big and powerful they can invent
> whatever pricing policy they want and rip off the consumer.  Of course, the
> argument is that as a consumer I can simply choose not to buy if I don't
> want to - and indeed we are now turning towards Polycom phones rather than
> Cisco.
>
> Cisco phones are already expensive enough - it is simply cheeky that they
> should have to charge further for the "software" that runs on the phone.
> That is a joke. All hardware includes software to some degree, yet one
> doesn't have to pay creative labs for the drivers that power their
> soundcards, nor Vegastream for the bundled web manage interface. And when
> bugs are fixed, it should be the responsibility of manufacturers to update
> them - the bugs shouldn't exist in the first place.
>
> Reading through some of the arguments on this thread (both pro & anti
> Cisco) it is interesting how some feel that we should be paying Cisco the
> money they are demanding because it funds research and development - ironic
> considering this very list is about community support for a community made
> project. Asterisk, like many other open source projects, prove that
> innovation CAN and DOES take place without direct financial incentive -
> indeed the likes of sendmail, bind, apache etc... were around years before
> Microshaft came out with its equivalent tripe - and they charge piss loads
> for what is effectively a piece of shite.
>
> For the Cisco phones we DO have, we don't have any purchased licenses and I
> don't ever intend on getting any either. Cisco can sue my ass if they
> really want to.
>
I have no problem with the idea of paying cisco for software that they write.
In fact I have no problem with with paying for software full stop. But I'd 
love to have enough money to sue them if that software proved to have 
security issues or proved to be not fit for purpose - eg if a phone had a bug 
in its implementation of SIP.
If people/companies want to charge for software fine (after all it takes 
time/money to develop) but they should be willing to take the responsibility 
that goes with it. Most companies don't - at least if you cantact cisco with 
a problem then they'll do their best to fix it or at least come up with a 
work-around, which is more than a certain other companies do.

Jon

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Re: [Asterisk-Users] VON Update - Pingtel Creates new SIP Open Source Group

[Ryan Thrash]

How did the launch meeting go?

rt

On Mar 29, 2004, at 1:36 PM, Steven M. Sokol wrote:

The VON show has started off with a number of interesting 
announcements.
First among these is a big announcement from Pingtel that they have 
created
a not-for-profit corporation called SIPFoundry.  This new company 
includes
Pingtel (which has recently open sourced their SIPExchange PBX), 
Vovida and
somebody else.

More on-the-scene reports to come.
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[Asterisk-Users] Re: What failed here?

[John Chambers]

Interalab Sales wrote:
Could you have asterisk running and not allowing you to overwrite while 
trying to install?  Do you have root rights to create files in the 
asterisk folders?
Well, one of the first things I found was that nothing at all worked
unless I was root, so I've done the entire install as root.  There was
an asterisk running, so I killed its processes (lots of them), and
tried again.  I got the same errors.  I also noticed that there is
in fact an "asterisk" executable created, and it runs (for some value
of "runs").  The "asterisk -" command produces lots of output, and
of course I don't understand any of it, so I don't know whether it's
sane or not.
Another worrying thing that I've noticed:  The stuff at the start
of the make (that scrolls off the top too fast to read ;-) first does
a mkdep, and then these messages appear:
cli.c:31:19: build.h: No such file or directory
dlfcn.c:40:25: mach-o/dyld.h: No such file or directory
dlfcn.c:41:26: mach-o/nlist.h: No such file or directory
dlfcn.c:42:28: mach-o/getsect.h: No such file or directory
Sure enough, those files don't exist. Some time later, build.h does
appear, when the Makefile runs make_build.h.  It seems a bit odd that
the Makefile would attempt to use build.h before creating it.  This
looks like a sign of something wrong, but I can't tell what. Any
idea how to fix this?  Or is it actually a problem?
I don't seem to see those mach-o files anywhere.  There's also no
mach-o directory; it never appears anywhere in the build tree.
John Chambers wrote:

After doing "cvs checkout -r v1-0_stable asterisk" and typing the
usual "make clean ; make install", I got these messages:
...
make[1]: Entering directory `/usr/src/asterisk/codecs'
make -C gsm lib/libgsm.a
make[2]: Entering directory `/usr/src/asterisk/codecs/gsm'
/var/spool/asterisk   -o src/k6opt.o src/k6opt.s
make[2]: execvp: /var/spool/asterisk: Permission denied
make[2]: *** [src/k6opt.o] Error 127
...

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Re: [Asterisk-Users] Queue_log field definitions

[Jeff Crews]

I cannot remember where I found this...I thought it was in
/usr/src/asterisk/doc or perhaps in /var/log/asterisk that this
appeared:
Queue Log Information
=
In order to properly manage ACD queues, it is important to be able
to
keep track of details of call setups and teardowns in much greater
detail
than traditional call detail records provide.  In order to support
this,
extensive and detailed tracing of every queued call is stored in 
the
queue log, located (by default) in /var/log/asterisk/queue_log.
These are the events (and associated information) in the queue
log:
ABANDON(position|origposition|waittime)
The caller abandoned their position in the queue.  The position is
the
caller's position in the queue when they hungup, the origposition 
is
the original position the caller was when they first entered the
queue, and the waittime is how long the call had been waiting in the

queue at the time of disconnect.
AGENTDUMP
The agent dumped the caller while listening to the queue
announcement.
AGENTLOGIN(channel)
The agent logged in.  The channel is recorded.
AGENTLOGOFF(channel|logintime)
The agent logged off.  The channel is recorded, along with the total
time
the agent was logged in.
COMPLETEAGENT(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated
normally
by the *agent*.  The caller's hold time and the length of the call
are both
recorded.  The caller's original position in the queue is recorded
in
origposition.
COMPLETECALLER(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated
normally
by the *caller*.  The caller's hold time and the length of the call
are both
recorded.  The caller's original position in the queue is recorded
in
origposition.
CONFIGRELOAD
The configuration has been reloaded (e.g. with asterisk -rx
reload)
CONNECT(holdtime)
The caller was connected to an agent.  Hold time represents the
amount
of time the caller was on hold.
ENTERQUEUE(url|callerid)
A call has entered the queue.  URL (if specified) and Caller*ID are
placed
in the log.
EXITWITHKEY(key|position)
The caller elected to use a menu key to exit the queue.  The key
and
the caller's position in the queue are recorded.
EXITWITHTIMEOUT(position)
The caller was on hold too long and the timeout expired.
QUEUESTART
The queueing system has been started for the first time this
session.
SYSCOMPAT
A call was answered by an agent, but the call was dropped because the

channels were not compatible.
TRANSFER(extension,context)
Caller was transferred to a different extension.  Context and
extension
are recorded.
I was not good (and neither were my users) at converting the time
variables in our heads...so I crafted a dirty little script (I am sure
someone could write something better...but this is what I can pull off in
a shell script) that runs every five minutes and writes a text file we
serve up to our Asterisk users with Apache:

[Asterisk-Users] Queue feature

[Chris A. Icide]

Before I go off and post a feature request on the bug tracker, I want to 
make sure I've not misgoogled or miswikkid and not found an existing 
capability.

What I'm looking for is the ability to determine whether or not a queue has 
any queue handlers (active agents), and if it does not, bypass sending the 
caller to the queue and pass them on to a message or IVR system.

Is there a variable or some application that allows one to check number of 
agents handling a queue, or is there a configuration flag that can be set 
that will cause the priority to jump +101 or something like that when 
placing a call into an unhandled queue?

-Chris



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[Asterisk-Users] IP aliases and *

[Erick Weber V.]

Hi to all:

I'm tring to bind asterisk to two IP one IP and one IP aliases but I can not
do it

My sip.conf has bind = 0.0.0.0 and I have my server with ifcfg-eth0 and
ifcfg-eth0:1

What do I have to do

Any help will be appreciate

Thanks

Erick


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Re: [Asterisk-Users] Queue_log field definitions

[Richard Lyman]

MIS wrote:
Can anyone tell me the field definitions for the queue_log file in the 
Asterisk log directory?

1080593958|1080593892.0|salesq|NONE|ABANDON|1|1|50

fprintf(qlog, "%ld|%s|%s|%s|%s|", (long)time(NULL), callid, 
queuename, agent, event);

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[Asterisk-Users] repost: SIP/Asterisk behavior

[Lal, Deepak (Contractor)]

In my
setup, When asterisk receives a SIP INVITE request, the request URI in my case
is [EMAIL PROTECTED]
. The SIP INVITE PDU's  message header also contains a To: field. In
my case the To: field is [EMAIL PROTECTED]
 . It seems that asterisk "accepts" the request-URI number as
the called number (555) and invokes the rule for the extension 555 as
defined in my extension.conf file. 

 

But,
as per SIP, the call is really intended for 4121891 and not 555.  I'd
like asterisk to "go to"  the rule for extension 4121891 when
it receives the SIP request. Is this possible? I imagine that Asterisk would
have to act as a proxy to do this. 

 

Thanks
in advance - DL

 

RE: [Asterisk-Users] snom 200

[Ernest W. Lessenger]

> having problems with snom phone installstion

Please tell us what's up. I recently installed several SNOM phones and
worked through many minor "issues." Let me know and I'll tell you what I can
:)

--Ernest

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Re: [Asterisk-Users] Cisco 7960 SIP Images

[Terence Parker]

> >
> > Wanta take a guess what would happen if Cisco decide to really enforce
> > the legal rules?
> >
>
> I'll bite:
>
> Their market share would plummet in all their markets, and then smaller,
> more innovative companies would become more able to compete with them,
> and the overall marketplace would be vastly improved because of more
> participants and more choices?
>
> B.


I can't wait for that day.

I don't deny that cisco make some nice products, but I don't like companies
who have the attitude that since they're big and powerful they can invent
whatever pricing policy they want and rip off the consumer.  Of course, the
argument is that as a consumer I can simply choose not to buy if I don't
want to - and indeed we are now turning towards Polycom phones rather than
Cisco.

Cisco phones are already expensive enough - it is simply cheeky that they
should have to charge further for the "software" that runs on the phone.
That is a joke. All hardware includes software to some degree, yet one
doesn't have to pay creative labs for the drivers that power their
soundcards, nor Vegastream for the bundled web manage interface. And when
bugs are fixed, it should be the responsibility of manufacturers to update
them - the bugs shouldn't exist in the first place.

Reading through some of the arguments on this thread (both pro & anti Cisco)
it is interesting how some feel that we should be paying Cisco the money
they are demanding because it funds research and development - ironic
considering this very list is about community support for a community made
project. Asterisk, like many other open source projects, prove that
innovation CAN and DOES take place without direct financial incentive -
indeed the likes of sendmail, bind, apache etc... were around years before
Microshaft came out with its equivalent tripe - and they charge piss loads
for what is effectively a piece of shite.

For the Cisco phones we DO have, we don't have any purchased licenses and I
don't ever intend on getting any either. Cisco can sue my ass if they really
want to.

- Terence

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[Asterisk-Users] Patch to chan_mgcp for IP10S for test

[Daniel ANDRE]

Hello,

I am working on making chan_mgcp work properly with IP10S from 
SwissVoice. I have patched chan_mgcp from asterisk 0.7.2 and it seems to 
work pretty well but not in all cases. I give the patch with this 
message and wait for all feedback.

Best regards,

Daniel ANDRE

--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
1d0
< 
375d373
< 
766,767c764
< //For IP10S
<   if ((sub) && !( sub->cxmode == MGCP_CX_MUTE)) {
---
>   if (sub) {
1389d1385
< 
1640d1635
< 
1643d1637
< 
2228,2229d2221
< // for MGCP IP10S
< #if 0 
2234d2225
< #endif 
2239,2240c2230
< // threewaycalling = conf capability?
< if ((!sub->outgoing) && (!sub->next->outgoing) && 
(p->threewaycalling)) {
---
> if ((!sub->outgoing) && (!sub->next->outgoing)) {
2265c2255
< // IP10S   transmit_modify_request(sub);
---
> transmit_modify_request(sub);
2272,2273c2262,2263
< // IP10Shandle_hd_hf(sub->next, ev);
< 
---
> handle_hd_hf(sub->next, ev);
> #if 0
2280a2271
> #endif
2298d2288
< 

[Asterisk-Users] forget using galaxyvoice

[kc2eni]

After almost a month of battling with them I've
cancelled my account at galaxyvoice.com.

My advice to anyone considering them as a voip provider
using * as a SIP client would be "don't". 

A more unhelpfull,unknowledgable and rude group of
people   you could never wish to do business with.

Mark
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RE: [Asterisk-Users] Can Asterisk ....

[Andy Powell]

Senad,

I can do better than that:

http://bugs.digium.com/bug_view_page.php?bug_id=214

which says that the patches have been merged into cvs :D

HTH

Andy

*** REPLY SEPARATOR  ***

On 30/03/2004 at 17:00 Senad Jordanovic wrote:

>Andy Powell wrote:
 - Let the caller know its position in the queue (ie: "you are number
 # in the queue, please hold and an operator will hang on you")
>>> 
>>> This is not possible at the moment.. Anyone know better?
>> 
>> Actually it is possible have a look at the bug tracker - I would
>> give you the url but I can't get to bugs.digium.com at the moment
>> 
>Hi,
>
>Do you know which section of the bug tracker this may be in...
>As you know browsing mantis is quite slow, and searching is just very
>difficult. :)
>
>Ta
>SJ
>
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Re: [Asterisk-Users] m0nowall and *

[Ariel Batista]

Michael Graves wrote:
> Hello All,
>
> I've finally become so frustrated with my current router that I'm
> seeking alternatives. I presently use a Linksys BEFSR81 which was
> chosen for its QoS capability. However, that device quite routinely
> loses WAN<>LAN connectivity requiring a reboot. Sometimes it goes day
> or weeks just fine, sometimes I have to reboot it 4 times a day.
> Readingwwwbroadbandreports.com's VOIP forum I see that this is a known
> problem with this device. Linksys will provide no assistance
> whatsoever.

We have switched our main router to the new m0n0wall.  The soerkis
motherboard were the ones used for there development.  We are running it on
a dell celeron 1U system with just a Flash disk drive in the system.  It is
fast and works correctly.  We had some some Linksys routers out in the other
office which we are looking at replacing with this new m0n0wall.  At present
cost is a factor.  linksys are under $ 100.00 for there vpn router and the
othe conbination is around $ 400 to $500 dollars more.  The QoS on ours seem
to be working just fine.  We have had no problems with.  We do have some
issues with there PPTP.  This has given us some problems so we have to work
on this part.


> Also at the broadband Reports VOIP forum I found a post from someone
> who had started using the m0nowall firewall with his Vonage service.
> m0nowall looks interesting as it boots from CD and stores its setup
> files on floppy. I can even be run on a Soerkis Engineering embedded
> PC booting from compact flash. It appears to provide "traffic shaping"
> which I'm given to understand is like the Linksys QoS controls.
>
> Does anyone here have experience with this or similar small Linux
> firewall/router apps? Care to comment?
>
> FWIW, my home office has three SNOM 200s, a Sipura-2000, Asterisk with
> outbound calling via VoicePulse Connect, 4 incomming PSTNs.
>
> Thanks,
>
> Michael

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[Asterisk-Users] Hot plug PCI?

[Charlie Hedlin]

The quick question: Do the digium drivers for the Digium Wildcard TE410P 
(4 port T1/E1/PRI 3.3v card) , the T100P (single port T1),  and the 
TDM400P support hot plug PCI?  I am also noting that while the TDM400P 
doesn't state the voltage requirements, it looks like a 5v card.  I hope 
that I am wrong on this.

What type of T1 channel bank would people recoment for ISDN (for video 
equipment) and FXS ports?

I am looking at implementing an Asterisk PBX at my office in June.  We 
are currently recieving PBX services as part of an office sublease and 
will purchase a replacement when we move in June.

There is a lot of resistance to using any PBX that isn't from a major 
vendor with a track record in the company management.  I am going out on 
a limb for * as the features and capabilities as well as the cost seem 
like exactly what we want and need.  If I go the Asterisk route it will 
be with a serious server, redunant power supplies, raid, etc...  I will 
have a much better chance at getting aproval for this if there is hot 
plug support to give this project all of the features found in 
comporable products from Nortel and Avaya.

Thank you,
Charlie Hedlin
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Re: [Asterisk-Users] Call routing based upon callerID

[Tilghman Lesher]

On Tuesday 30 March 2004 06:25, John F. Baird wrote:
> Thanks to all those who replied. The anti ex-girlfriend facility
> seems to be doing just what I was after. Maybe I just didn't have
> enough ex-girlfriends; or maybe just not enough that turned into
> stalkers.

The anti-ex-girlfriend facility is actually more humorous than the
example provided:  the original usage was to forward the call to the
ex-gf's cell when the home number called and to the home number
when the cell number called.

-Tilghman

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RE: [Asterisk-Users] Can Asterisk ....

[Senad Jordanovic]

Andy Powell wrote:
>>> - Let the caller know its position in the queue (ie: "you are number
>>> # in the queue, please hold and an operator will hang on you")
>> 
>> This is not possible at the moment.. Anyone know better?
> 
> Actually it is possible have a look at the bug tracker - I would
> give you the url but I can't get to bugs.digium.com at the moment
> 
Hi,

Do you know which section of the bug tracker this may be in...
As you know browsing mantis is quite slow, and searching is just very
difficult. :)

Ta
SJ

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[Asterisk-Users] No audio on outgoing SIP calls over ISDN BRI line

[Leandro Morgado]

Hello,

I have asterisk installed and working nicely for internal calls using
SIP. However, when I establish an outside call, it rings and connects
properly but I get no audio on either end (the call stays connected).

Asterisk's logs say the following:

-- Executing Wait("Modem[i4l]/ttyI0", "1") in new stack
-- Executing Dial("Modem[i4l]/ttyI0",
"SIP/111|30|Ttr|SIP_CODEC=alaw") in new stack
-- Called 111
-- SIP/111-6e58 is ringing
Mar 30 14:46:27 NOTICE[278545]: channel.c:1451 ast_set_write_format:
Unable to find a path from UNKN to SLINR
-- Got SIP response 603 "Decline" back from 192.168.0.27
  == No one is available to answer at this time
-- Executing Hangup("Modem[i4l]/ttyI0", "") in new stack

I suspect it might be a codec related problem (Unable to find a path
from UNKN to SLINR) but as far as I am aware, ISDN uses G.711 and I have
both a-law and u-law activated in asterisk. I wonder why Asterik says
the outside call has an UNKN codec!

The only other possible cause I can think of, is some kind of problem
between my ISDN BRI adapter and Asterisk. I am using isdn4linux. Here
are some logs which might be usefull:

(dmesg)
ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1
loaded
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 1.1.4.1
HiSax: Layer2 Revision 1.1.4.1
HiSax: TeiMgr Revision 1.1.4.1
HiSax: Layer3 Revision 1.1.4.1
HiSax: LinkLayer Revision 1.1.4.1
HiSax: Approval certification failed because of
HiSax: unauthorized source code changes
HiSax: Total 1 card defined
HiSax: Card 1 Protocol EDSS1 Id=HiSax (0)
HiSax: Traverse Tech. NETjet-S driver Rev. 1.1.4.1
PCI: Found IRQ 9 for device 00:0d.0
PCI: Sharing IRQ 9 with 00:04.2
PCI: Sharing IRQ 9 with 00:09.0
PCI: Setting latency timer of device 00:0d.0 to 64
NETjet-S: PCI card configured at 0xb000 IRQ 9
NETjet-S: ISAC version (0): 2086/2186 V1.1
NETjet-S: IRQ 9 count 0
NETjet-S: IRQ 9 count 2
HiSax: DSS1 Rev. 1.1.4.1
HiSax: 2 channels added
HiSax: MAX_WAITING_CALLS added
HiSax: debugging flags card 1 set to 1f

(asterisk)
[chan_modem.so] => (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated
Modem Driver)
-- Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
-- Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)
  == Registered channel type 'Modem' (Generic Voice Modem Channel
Driver)

(lsmod)
Module  Size  Used byNot tainted
hisax 470288   2
isdn  122688   3  [hisax]
slhc5088   0  [isdn]

(lspci)
00:0d.0 Network controller: Tiger Jet Network Inc. Intel 537
Subsystem: Tiger Jet Network Inc. (Wrong ID) 128k ISDN-S/T
Adapter
Flags: bus master, medium devsel, latency 0, IRQ 9
I/O ports at b000 [size=256]
Memory at df80 (32-bit, non-prefetchable) [size=4K]


Anyone has any clue what might be causing this strange behaviour?

Thanks in advance,

Leandro Morgado
Eurotux / Portugal


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