date:20060105

In article [EMAIL PROTECTED], 
[EMAIL PROTECTED] says...
 Try experimenting with this:
 
 [general]
 featuredigittimeout = 1000  ; Max time (ms) between digits for
  ; feature activation.  Default is 500

It seams it works. Thank you.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Incoming PSTN Calls

2006-01-05 Thread Aisling

Hi all,

I am having difficulty getting incoming PSTN calls working.
I have set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc

My provider told me to change my sip.conf as follows

; To receive incoming calls specify this block and replace yourcontext for your dial plan.
[blueface-in]
type=peer
host=sip.blueface.ie
context=incomingpstn

And then in my extensions.conf to have something similar to
the following (or however I wanted to handle my incoming calls)

[incomingpstn]

exten =
2093,1,Wait(1)

exten =
2093,n,Background(MainMenu)

exten =
1,1,Goto(InternalExtension,2093,1) //press
1 for internal extensions.

This didnt work and I kept getting a 404 not found
error saying the user didnt exist. I tried creating the user in sip.conf
and pointing it to the appropriate context but that didnt work either.
The only way I can get it to work is to copy the code I had in the incomingpstn context of my extension.conf
to the default context. i.e.

[default]

exten =
2093,1,Wait(1)

exten =
2093,n,Background(MainMenu)

exten =
1,1,Goto(InternalExtension,2093,1)

Then the file would play. First of all I dont get why
this isIt doesnt even seem to refer to the code in my sip.confI
dont get it. Secondly whilst moving this code to the default context
means I can hear my initial welcome menu, when I press 1 to
interrupt the menu and move to menu option 1 (another sound file) it wont
let me interrupt and I eventually get the error Timeout but no rule t
in context default.

Does anyone have any ides where the problem might be?

Many thanks,

Aisling.

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[Asterisk-Users] RE: Re: Ominiis Asterisk TAPI driver

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 CounterPath's X-Pro Tapi softphone has this I think?
 
 http://www.xten.com/index.php?menu=X-Series  (select the EU region)
 
 I think they have a trial...downloading it now.

Thank you.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Re: FC3 or FC1 (or something else?)

2006-01-05 Thread Axel Thimm

On Tue, Jan 03, 2006 at 06:28:24PM -0500, Michael Stearne wrote:
 On 1/3/06, Technical Support [EMAIL PROTECTED] wrote:
  We do a lot of installs on Fedora (slowly becoming our favorite).  Initially
  clients asked for FC because of compatibility with Red Hat, great package
  management, etc.   With FC4, you get a great set of packages, and not a lot
  of add-ons required.
 
  Asterisk has perfect compatibility with FC3  FC4 - a good choice.
 
 
 I am having trouble with FC3.
 
 After doing a yum update (of 1264 packages) I still cannont compile
 1.2.1 from source:
 
 make[1]: `libedit.a' is up to date.
 make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
 make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast'
 make[1]: `libdb1.a' is up to date.
 make[1]: Leaving directory `/usr/src/asterisk-1.2.1/db1-ast'
 make[1]: Entering directory `/usr/src/asterisk-1.2.1/stdtime'
 make[1]: *** No rule to make target
 `/usr/lib/gcc/i386-redhat-linux/3.4.2/include/stddef.h', needed by
 `localtime.o'.  Stop.
 make[1]: Leaving directory `/usr/src/asterisk-1.2.1/stdtime'
 make: *** [stdtime/libtime.a] Error 2
 
 and when I try to update from binary:
 
 [EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm
 warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature: NOKEY,
 key ID 66534c2b
 error: Failed dependencies:
 libpri.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386
 libspandsp.so.0 is needed by asterisk-1.2.1-15.rhfc3.at.i386
 libtonezone.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386
 
 I have compiled from source 1.0.9 without problem on this machine.
 
 Any ideas why my attempts are now failing?

The packages containing the missing bits are also at the same place
you got this package, ATrpms. Just point apt/yum/smart/up2date to
ATrpms and have it automagically get the required dependencies
including kernel modules (kmdls).
-- 
Axel.Thimm at ATrpms.net


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[Asterisk-Users] Re: FC3 or FC1 (or something else?)

2006-01-05 Thread Axel Thimm

On Tue, Jan 03, 2006 at 04:33:49PM -, Brett, Gary wrote:
 I wish to install asterisk 1.2 (the latest tar.gz from the site not the
 CVS version) on an HP box with a TE110P (single port E1/T1)
 
 My question is which OS would be preferred in this configuration Fedora Core
 1 or Fedora Core 3, and are there any install guides out there that are
 recent enough for asterisk 1.2
 
 I am also open to suggestions for other Operating Systems if any of you feel
 that FC1/3 are not the best for the job, my only definates are that I use
 the latest tar.gz from the asterisk.org website not the CVS and also that I
 will be using the TE110p

Most people answered FC1-4 or CentOS (or RHEL in general). Whatever
you choose, there is full binary package support at ATrpms.net. So the
asterisk part should be the same on each of those distros, and you can
base your choice on other factors of the distribution like stability,
EOL dates etc.
-- 
Axel.Thimm at ATrpms.net


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[Asterisk-Users] Re: SIP security

In article [EMAIL PROTECTED], trixter@
0xdecafbad.com says...
 to add to this, given the state of MD5 and its 'security' or lack
 thereof, its a bit over simplistic to just say md5 without adding that
 its actually 3 md5 hashes...   Precomputing is harder (but not
 impossible) because of the way its done.  The nonce makes it a little
 harder - although the nonce is known even by an attacker ...

To make long story short, SIP can be cracked (like evrything else). It 
isn't so simple like sniffing the line because data is encripted. I 
don't have to put anything extra in my sip.conf (or any other conf file) 
or in my softphone for basic security (md5 encription), because all is 
allready there.

Have I got that right?


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Problem with blind transfer and Polycom phones

2006-01-05 Thread Kib Eki


Hi,

we just set up an asterisk with 55 Polycom 500 IP phones.

The blind transfer does not work.

The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey	- the display shows Blind transfer to: and cursor is 
in the second line
4. enter the number	- when we enter the second digit of the number the display 
jumps back to Hold: number view.


It is reproducible.

Attended transfer works.

Any help is welcome! :-))

Regards,
BK

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[Asterisk-Users] Iaxy Ringtone

2006-01-05 Thread bails


Hi all, I have a small query regarding ringing tones on an iaxy2.

I have a customer who uses an iaxy to breakout to pstn via our *.

However the customer complains that he gets no ringing tone whislt 
making calls, i just visited the site and can confirm this.
I also have another customer who is presently in canada with an iaxy 
calling thru our * , he doesnt have this issue.


I presume that the ringing tone is generated by the iaxy itself, and 
that therefore the one with no ringing tone is faulty.


Can anyone confirm this?

Thanks in advance

Bails
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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones !! more info

2006-01-05 Thread Kib Eki


I have to correct myself.
The problem occurs only when we try dial numbers with 10 or 11 at the beginning.


Kib Eki wrote:

Hi,

we just set up an asterisk with 55 Polycom 500 IP phones.

The blind transfer does not work.

The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey- the display shows Blind transfer to: and 
cursor is in the second line
4. enter the number- when we enter the second digit of the number 
the display jumps back to Hold: number view.


It is reproducible.

Attended transfer works.

Any help is welcome! :-))

Regards,
BK

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[Asterisk-Users] Re: [Web-MeetM] Seeking Beta testers

In article [EMAIL PROTECTED]
exch2k3.phoenix.com, [EMAIL PROTECTED] says...
 Please contact me off list if you'd like to give it a try.  

Any link or something?


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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[Asterisk-Users] Asterisk CLI | more

What is command when I wona to list something page by page in * CLI? 
Something that works like |less or |more.

Have a nice day!


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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Re: [Asterisk-Users] Re: SIP security

2006-01-05 Thread trixter aka Bret McDanel

On Thu, 2006-01-05 at 14:05 +0100, Tomislav Parcina wrote:
 In article [EMAIL PROTECTED], trixter@
 0xdecafbad.com says...
  to add to this, given the state of MD5 and its 'security' or lack
  thereof, its a bit over simplistic to just say md5 without adding that
  its actually 3 md5 hashes...   Precomputing is harder (but not
  impossible) because of the way its done.  The nonce makes it a little
  harder - although the nonce is known even by an attacker ...
 
 To make long story short, SIP can be cracked (like evrything else). It 
 isn't so simple like sniffing the line because data is encripted. I 
 don't have to put anything extra in my sip.conf (or any other conf file) 
 or in my softphone for basic security (md5 encription), because all is 
 allready there.
 
 Have I got that right?
 
 

Yeah pretty much.  While SIP can be cracked I would like to emphaise
that the benfit to 'work' ratio is such that its not likely that osmeone
would even try anything more than a simple dictionary attack so choosing
good passwords helps a lot in this regard.


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


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[Asterisk-Users] Remotely reboot SIP Phones ?

2006-01-05 Thread Jian Hong GUAN


Hi,
Can you give me some councils of remotely rebooting sip phones in asterisk 
server? How to configure sip_notify.conf and sip.conf? Kind regards,

Guan

; Reboot Polycom Phone
Event=check-sync
Content-Length=0

; Untested (Reboot Sipura Phone)
Event=resync
Content-Length=0

; Untested (Reboot GrandStream Phone)
Event=sys-control

; Untested (Reboot Cisco Phone)
Event=check-sync
Content-Length=0 


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Re: [Asterisk-Users] SIP/IAX softphones for use in call centre environments

2006-01-05 Thread Jens Vagelpohl



On 5 Jan 2006, at 09:45, Zoa wrote:



Have a look at our idefisk softphone. (available for windows, mac  
and linux).


The download links at http://www.asteriskguru.com/tools/ 
idefisk_beta.php only lead to Windoze versions, how do I get the Maxc  
version?


Thanks!

jens


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Re: [Asterisk-Users] Asterisk CLI | more

2006-01-05 Thread Tzafrir Cohen

On Thu, Jan 05, 2006 at 02:59:34PM +0100, Tomislav Parcina wrote:
 What is command when I wona to list something page by page in * CLI? 
 Something that works like |less or |more.

Scroll back in your terminal? Use screen if your terminal is not capable
of that?

less /var/log/asterisk/messages ?
(tidbit from less's help: F (shift-f): Forward forever; like tail -f)

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

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[Asterisk-Users] Re: Problem with blind transfer and Polycom phones !! more info

2006-01-05 Thread Noah Miller

Hi BK -

 The blind transfer does not work.
 
 The way we try to blind transfer a call:
 1. answer the call
 2. press transfer
 3. press blind softkey- the display shows Blind transfer to: and
 cursor is in the second line
 4. enter the number- when we enter the second digit of the number
 the display jumps back to Hold: number view.

 I have to correct myself.
 The problem occurs only when we try dial numbers with 10 or 11 at the
 beginning.

This is the Digit Map on the Polycom phone.  By default it uses 10 and 11 as
special cases (I don't know why).  You can adjust the digit map in sip.cfg.
Just take a look at the line that looks like this:

digitmap 
dialplan.digitmap=[2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[2
-9]xxxT dialplan.digitmap.timeOut=3/

You can add in more patterns separated by '|' characters for any cases you
want to cover.


- Noah

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Re: [Asterisk-Users] Zap channel instances


ast guy wrote:


for what purpose  logical channels are used?


Call waiting, three-way calling, etc.
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Re: [Asterisk-Users] Linksys SPA900 IP Key System


Kerry Garrison wrote:

Announced today, Linksys SPA9000 IP Telephony Key System

http://voipspeak.net/index.php?/content/view/60/2/


Do not post advertisements for products on this list, whether you are 
selling them or not.

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[Asterisk-Users] Second edition of my * book has been released

2006-01-05 Thread Paul Mahler

The second edition of my Asterisk book VoIP Telephony with Asterisk is now
in print. It's reorganized and expanded. 

TKS

Paul Mahler


Paul Mahler
[EMAIL PROTECTED]
 
www.signate.com
   

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[Asterisk-Users] Bizarre Answering Behavior

2006-01-05 Thread casasterisk

Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the 
lines when I can call out and the panel shows the call coming in - well 
something bizarre has happened. 
I set up inbound routing to ring my extension if a call comes in - and my 
extension rings but when I pick it up I get a dial tone. The whole time after I 
answer I hear the phone I originated the call on just ring and ring and ring, 
even though I answer the IP phone 
Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
So, why would the originating phone ring and ring if the PBX is picking up and 
routing? And why would I get dial tone on the answering phone when the incoming 
call rings to it? 
Bizarre! 
Here is the real time status from CLI: 
asterisk1*CLI 
-- Starting simple switch on 'Zap/2-1' 
-- Executing SetVar(Zap/2-1, FROM_DID=s) in new stack 
-- Executing Answer(Zap/2-1, ) in new stack 
-- Executing Wait(Zap/2-1, 0) in new stack 
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack 
-- Goto (ext-local,*101,1) 
-- Executing Macro(Zap/2-1, vm|101) in new stack 
-- Executing Macro(Zap/2-1, user-callerid) in new stack 
-- Executing DBget(Zap/2-1, AMPUSER=DEVICE//user) in new stack 
-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
-- DBget: Value not found in database. 
-- Executing DBget(Zap/2-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack 
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
-- DBget: Value not found in database. 
-- Executing GotoIf(Zap/2-1, 1?5) in new stack 
-- Goto (macro-user-callerid,s,5) 
-- Executing NoOp(Zap/2-1, Using CallerID ) in new stack 
-- Executing Goto(Zap/2-1, s-|1) in new stack 
-- Goto (macro-vm,s-,1) 
-- Executing VoiceMail(Zap/2-1, u101) in new stack 
-- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language 'en') 
-- Playing 'vm-intro' (language 'en') 
-- Playing 'beep' (language 'en') 
-- Recording the message 
-- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav49, 0x9f56790 
-- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav, 0x9f73680 
Any clues?


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Re: [Asterisk-Users] TE410P E1 Red Alarm

2006-01-05 Thread Olivier Perrin

Hi,
You could only take timing from one E1 per card.  So you should use :

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4

instead of :

span=1,1,0,ccs,hdb3,crc4
span=2,1,0,ccs,hdb3,crc4
span=3,1,0,ccs,hdb3,crc4
span=4,1,0,ccs,hdb3,crc4



Le dimanche 25 décembre 2005 à 03:07 -0600, Diyanat Ali a écrit :
 Hello!
 
 I have a TE410P quad span card with 4 E1, i am using asterisk 1.2.1, i was 
 using it without any issues earlier with just 1 E1 on span 1 and i recently 
 plugged in 3 more E1's, only span 1 is working, the e1 for span 1 is from a 
 different provider then the rest, the settings are same for both, but i 
 constantly get red alaram on the span 2,3,4, i tried all settings, including 
 the timming source , framing, coding , signalling type etc, without any 
 sucesss
 
 what maybe the cause of the red alarm
 
 Regards
 
 Diyanat
 
 lspci -vvv
 06:01.0 Communication controller: Unknown device d161:0410 (rev 02)
 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ 
 Stepping- SERR+ FastB2B-
 Status: Cap- 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- 
 TAbort- MAbort- SERR- PERR-
 Latency: 64
 Interrupt: pin A routed to IRQ 74
 Region 0: Memory at fdff (32-bit, non-prefetchable) [size=128]
 
 lsmod
 Module  Size  Used byNot tainted
 wct4xxp78432 124
 zaptel183776 250  [wct4xxp]
 
 
 
 cat /proc/zaptel/*
 Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4
 1 TE4/0/1/1 Clear (In use)  upto
 16 TE4/0/1/16 HDLCFCS (In use)  upto
 31 TE4/0/1/31 Clear (In use)
 Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED
 32 TE4/0/2/1 Clear (In use)  upto
 47 TE4/0/2/16 HDLCFCS (In use)  upto
 62 TE4/0/2/31 Clear (In use)
 Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED
 63 TE4/0/3/1 Clear (In use)  upto
 78 TE4/0/3/16 HDLCFCS (In use)  upto
 93 TE4/0/3/31 Clear (In use)
 Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED
 94 TE4/0/4/1 Clear (In use)  upto
 109 TE4/0/4/16 HDLCFCS (In use)  upto
 124 TE4/0/4/31 Clear (In use)
 
 
 ztcfg -v
 SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
 124 channels configured.
 
 zttool
 Alarms  Span
 OK  T4XXP (PCI) Card 0 Span 1
 RED T4XXP (PCI) Card 0 Span 2
 RED T4XXP (PCI) Card 0 Span 3
 RED T4XXP (PCI) Card 0 Span 4
 
 CLI zap show status
 Description  Alarms IRQbpviol 
 CRC4
 T4XXP (PCI) Card 0 Span 1OK 0  0  0
 T4XXP (PCI) Card 0 Span 2RED0  0  0
 T4XXP (PCI) Card 0 Span 3RED0  0  0
 T4XXP (PCI) Card 0 Span 4RED0  0  0
 
 
 
 alpha*CLI pri show span 1
 Primary D-channel: 16
 Status: Provisioned, Up, Active
 Switchtype: EuroISDN
 Type: CPE
 Window Length: 0/7
 Sentrej: 0
 SolicitFbit: 0
 Retrans: 0
 Busy: 0
 Overlap Dial: 0
 T200 Timer: 1000
 T203 Timer: 1
 T305 Timer: 3
 T308 Timer: 4000
 T313 Timer: 4000
 N200 Counter: 3
 
 
 alpha*CLI pri show span 2
 Primary D-channel: 47
 Status: Provisioned, In Alarm, Down, Active
 Switchtype: EuroISDN
 Type: CPE
 Window Length: 0/7
 Sentrej: 0
 SolicitFbit: 0
 Retrans: 0
 Busy: 0
 Overlap Dial: 0
 T200 Timer: 1000
 T203 Timer: 1
 T305 Timer: 3
 T308 Timer: 4000
 T313 Timer: 4000
 N200 Counter: 3
 its the same as span for the rest as upto span 4
 
 
 CLIpri intense debug span 2
 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
 Sending Set Asynchronous Balanced Mode Extended
 
 same for the rest
 
 
 zaptel.conf
 span=1,1,0,ccs,hdb3,crc4
 bchan = 1-15
 dchan = 16
 bchan = 17-31
 span=2,1,0,ccs,hdb3,crc4
 bchan = 32-46
 dchan = 47
 bchan = 48-62
 span=3,1,0,ccs,hdb3,crc4
 bchan = 63-77
 dchan = 78
 bchan = 79-93
 span=4,1,0,ccs,hdb3,crc4
 bchan = 94-108
 dchan = 109
 bchan = 110-124
 loadzone=se
 defaultzone=se
 
 
 
 zapata.conf
 [channels]
 language=us
 context=sip
 switchtype=euroisdn
 pridialplan=unknown
 prilocaldialplan=unknown
 signalling=pri_cpe
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 callreturn=yes
 echocancel=yes
 echocancelwhenbridged=yes
 echotraining=yes
 cidsignalling=dtmf
 cidstart=ring
 rxgain=0.0
 txgain=0.0
 group=1
 callgroup=1
 pickupgroup=1
 immediate=no
 callerid=asreceived
 
 group=1
 channel = 1-15
 channel = 17-31
 
 group=2
 channel = 32-46
 channel = 48-62
 
 group=3
 channel = 63-77
 channel =

[Asterisk-Users] zaptel does not compile with kernel 2.6.15

2006-01-05 Thread John Covici

Hi.  If I use kernel 2.6.15 I cannot compile zaptel modules.  I get
the following error(s) using gcc4.

  CC [M]  /usr/src/zaptel/zaptel.o
/usr/src/zaptel/zaptel.c: In function 'zt_ppp_xmit':
/usr/src/zaptel/zaptel.c:1533: warning: comparison of distinct pointer types 
lacks a cast
/usr/src/zaptel/zaptel.c: In function 'zt_register':
/usr/src/zaptel/zaptel.c:4448: warning: passing argument 2 of 
'class_device_create' makes pointer from integer without a cast
/usr/src/zaptel/zaptel.c:4448: warning: passing argument 3 of 
'class_device_create' makes integer from pointer without a cast
/usr/src/zaptel/zaptel.c:4448: warning: passing argument 4 of 
'class_device_create' from incompatible pointer type
/usr/src/zaptel/zaptel.c:4448: error: too few arguments to function 
'class_device_create'
/usr/src/zaptel/zaptel.c: In function 'zt_init':
/usr/src/zaptel/zaptel.c:6507: warning: passing argument 2 of 
'class_device_create' makes pointer from integer without a cast
/usr/src/zaptel/zaptel.c:6507: warning: passing argument 3 of 
'class_device_create' makes integer from pointer without a cast
/usr/src/zaptel/zaptel.c:6507: warning: passing argument 4 of 
'class_device_create' from incompatible pointer type
/usr/src/zaptel/zaptel.c:6507: error: too few arguments to function 
'class_device_create'
/usr/src/zaptel/zaptel.c:6508: warning: passing argument 2 of 
'class_device_create' makes pointer from integer without a cast
/usr/src/zaptel/zaptel.c:6508: warning: passing argument 3 of 
'class_device_create' makes integer from pointer without a cast
/usr/src/zaptel/zaptel.c:6508: warning: passing argument 4 of 
'class_device_create' from incompatible pointer type
/usr/src/zaptel/zaptel.c:6508: error: too few arguments to function 
'class_device_create'
/usr/src/zaptel/zaptel.c:6509: warning: passing argument 2 of 
'class_device_create' makes pointer from integer without a cast
/usr/src/zaptel/zaptel.c:6509: warning: passing argument 3 of 
'class_device_create' makes integer from pointer without a cast
/usr/src/zaptel/zaptel.c:6509: warning: passing argument 4 of 
'class_device_create' from incompatible pointer type
/usr/src/zaptel/zaptel.c:6509: error: too few arguments to function 
'class_device_create'
/usr/src/zaptel/zaptel.c:6510: warning: passing argument 2 of 
'class_device_create' makes pointer from integer without a cast
/usr/src/zaptel/zaptel.c:6510: warning: passing argument 3 of 
'class_device_create' makes integer from pointer without a cast
/usr/src/zaptel/zaptel.c:6510: warning: passing argument 4 of 
'class_device_create' from incompatible pointer type
/usr/src/zaptel/zaptel.c:6510: error: too few arguments to function 
'class_device_create'
make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1
make[1]: *** [_module_/usr/src/zaptel] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.15'
make: *** [linux26] Error 2

Any assistance would be appreciated.


-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread Chris Bagnall

 Single port GSM Gateway support 900 / 1800 GSM mode with 
 external antenna.
 Brand new unit and all of them will be tested before dispatch.
 Extremely easy to setup and can be used out of the box 
 without any configuration. So should be good alternatively of 
 phonecell or nokia pbx etc..
 Units are located in UK and £60 GBP per unit excluding shipping.

Has anyone bought one of these and able to offer some feedback? I'm
seriously considering a GSM gateway to take advantage of the spare SIM cards
lying around still inside their 12-month contracts.

Looking at the website in question, delivery is £17.37 for a 6-day delivery,
or £10 for a 30+ day delivery, both of which seem a bit high for an item
apparently located in the UK.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] Reading sound and recognizing DTMF sounds in eagi script ?

2006-01-05 Thread Robert Rozman


Hi,

we've connected Sphinx4 through eagi script (modified eagi example) to
Asterisk. Users can now say their wishes - but for gradual evolution we
would like also to provide older way of DTMF navigation too - can we 
recognize

DTMF while reading sound in eagi ?

Any advice or examples ?

Thanks in advance,

regards,

Rob.



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Re: [Asterisk-Users] call monitoring from 3th phone

2006-01-05 Thread Matt

I've found that chanspy crashes asterisk after about 10 channel spys..
asterisk just stops responding, and I have to restart it.

On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote:
 correct it only works with bridged calls.
 On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
  Tom Vile wrote:
 
  use chanspy or zapbarge
  
  
  
  That slipped my mind :). Had always been using the conf method since pre
  1.0. Does app_chanspy work with reinvite=yes? I understand it only works
  with bridged calls.
 
  On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
  
  
  [EMAIL PROTECTED] wrote:
  
  
  
  is it possible only monitoring call between phone A and B from phone C?
  
  
  
  
  
  I think you want to do service observation? You can do the following:
  a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN
  to log in and also doesn't play a tone on entry/exit (may not be legal
  in your country).
  b. Use manager API to redirect 'A' and 'B' to the conference room.
  c. 'C' joins the conference room with the mute option.
  d. C will now be able to hear what A and B are saying.
  
  
  
  
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  --
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  Baldwin Technology Solutions, Inc
  Consulting - Web Design - VoIP Telephony
  www.baldwintechsolutions.com
  Phone: 518-631-2855 x205
  Fax: 518-631-2856
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 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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[Asterisk-Users] Fax with Asterisk and Sipura 2100

2006-01-05 Thread Darrell Long

I know the subject of faxing has been covered in some detail, but I was 
wondering if anyone has a hardware configuration similar to ours that 
has faxes working successfully and would be willing to share any 
settings/insight.


We are unable to fax reliably with a Sipura 2100 connected to Asterisk. 
We do not route calls over the Internet and our network has very low 
latency. The Asterisk servers connect to Cisco Routers that have PRIs 
from various carriers. We have all the recommended settings in the 
Sipura ATA, with Echo Cancellation and Silence Suppression off, uLaw 
only for the codec, etc.


While I realize that no faxes going through passthrough like this will 
work 100% of the time, we currently have a less than 40% success rate 
with inbound faxes being the worst.


Any insight anyone has would be greatly appreciated!

Best Regards,

--
Darrell S. Long
BestWeb Corporation
 	  


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Re: [Asterisk-Users] zaptel does not compile with kernel 2.6.15


John Covici wrote:

Hi.  If I use kernel 2.6.15 I cannot compile zaptel modules.  I get
the following error(s) using gcc4.


Without telling us exactly what version of Zaptel you are trying to 
build. your report is nearly useless.


Zaptel was updated to take these API changes into account during the 
2.6.15-rc2 timeframe, I believe, so the current Zaptel in Subversion 
should be able to compile.

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Re: [Asterisk-Users] call monitoring from 3th phone

2006-01-05 Thread Tom Vile

I have not had that issue.  Are you saying 10 concurrent channels
being spied on or after the 10th it starts to crash?

On 1/5/06, Matt [EMAIL PROTECTED] wrote:
 I've found that chanspy crashes asterisk after about 10 channel spys..
 asterisk just stops responding, and I have to restart it.

 On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote:
  correct it only works with bridged calls.
  On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
   Tom Vile wrote:
  
   use chanspy or zapbarge
   
   
   
   That slipped my mind :). Had always been using the conf method since pre
   1.0. Does app_chanspy work with reinvite=yes? I understand it only works
   with bridged calls.
  
   On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
   
   
   [EMAIL PROTECTED] wrote:
   
   
   
   is it possible only monitoring call between phone A and B from phone C?
   
   
   
   
   
   I think you want to do service observation? You can do the following:
   a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN
   to log in and also doesn't play a tone on entry/exit (may not be legal
   in your country).
   b. Use manager API to redirect 'A' and 'B' to the conference room.
   c. 'C' joins the conference room with the mute option.
   d. C will now be able to hear what A and B are saying.
   
   
   
   
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   --
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   Baldwin Technology Solutions, Inc
   Consulting - Web Design - VoIP Telephony
   www.baldwintechsolutions.com
   Phone: 518-631-2855 x205
   Fax: 518-631-2856
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  --
  Tom Vile
  Baldwin Technology Solutions, Inc
  Consulting - Web Design - VoIP Telephony
  www.baldwintechsolutions.com
  Phone: 518-631-2855 x205
  Fax: 518-631-2856
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] zaptel does not compile with kernel 2.6.15

2006-01-05 Thread John covici

I did get the latest zaptel from cvs, but maybe this isn't up to date
-- sorry for the confusion.

How do y9ou determine the zaptel version for future reference?


on Thursday 01/05/2006 Kevin P. Fleming([EMAIL PROTECTED]) wrote
  John Covici wrote:
   Hi.  If I use kernel 2.6.15 I cannot compile zaptel modules.  I get
   the following error(s) using gcc4.
  
  Without telling us exactly what version of Zaptel you are trying to 
  build. your report is nearly useless.
  
  Zaptel was updated to take these API changes into account during the 
  2.6.15-rc2 timeframe, I believe, so the current Zaptel in Subversion 
  should be able to compile.
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-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

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 [EMAIL PROTECTED]
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Re: [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source

2006-01-05 Thread Brian McEntire

Wow! Thanks for all the responses! Very informative.

Erik: I'm just looking for simple dial-out and pass-along incoming cell
calls to *. Looks like the doc-n-talk should do it, except I checked
with them and, silly me, the new Samsung t309 phone I just got is not
supported yet. Hopefully it will be in a few months.

I'll check the rest of the links you and others provided in this thread.

Thanks!!On 1/3/06, GeekSpeed [EMAIL PROTECTED] wrote:
Has anyone checked out the UNIDEN ELBT-595(http://www.uniden.com/elbt/index.html)
It supposedly is a handset that can provide the same services. I have not seen any info about * compatibility though.


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Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100

2006-01-05 Thread Tom Vile

This is what I set on my Sipura:

You have to be in as admin and then advanced settings.

On the SIP page change:

RTP Packet Size: 0.010

On the Line Page:

FAX CED Detect Enable: Yes
FAX CNG Detect Enable: Yes
FAX Passthru Codec: G711u
FAX Codec Symmetric: No
FAX Passthru Method: NSE
FAX Process NSE: Yes
Release Unused Codec: Yes

Click Submit All Changes

Have not had 1 dropped fax.

On 1/5/06, Darrell Long [EMAIL PROTECTED] wrote:
 I know the subject of faxing has been covered in some detail, but I was
 wondering if anyone has a hardware configuration similar to ours that
 has faxes working successfully and would be willing to share any
 settings/insight.

 We are unable to fax reliably with a Sipura 2100 connected to Asterisk.
 We do not route calls over the Internet and our network has very low
 latency. The Asterisk servers connect to Cisco Routers that have PRIs
 from various carriers. We have all the recommended settings in the
 Sipura ATA, with Echo Cancellation and Silence Suppression off, uLaw
 only for the codec, etc.

 While I realize that no faxes going through passthrough like this will
 work 100% of the time, we currently have a less than 40% success rate
 with inbound faxes being the worst.

 Any insight anyone has would be greatly appreciated!

 Best Regards,

 --
 Darrell S. Long
 BestWeb Corporation


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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100

2006-01-05 Thread Remco Barende

I tried to get it working for a very long time (over a year) with every 
possible set of config parameters I could find both for * as well as for 
the Sipura's. Echo cancelling etc. etc. all changed but still problems.


I tried to get it working on an * box with a BRI line.

Finally I have given up and attached a traditional ISDN - Analog (A/B) 
converter to the ISDN line for the faxing bit next to Asterisk.


I have yet to find a similar solution for faxing with a PRI, I'm afraid it 
will be impossible because as far as I know it's not possible to hook up 
some sort of A/B adapter next to the * box on one pri line.


I think it can work if your fax machines are capable of capping fax tx/rx 
speeds to 9600 baud maximum without error correction. However it seems 
that not a single producer of FAX equipment (be it modems, all-in-one 
devices or even dedicated fax machines) offer such an option. HP doesn't 
seem very interested in capping the fax speeds for their all-in-one 
thingies.


All fax products keep trying to transmit/receive at higher speeds 
after which the fax will fail completely or after the second page.


Maybe there is a solution coming for PRI faxing. Junghanns informed me 
some time ago that they were working on a PRI card with a possibility to 
sync the clock to other cards.


If this works in theory you could use a Junghanns PRI card and a Junghanns 
BRI card, sync the clocks and keep the path fully digital without lost 
frames. On their website however they only mention the possibility to 
interconnect the PRI cards, not (yet?) PRI - BRI.




On Thu, 5 Jan 2006, Darrell Long wrote:

I know the subject of faxing has been covered in some detail, but I was 
wondering if anyone has a hardware configuration similar to ours that has 
faxes working successfully and would be willing to share any 
settings/insight.


We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do 
not route calls over the Internet and our network has very low latency. The 
Asterisk servers connect to Cisco Routers that have PRIs from various 
carriers. We have all the recommended settings in the Sipura ATA, with Echo 
Cancellation and Silence Suppression off, uLaw only for the codec, etc.


While I realize that no faxes going through passthrough like this will work 
100% of the time, we currently have a less than 40% success rate with inbound 
faxes being the worst.


Any insight anyone has would be greatly appreciated!

Best Regards,



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Re: [Asterisk-Users] Linksys SPA900 IP Key System


Kerry Garrison wrote:

Announced today, Linksys SPA9000 IP Telephony Key System

http://voipspeak.net/index.php?/content/view/60/2/


Do not post advertisements for products on this list, whether you are 
selling them or not.

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Re: [Asterisk-Users] TE410P E1 Red Alarm

2006-01-05 Thread Simone Cittadini


Olivier Perrin ha scritto:


Hi,
You could only take timing from one E1 per card.  So you should use :

span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4

instead of :

span=1,1,0,ccs,hdb3,crc4
span=2,1,0,ccs,hdb3,crc4
span=3,1,0,ccs,hdb3,crc4
span=4,1,0,ccs,hdb3,crc4


 

Anyway it always worked for me with timing = 1 for all spans, if I 
unplug one span I see a nessage about changing the timing source and all 
keeps working ...

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[Asterisk-Users] Call Group Limit

I recollect that there used to be a fixed, finite limit to the number of call 
groups that could exist. Does anyone know if that limitation still exists in 
1.2.1, or maybe where I could look in the code to find out if it's a fixed 
length array or not? Thanks.

Doug.
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Re: [Asterisk-Users] Call Group Limit

2006-01-05 Thread BJ Weschke

 From ast_get_group(char *s) in channel.c:

for (x = start; x = finish; x++) {
if ((x  63) || (x  0)) {
ast_log(LOG_WARNING, Ignoring invalid
group %d (maximum group is 63)\n, x);
} else


 Is this what you're looking for?

On 1/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:
 I recollect that there used to be a fixed, finite limit to the number of call 
 groups that could exist. Does anyone know if that limitation still exists in 
 1.2.1, or maybe where I could look in the code to find out if it's a fixed 
 length array or not? Thanks.

 Doug.
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--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Michael Graves

Hi All,

Until now I've only used IAX2 to connect to ITSPs. I've been toying
with a SIP connection to Gizmo Project, but not yet successfully. It
brings to mind a question. At what point does it make sense to consider
a SIP-aware firewall such as those from Ingate? 

I'd hate to move away from my m0n0wall, which is open source, easy to
manage and has served me brilliantly for two years.

Thanks,

Michael Graves
--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread bails


Chris Bagnall wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with 
external antenna.

Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box 
without any configuration. So should be good alternatively of 
phonecell or nokia pbx etc..

Units are located in UK and £60 GBP per unit excluding shipping.



Has anyone bought one of these and able to offer some feedback? I'm
seriously considering a GSM gateway to take advantage of the spare SIM cards
lying around still inside their 12-month contracts.

Looking at the website in question, delivery is £17.37 for a 6-day delivery,
or £10 for a 30+ day delivery, both of which seem a bit high for an item
apparently located in the UK.

Regards,

Chris


We were working in the area (Reading) and offered to pay cash and 
collect from their site, but the response was;


that they could only be sent direct from the far east

We weren't prepared to take the risk, I mean they turned down cash!

Bails
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Re: [Asterisk-Users] Bind asterisk to multiple IPs (reply problem)


Ales Vizdal, AVONET, s.r.o. wrote:


I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0
(ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA
registers to a.b.c.e, asterisk sends register response from a.b.c.d and
client ignores reply, because a.b.c.e != a.b.c.d. Is it bug, feature or
some kind of misconfiguration?


It's a known bug. It is being worked on, but the results won't be in an 
Asterisk release until 1.4.

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[Asterisk-Users] callback on busy

2006-01-05 Thread Hill, John


I was looking for a way to catch the zap busy return and do a redial.

I would dial out on a zap channel. If the call is busy it would then hangup
the zap channel and ask if I wanted to redial press 1 to redial or hangup
to quit.
On the 1 it would hangup the extension redial the number and call back the
extension after x number of tries. If the number still was busy then it
plays a message that it was busy would I like to continue or quit.

Thanks

--John Hill




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RE: [Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Chris Bagnall

 Until now I've only used IAX2 to connect to ITSPs. I've been 
 toying with a SIP connection to Gizmo Project, but not yet 
 successfully. It brings to mind a question. At what point 
 does it make sense to consider a SIP-aware firewall such as 
 those from Ingate? 

You should be able to run SIP through m0n0wall quite happily - we have a
number of client sites with SIP phones offsite which connect to the * server
behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
appropriate port range (as definted in rtp.conf) for the RTP streams.

You'll obviously also need to apply any QoS rules to both the SIP and RTP
streams.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] red alarm when modprobe wcte11xp

2006-01-05 Thread Phuong Nguyen

Hi all,

I have an TE11 card and I installed the zaptel driver from digium.
The zaptel.conf look like:
span=1,1,0,esf,b8sz,yellow
bchan=1-23
dchan=24

when I tried modprobe -v wcte11xp without any error message
and then ztttol
I received the error Red alarm

What would be the problem?
Thanks in advance for any suggesttion.

-- 
DSL-Aktion wegen großer Nachfrage bis 28.2.2006 verlängert:
GMX DSL-Flatrate 1 Jahr kostenlos* http://www.gmx.net/de/go/dsl
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread bbench

On Thursday 05 January 2006 17:09, Chris Bagnall wrote:
  Single port GSM Gateway support 900 / 1800 GSM mode with
  external antenna.
  Brand new unit and all of them will be tested before dispatch.
  Extremely easy to setup and can be used out of the box
  without any configuration. So should be good alternatively of
  phonecell or nokia pbx etc..
  Units are located in UK and £60 GBP per unit excluding shipping.

 Has anyone bought one of these and able to offer some feedback? I'm
 seriously considering a GSM gateway to take advantage of the spare SIM
 cards lying around still inside their 12-month contracts.

 Looking at the website in question, delivery is £17.37 for a 6-day
 delivery, or £10 for a 30+ day delivery, both of which seem a bit high for
 an item apparently located in the UK.
I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the 
pin:) so keep this in mind). There are 2 fxs ports, but I use just one; 
points to a SPA3000. The other could go to a phone set, too(I did test it)
And that's it... pretty much .  Anything else you want to do is * job and dial 
plans. When one calls from outside, first is getting authenticated against
CallerId and could then dial internal or any other destination.

It's a week I have it and works no problem. It is a little big, but much 
cheaper than other solutions, I have checked around.

The one I have came from HK during Xmas and took a little longer, but 
the freight was fine with me because I like things that just work.
Hope that helps,
benchev
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Re: [Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Erwin de Raad

- Original Message - 
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?

  Until now I've only used IAX2 to connect to ITSPs. I've been
  toying with a SIP connection to Gizmo Project, but not yet
  successfully. It brings to mind a question. At what point
  does it make sense to consider a SIP-aware firewall such as
  those from Ingate?

 You should be able to run SIP through m0n0wall quite happily - we have a
 number of client sites with SIP phones offsite which connect to the *
server
 behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
 appropriate port range (as definted in rtp.conf) for the RTP streams.

 You'll obviously also need to apply any QoS rules to both the SIP and RTP
 streams.

Totally agree. I moved from Kerio WinRoute (claims to be SIP aware  not) to
Monowall and all SIP/NAT issues went away.
It doesn't do QoS but you can do bandwith/traffic shaping which also should
work fine.

Erwin

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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones!! more info

2006-01-05 Thread Anthony Rodgers


Hi Kib,

Can you paste the dialplan string from your sip.cfg (this is the 
pattern matching string in the phone setup)?


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On Jan 5, 2006, at 5:34 AM, Kib Eki wrote:


I have to correct myself.
The problem occurs only when we try dial numbers with 10 or 11 at the 
beginning.



Kib Eki wrote:
 Hi,

 we just set up an asterisk with 55 Polycom 500 IP phones.

 The blind transfer does not work.

 The way we try to blind transfer a call:
 1. answer the call
 2. press transfer
 3. press blind softkey    - the display shows Blind transfer to: 
and

 cursor is in the second line
 4. enter the number    - when we enter the second digit of the 
number

 the display jumps back to Hold: number view.

 It is reproducible.

 Attended transfer works.

 Any help is welcome! :-))

 Regards,
 BK

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[Asterisk-Users] Problem when i make a DATA CALL

2006-01-05 Thread Dilson Freitas

We have here a TandBerg videoconferencing system connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtainingaudio calls, the video not appears.When i try to make a 64K calls (data calls), the call does not complete.


I`mtrying with asterisk branchand trunk. The problem appears with both versions.

I think that the problem is in libpri with User Information Layer 1,because when i try to make the callthe follow pri debug span 1 appears:

* When i make a speech call: (i have sucess here) ***

-- Executing Dial(mISDN/1-u3, Zap/g2d/0211130710535|20|Tt) in new stack-- Making new call for cr 32776 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: 
Q.931 (8) len=35 Call Ref: len= 2 (reference 8/0x8) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35)

*** When i make a data call: (i don't have sucess here, it does not complete thecall) **

 -- Executing Dial(mISDN/1-u4, Zap/g2d/0211130710535|20|Tt) in new stack-- Making new call for cr 32777 -- Requested transfer capability: 0x08 - DIGITAL Protocol Discriminator: 
Q.931 (8) len=45 Call Ref: len= 2 (reference 9/0x9) (Originator) Message type: SETUP (5) [04 02 88 90] Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8)
 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24)

Someone has some idea? Thanks a lot.


Cleudson Freitas


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RE: [Asterisk-Users] Call Group Limit

Thanks for that. I wonder if I could just change the x63 to something higher...

-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 05, 2006 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call Group Limit


 From ast_get_group(char *s) in channel.c:

for (x = start; x = finish; x++) {
if ((x  63) || (x  0)) {
ast_log(LOG_WARNING, Ignoring invalid
group %d (maximum group is 63)\n, x);
} else


 Is this what you're looking for?

On 1/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:
 I recollect that there used to be a fixed, finite limit to the number of call 
 groups that could exist. Does anyone know if that limitation still exists in 
 1.2.1, or maybe where I could look in the code to find out if it's a fixed 
 length array or not? Thanks.

 Doug.
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http://www.btwtech.com/
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RE: [Asterisk-Users] OT: SIP aware firewalls?

2006-01-05 Thread Erwin de Raad

- Original Message - 
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 05, 2006 5:33 PM
Subject: RE: [Asterisk-Users] OT: SIP aware firewalls?

  Until now I've only used IAX2 to connect to ITSPs. I've been
  toying with a SIP connection to Gizmo Project, but not yet
  successfully. It brings to mind a question. At what point
  does it make sense to consider a SIP-aware firewall such as
  those from Ingate?

 You should be able to run SIP through m0n0wall quite happily - we have a
 number of client sites with SIP phones offsite which connect to the *
server
 behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
 appropriate port range (as definted in rtp.conf) for the RTP streams.

 You'll obviously also need to apply any QoS rules to both the SIP and RTP
 streams.

Totally agree. I moved from Kerio WinRoute (claims to be SIP aware  not) to
Monowall and all SIP/NAT issues went away.
It doesn't do QoS but you can do bandwith/traffic shaping which also should
work fine.

Erwin

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Re: [Asterisk-Users] Bind asterisk to multiple IPs (reply problem)

2006-01-05 Thread Mr. James W. Laferriere


Hello Kevin ,

On Thu, 5 Jan 2006, Kevin P. Fleming wrote:

Ales Vizdal, AVONET, s.r.o. wrote:

I bind asterisk sip to all interfaces/addresses by bind=0.0.0.0
(ie. a.b.c.d, a.b.c.e IPs assigned on PC), I experienced problem when UA
registers to a.b.c.e, asterisk sends register response from a.b.c.d and
client ignores reply, because a.b.c.e != a.b.c.d. Is it bug, feature or
some kind of misconfiguration?


It's a known bug. It is being worked on, but the results won't be in an 
Asterisk release until 1.4.

Is the in developement functionality in the svn ?

Ie: can I do ..
svn update http://svn.digium.com/svn/asterisk/trunk asterisk
to acquire it ?
Tia ,  JimL
--
+--+
| James   W.   Laferriere | SystemTechniques | Give me VMS |
| NetworkEngineer | 3542 Broken Yoke Dr. |  Give me Linux  |
| [EMAIL PROTECTED] | Billings , MT. 59105 |   only  on  AXP |
+--+
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Re: [Asterisk-Users] New Mail Message Waiting

2006-01-05 Thread Joe Pukepail

My snom phone shows how many messages, it is a snom 360, there is the MWI light and on the lcd display it will say 5 New and 5 old messages. 
On 1/4/06, Aaron Daniel [EMAIL PROTECTED] wrote:
If the voicemail is stored locally on the server that the phone isregistering to, the phone should automatically turn MWI on.
AaronForrest Beck wrote: I am looking for a way to notify my users that there is a message waiting in voicemail.Just a simple text on the phone that says there is a new message in the mailbox.Any ideas???I sniffed around
 VoiceMail.conf samples and didn't see anything. BTW.This is a SIP 7912G Phone. Thanks!! ___ --Bandwidth and Colocation provided by 
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RE: [Asterisk-Users] Fax with Asterisk and Sipura 2100

2006-01-05 Thread Joash Herbrink

You could use a cisco ata 186.
There aren't very cheap, but I have made them work on several of my
customer sites with faxes.

The ata just registers to the * server as a SIP endpoint.

Also, echo cancelling and other intelligent things are bad when
dealing with faxes and modems.

Just use the cisco ATA (or any simple vegastream ATA device) to send /
receive faxes.

Codec should always be G.711 and no CNG or VAD or echo canceling should
be used, fax machines take care of that themselves.

(Contact me of list if you any of the mentioned devices)

Joash
[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Thursday, January 05, 2006 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100

I tried to get it working for a very long time (over a year) with every 
possible set of config parameters I could find both for * as well as for

the Sipura's. Echo cancelling etc. etc. all changed but still problems.

I tried to get it working on an * box with a BRI line.

Finally I have given up and attached a traditional ISDN - Analog (A/B) 
converter to the ISDN line for the faxing bit next to Asterisk.

I have yet to find a similar solution for faxing with a PRI, I'm afraid
it 
will be impossible because as far as I know it's not possible to hook up

some sort of A/B adapter next to the * box on one pri line.

I think it can work if your fax machines are capable of capping fax
tx/rx 
speeds to 9600 baud maximum without error correction. However it seems 
that not a single producer of FAX equipment (be it modems, all-in-one 
devices or even dedicated fax machines) offer such an option. HP doesn't

seem very interested in capping the fax speeds for their all-in-one 
thingies.

All fax products keep trying to transmit/receive at higher speeds 
after which the fax will fail completely or after the second page.

Maybe there is a solution coming for PRI faxing. Junghanns informed me 
some time ago that they were working on a PRI card with a possibility to

sync the clock to other cards.

If this works in theory you could use a Junghanns PRI card and a
Junghanns 
BRI card, sync the clocks and keep the path fully digital without lost 
frames. On their website however they only mention the possibility to 
interconnect the PRI cards, not (yet?) PRI - BRI.



On Thu, 5 Jan 2006, Darrell Long wrote:

 I know the subject of faxing has been covered in some detail, but I
was 
 wondering if anyone has a hardware configuration similar to ours that
has 
 faxes working successfully and would be willing to share any 
 settings/insight.

 We are unable to fax reliably with a Sipura 2100 connected to
Asterisk. We do 
 not route calls over the Internet and our network has very low
latency. The 
 Asterisk servers connect to Cisco Routers that have PRIs from various 
 carriers. We have all the recommended settings in the Sipura ATA, with
Echo 
 Cancellation and Silence Suppression off, uLaw only for the codec,
etc.

 While I realize that no faxes going through passthrough like this will
work 
 100% of the time, we currently have a less than 40% success rate with
inbound 
 faxes being the worst.

 Any insight anyone has would be greatly appreciated!

 Best Regards,


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[Asterisk-Users] Asterisk Debugging

I'd like to have Asterisk log useful messages during operation.

Is there any way in extensions.conf that I can manually log messages to a file, 
say via syslog()? The console output is ugly, with all the extra Executing 
NoOp(SIP/pstn.voip.com-08a28bd0, crud at the front of each line. I'm not 
sure how to save console output anyway.

Thanks,
Doug.
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Re: [Asterisk-Users] New Mail Message Waiting

2006-01-05 Thread Aaron Daniel


That's cool... the cisco's only turn a light on.

Aaron

Joe Pukepail wrote:
My snom phone shows how many messages, it is a snom 360, there is the 
MWI light and on the lcd display it will say 5 New and 5 old messages.


On 1/4/06, *Aaron Daniel* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
wrote:


If the voicemail is stored locally on the server that the phone is
registering to, the phone should automatically turn MWI on.

Aaron

Forrest Beck wrote:
  I am looking for a way to notify my users that there is a message
  waiting in voicemail.  Just a simple text on the phone that says
there
  is a new message in the mailbox.  Any ideas???  I sniffed around
  VoiceMail.conf samples and didn't see anything.
 
  BTW.  This is a SIP 7912G Phone.
 
  Thanks!!
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Re: [Asterisk-Users] TDM400P modules not found

2006-01-05 Thread Ira


At 11:48 PM 01/04/2006, you wrote:

I have a TDM400P board with two FXO modules. But the modules are not
detected when the kernel modules starts. Any ideas?


When I bought my TDM it had the 2 modules in sockets 3 and 4, as soon 
as I moved them to sockets 1 and 2 it started working.


Ira 


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Re: [Asterisk-Users] SIP/IAX softphones for use in call centre environments

2006-01-05 Thread Zoa

I will send you the alpha version offlist if you promise to give me some 
feedback :)


Zoa

Jens Vagelpohl wrote:



On 5 Jan 2006, at 09:45, Zoa wrote:



Have a look at our idefisk softphone. (available for windows, mac  
and linux).



The download links at http://www.asteriskguru.com/tools/ 
idefisk_beta.php only lead to Windoze versions, how do I get the Maxc  
version?


Thanks!

jens


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[Asterisk-Users] UserEvent() with multiple body lines

2006-01-05 Thread amaury BOSSE









Hi,

I have tried to use UserEvent() command to send data
to Asterisk Manager from my dialplan.

It works fine if the body only contains 1 line but I
dont know how to send multiple arguments in the body.



I have tested:

UserEvent(eventname|body1|body2)

UserEvent(eventname|body1\r\nbody2)

But no one seems to work.



Is it possible to do that and what is the correct syntax?



Amaury BOSSÉ






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Re: [Asterisk-Users] Iaxy Ringtone

in the iaxy's context, do you Answer before Dial?  I think this might 
remove ringing indications.  I think you either Dial first, or if Answer 
has to be first, add the r option to the Dial cmd?


Hope this helps :)

bails wrote:

Hi all, I have a small query regarding ringing tones on an iaxy2.

I have a customer who uses an iaxy to breakout to pstn via our *.

However the customer complains that he gets no ringing tone whislt 
making calls, i just visited the site and can confirm this.
I also have another customer who is presently in canada with an iaxy 
calling thru our * , he doesnt have this issue.


I presume that the ringing tone is generated by the iaxy itself, and 
that therefore the one with no ringing tone is faulty.


Can anyone confirm this?

Thanks in advance

Bails
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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones !! more info

Because the Polycom softkey menus were so cumbersome, we chose to use 
Asterisk's attended and blind transfers facility-wide.  we press ## for 
blind transfer (and Allison asks Transfer, and you type the exten num) 
and ** for attended transfer.


One more reason we chose this setup was if we added more non-polycom 
phones in the future, the users would already know how to do transfers 
without a retraining session :)


Moj

Kib Eki wrote:

I have to correct myself.
The problem occurs only when we try dial numbers with 10 or 11 at the beginning.


Kib Eki wrote:


Hi,

we just set up an asterisk with 55 Polycom 500 IP phones.

The blind transfer does not work.

The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey- the display shows Blind transfer to: and 
cursor is in the second line
4. enter the number- when we enter the second digit of the number 
the display jumps back to Hold: number view.


It is reproducible.

Attended transfer works.

Any help is welcome! :-))

Regards,
BK

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Re: [Asterisk-Users] Remotely reboot SIP Phones ?


I use polycoms, but I imagine the process is similar for many.

These three lines are the content of my sip_notify.conf:
[polycom-check-cfg]
Event=check-sync
Content-Length=0

An example SIP friend is defined as [112], so we could now type, from 
the CLI:


sip notify polycom-check-cfg 112

or to reboot multiple phones:

sip notify polycom-check-cfg 112 113 114 115

Moj

Jian Hong GUAN wrote:

Hi,
Can you give me some councils of remotely rebooting sip phones in asterisk 
server? How to configure sip_notify.conf and sip.conf? Kind regards,

Guan

; Reboot Polycom Phone
Event=check-sync
Content-Length=0

; Untested (Reboot Sipura Phone)
Event=resync
Content-Length=0

; Untested (Reboot GrandStream Phone)
Event=sys-control

; Untested (Reboot Cisco Phone)
Event=check-sync
Content-Length=0 


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re: [Asterisk-Users] Asterisk Debugging

2006-01-05 Thread Alyed Tzompa

I don't find the console output ugly, maybe messy, but never ugly :P  If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file.  Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (the default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console. i.e.  ;logger.conf  [logfiles] mylogfile = verbose Alyed I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] ChanSpy via external application

2006-01-05 Thread Dov Bigio




Hi,

I have developped an application that monitors the 
status of my queues through the events triggered on the Manager 
Interface.

This way, I can know the status of my Agent real 
time.

Now, I have a new requirement that I must allow a 
manager to click on the Agent he wants to monitor and be able to monitor the 
call.

My idea was to, when the user clicks on the Agent, 
I would Originate a call between his extension and the extension I have 
for ChanSpy, passing as parameter the Agent number.

For testing this, I tried a call file on 
/var/spool/asterisk/outgoing

Channel: 
SIP/dov.bigio 
--- This is meMaxRetries: 3RetryTime: 40WaitTime: 
25Context: 01.telecomApplication: ChanSpyData: 
Agent/5450- 
This is the Agent I want to monitorPriority: 1
The problem is that ChanSpy doesn't accept 
"Agent/" as parameter, just "Agent".
Is there a way to ChanSpy a specific know 
Agent?
(Or at least to send via dtmf the Agent Number I 
want to monitor right after the ChanSpy application is called?

Thank you very much!Dov

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RE: [Asterisk-Users] Asterisk Debugging

Well,
I want the output that the NoOp's generate. I want to be able to manually log
lines to a file through some mechanism. I just wish I could do it without all
the extra NoOp stuff at the front.

I just
tried using:
mylogfile = verbose

in
logger.conf but all I got was the startup/shutdown asterisk messages. Besides,
this isn't what I wan't. I don't want Asterisk internal generated log messages.
I want my OWN log messages, that I specify.

Doug

-Original Message-From: Alyed Tzompa
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006
11:18 AMTo: asterisk-users@lists.digium.comSubject: re:
[Asterisk-Users] Asterisk DebuggingI don't find the console output ugly, maybe messy, but never ugly
:PIf u don't like those NoOp, just take them away from ur
extensions.conf. BTW, to save the console output to a given file, just
edit your logger.conf file. Say you only want the console output, then
just add to your filename the verbose option . The file will be saved wherever
is defined in the asterisk.conf (thedefault is /var/log/asterisk)
after editing the file you'll need to do either an Asterisk restart or input
CLI logger rotate at the Asterisk console.i.e.
;logger.conf[logfiles]mylogfile = verboseAlyed

I'd like to have Asterisk log useful messages during
operation.Is there any way in extensions.conf that I can manually log
messages to a file, say via syslog()? The console output is ugly, with all the
extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each
line. I'm not sure how to save console output
anyway.Thanks,Doug.___--Bandwidth
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[Asterisk-Users] Hardware Manual

2006-01-05 Thread [EMAIL PROTECTED]


hi,

I have hacked the interface for the TE110P board to see how much of the 
Falc56 that I could access as I want a driver with a different design 
than the zaptel. I am willing to contribute to a hardware interface 
manual if someone else want to pick up the task.


Jan

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[Asterisk-Users] Call logging

2006-01-05 Thread Mark Welch









Hello all, is anyone aware of any open source call
accounting software for Asterisk? Something that can parse out Asterisk's
call detail records and generate on-demand reports?



Thanks,

Mark






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RE: [Asterisk-Users] Asterisk Debugging

2006-01-05 Thread Alyed Tzompa

Then stop looking for easy solutions and get your hands dirty changing your c files Alyed Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front. I just tried using: mylogfile = verbose in logger.conf but all I got was the startup/shutdown asterisk messages. Besides, this isn't what I wan't. I don't want Asterisk internal generated log messages. I want my OWN log messages, that I specify. Doug-Original Message-From: Alyed Tzompa [mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006 11:18 AMTo: asterisk-users@lists.digium.comSubject: re: [Asterisk-Users] Asterisk DebuggingI don't find the console output ugly, maybe messy, but never ugly :PIf u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (thedefault is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console.i.e. ;logger.conf[logfiles]mylogfile = verboseAlyed I'd like to have Asterisk log useful messages during operation.Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway.Thanks,Doug.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Asterisk Debugging

Not
everyone is a C programmer extraordinairre.

-Original Message-From: Alyed Tzompa
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006
11:59 AMTo: Douglas Garstang;
asterisk-users@lists.digium.comSubject: RE: [Asterisk-Users]
Asterisk DebuggingThen stop
looking for easy solutions and get your hands dirty changing your c
filesAlyed

Well, I want the output that the NoOp's generate. I want to be able to
manually log lines to a file through some mechanism. I just wish I could do it
without all the extra NoOp stuff at the front.

I
just tried using:
mylogfile = verbose

in
logger.conf but all I got was the startup/shutdown asterisk messages. Besides,
this isn't what I wan't. I don't want Asterisk internal generated log
messages. I want my OWN log messages, that I specify.

Doug

-Original Message-From: Alyed Tzompa
[mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006
11:18 AMTo: asterisk-users@lists.digium.comSubject:
re: [Asterisk-Users] Asterisk DebuggingI don't find the console output ugly, maybe messy, but never ugly
:PIf u don't like those NoOp, just take them away from ur
extensions.conf. BTW, to save the console output to a given file, just
edit your logger.conf file. Say you only want the console output, then
just add to your filename the verbose option . The file will be saved
wherever is defined in the asterisk.conf (thedefault is
/var/log/asterisk) after editing the file you'll need to do either an
Asterisk restart or input CLI logger rotate at the Asterisk
console.i.e. ;logger.conf[logfiles]mylogfile =
verboseAlyed

I'd like to have Asterisk log useful messages during
operation.Is there any way in extensions.conf that I can manually
log messages to a file, say via syslog()? The console output is ugly, with
all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the
front of each line. I'm not sure how to save console output
anyway.Thanks,Doug.___--Bandwidth
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[Asterisk-Users] Agent Call Recording





I'm trying to 
record calls for SPECFIC agents, which queues.conf and agents.conf don't seem to 
support. Someone suggested I just put a monitor() command before the Dial() so 
that when the Queue dials the agent, it will start 
recording.

exten = 
a00090101,1,Monitor(wav||m)
exten = a00090101,2,Dial(SIP/a00090101,20,tr)

Doing this gets me a 
few seconds ofaudio and that's it. I'm sure I had this working 
Friday.Maybe I just didn't notice that the recording was stopping. Anyone 
know how to do this?

Thanks,
Doug.


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[Asterisk-Users] DEFAULT_USERAGENT

2006-01-05 Thread Thczv F. Thczv

I work for a telecom company that allows me to peer my Asterisk box to
their system for free.  Pretty neat.  I have everything working except
that I can't get inbound VoIP calls using the DID number that my
company assigned for me.  Today, I finally discovered the source of
the problem: For various reasons (according to the technical person
who figured this out for me), the company's gear is not doing a SIP
rewrite to fix NAT issues when they get messages from a SIP endpoint
of type Asterisk.

Because of workload issues (and this being a fun project for me
personally, rather than a revenue producing project for the company),
they aren't going to fix this problem any time soon.  So, I am told,
the only solution is for me to change the Default Useragent to
something other than Asterisk PBX.

Would there be any other nasty consequences of making that change? 
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?

Thanks,

Dave
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread Jean-Michel Hiver




I have got one and and is working fine. It's exactly for
cards lying around still inside their 12-month contracts..
Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the 
pin:) so keep this in mind). There are 2 fxs ports, but I use just one; 
points to a SPA3000. The other could go to a phone set, too(I did test it)
And that's it... pretty much .  Anything else you want to do is * job and dial 
plans. When one calls from outside, first is getting authenticated against

CallerId and could then dial internal or any other destination.

It's a week I have it and works no problem. It is a little big, but much 
cheaper than other solutions, I have checked around.
 

Sounds pretty cool! Is the antenna detachable? Can you replace it with a 
longer antenna which can be stuck somewhere with decent GSM reception?


Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones !! more info

2006-01-05 Thread Jerry Jones


sounds like a digitmap issue.

We looked at using # originally, but interferred with too many IVR  
type applications from people.



On Jan 5, 2006, at 12:00 PM, Mojo with Horan  Company, LLC wrote:

Because the Polycom softkey menus were so cumbersome, we chose to  
use Asterisk's attended and blind transfers facility-wide.  we  
press ## for blind transfer (and Allison asks Transfer, and you  
type the exten num) and ** for attended transfer.


One more reason we chose this setup was if we added more non- 
polycom phones in the future, the users would already know how to  
do transfers without a retraining session :)


Moj

Kib Eki wrote:

I have to correct myself.
The problem occurs only when we try dial numbers with 10 or 11 at  
the beginning.

Kib Eki wrote:

Hi,

we just set up an asterisk with 55 Polycom 500 IP phones.

The blind transfer does not work.

The way we try to blind transfer a call:
1. answer the call
2. press transfer
3. press blind softkey- the display shows Blind transfer  
to: and cursor is in the second line
4. enter the number- when we enter the second digit of the  
number the display jumps back to Hold: number view.


It is reproducible.

Attended transfer works.

Any help is welcome! :-))

Regards,
BK

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--
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Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Asterisk Debugging

2006-01-05 Thread Mark Phillips


They're not? They have no business in an open source world then ;-}

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Douglas Garstang wrote:

Not everyone is a C programmer extraordinairre.

-Original Message-
*From:* Alyed Tzompa [mailto:[EMAIL PROTECTED]
*Sent:* Thursday, January 05, 2006 11:59 AM
*To:* Douglas Garstang; asterisk-users@lists.digium.com
*Subject:* RE: [Asterisk-Users] Asterisk Debugging

Then stop looking for easy solutions and get your hands dirty
changing your c files

Alyed



Well, I want the output that the NoOp's generate. I want to be able
to manually log lines to a file through some mechanism. I just wish
I could do it without all the extra NoOp stuff at the front.
 
I just tried using:

mylogfile = verbose
 
in logger.conf but all I got was the startup/shutdown asterisk

messages. Besides, this isn't what I wan't. I don't want Asterisk
internal generated log messages. I want my OWN log messages, that I
specify.
 
Doug
 
 


-Original Message-
*From:* Alyed Tzompa [mailto:[EMAIL PROTECTED]
*Sent:* Thursday, January 05, 2006 11:18 AM
*To:* asterisk-users@lists.digium.com
*Subject:* re: [Asterisk-Users] Asterisk Debugging

I don't find the console output ugly, maybe messy, but never ugly :P

If u don't like those NoOp, just take them away from ur
extensions.conf. BTW, to  save the console output to a given
file, just edit your logger.conf file.
Say you only want the console output, then just add to your
filename the verbose option . The file will be saved wherever is
defined in the asterisk.conf (the
 default is /var/log/asterisk) after editing the file you'll
need to do either an Asterisk restart or input CLI logger
rotate  at the Asterisk console.
i.e.
;logger.conf

[logfiles]
mylogfile = verbose


Alyed



I'd like to have Asterisk log useful messages during operation.

Is there any way in extensions.conf that I can manually log
messages to a file, say via syslog()? The console output is
ugly, with all the extra Executing
NoOp(SIP/pstn.voip.com-08a28bd0, crud at the front of each
line. I'm not sure how to save console output anyway.

Thanks,
Doug.
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Re: [Asterisk-Users] Asterisk Debugging

2006-01-05 Thread Jerry Jones


No but shell scripts are pretty easy and will cleanup your file for you.

On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote:


Not everyone is a C programmer extraordinairre.
-Original Message-
From: Alyed Tzompa [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 05, 2006 11:59 AM
To: Douglas Garstang; asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Asterisk Debugging

Then stop looking for easy solutions and get your hands dirty  
changing your c files


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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread stotaro





 I have got one and and is working fine. It's exactly for
 cards lying around still inside their 12-month contracts..
 Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the
 pin:) so keep this in mind). There are 2 fxs ports, but I use just one;
 points to a SPA3000. The other could go to a phone set, too(I did test
it)
 And that's it... pretty much .  Anything else you want to do is * job and
dial
 plans. When one calls from outside, first is getting authenticated
against
 CallerId and could then dial internal or any other destination.
 
 It's a week I have it and works no problem. It is a little big, but much
 cheaper than other solutions, I have checked around.
 
 
 Sounds pretty cool! Is the antenna detachable? Can you replace it with a
 longer antenna which can be stuck somewhere with decent GSM reception?

 Cheers,
 Jean-Michel.


Can it be used to send SMS via asterisk?

Thanks,
Steve

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Re: [Asterisk-Users] Asterisk Debugging

2006-01-05 Thread Alyed Tzompa

I do agree, plus even if you don't know anything about scripting there are plenty of shell tutorials out thereAlyed No but shell scripts are pretty easy and will cleanup your file for you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote: Not everyone is a C programmer extraordinairre. -Original Message- From: Alyed Tzompa [mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006 11:59 AM To: Douglas Garstang; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk Debugging Then stop looking for easy solutions and get your hands dirty  changing your c files___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Bizarre Answering Problem - 2ND REQUEST

2006-01-05 Thread casasterisk

Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the 
lines when I can call out and the panel shows the call coming in - well 
something bizarre has happened. 
I set up inbound routing to ring my extension if a call comes in - and my 
extension rings but when I pick it up I get a dial tone. The whole time after I 
answer I hear the phone I originated the call on just ring and ring and ring, 
even though I answer the IP phone 
Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
So, why would the originating phone ring and ring if the PBX is picking up and 
routing? And why would I get dial tone on the answering phone when the incoming 
call rings to it? 
Bizarre! 
Here is the real time status from CLI: 
asterisk1*CLI 
-- Starting simple switch on 'Zap/2-1' 
-- Executing SetVar(Zap/2-1, FROM_DID=s) in new stack 
-- Executing Answer(Zap/2-1, ) in new stack 
-- Executing Wait(Zap/2-1, 0) in new stack 
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack 
-- Goto (ext-local,*101,1) 
-- Executing Macro(Zap/2-1, vm|101) in new stack 
-- Executing Macro(Zap/2-1, user-callerid) in new stack 
-- Executing DBget(Zap/2-1, AMPUSER=DEVICE//user) in new stack 
-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
-- DBget: Value not found in database. 
-- Executing DBget(Zap/2-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack 
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
-- DBget: Value not found in database. 
-- Executing GotoIf(Zap/2-1, 1?5) in new stack 
-- Goto (macro-user-callerid,s,5) 
-- Executing NoOp(Zap/2-1, Using CallerID ) in new stack 
-- Executing Goto(Zap/2-1, s-|1) in new stack 
-- Goto (macro-vm,s-,1) 
-- Executing VoiceMail(Zap/2-1, u101) in new stack 
-- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language 'en') 
-- Playing 'vm-intro' (language 'en') 
-- Playing 'beep' (language 'en') 
-- Recording the message 
-- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav49, 0x9f56790 
-- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav, 0x9f73680 
Any clues?


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RE: [Asterisk-Users] Re: [Web-MeetM] Seeking Beta testers

2006-01-05 Thread Dan Austin


 Please contact me off list if you'd like to give it a try.  

 Any link or something?

The installation process might need more documentation, so I
asked interested parties to contact me off list.  That way
I can improve the documentation before the general public
attempts to install it.

If you'd like to check it out and provide feedback, let me
know off list.

I should point out that after I get some feedback, and make
any suggested documetation improvements, the final release
will be announced here.  The package will remain free.

Dan
 
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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones

2006-01-05 Thread C F

Looks like a digitmap problem within the Polycom configs.

On 1/5/06, Kib Eki [EMAIL PROTECTED] wrote:
 Hi,

 we just set up an asterisk with 55 Polycom 500 IP phones.

 The blind transfer does not work.

 The way we try to blind transfer a call:
 1. answer the call
 2. press transfer
 3. press blind softkey  - the display shows Blind transfer to: and cursor 
 is
 in the second line
 4. enter the number - when we enter the second digit of the number the 
 display
 jumps back to Hold: number view.

 It is reproducible.

 Attended transfer works.

 Any help is welcome! :-))

 Regards,
 BK

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RE: [Asterisk-Users] Meetme user join/leave

2006-01-05 Thread Dan Austin

 The new meetme  i  feature in asterisk1.2.1 for annoucing user
join/leave 
 is good, but the initial steps to record the name and confirm seems
lenghty, 
 the user shoudl just say the name and get into the conference, How can
i 
 disable the confirmation of the name recorded before entering the
conference

It is not configurable at the moment.  I'm think to add that feature,
since my users tend to agree with you.  The function used by app_meetme
to record the names is only used by app_meetme at the moment, so it
might
be safe to add a flag to make the review optional.

The function ast_record_review() is in app.c if you want to
unconditionally disable the review for now.

Dan

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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread bbench

On Friday 06 January 2006 00:19, Jean-Michel Hiver wrote:
 I have got one and and is working fine. It's exactly for
 cards lying around still inside their 12-month contracts..
 Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock the
 pin:) so keep this in mind). There are 2 fxs ports, but I use just one;
 points to a SPA3000. The other could go to a phone set, too(I did test it)
 And that's it... pretty much .  Anything else you want to do is * job and
  dial plans. When one calls from outside, first is getting authenticated
  against CallerId and could then dial internal or any other destination.
 
 It's a week I have it and works no problem. It is a little big, but much
 cheaper than other solutions, I have checked around.

 Sounds pretty cool! Is the antenna detachable? Can you replace it with a
 longer antenna which can be stuck somewhere with decent GSM reception?
For Remco, no I don't know who the producer is, but 
as far as I can tell the box is Chinese or something close.

The antenna is 30cm tall, on magnetic stand connected to a cable 
about 1.5m long, which could become longer I guess. One could substitute the
body with a longer on, unscrewing it from the stand
I'm keeping it sticked upon my metal desk light, hanging from the ceiling
upside down, but looking through the window for a gsm cell :)
Hope you'll like it.
benchev
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Re: [Asterisk-Users] DEFAULT_USERAGENT

2006-01-05 Thread Florian Overkamp


Hi,

Thczv F. Thczv wrote:

Would there be any other nasty consequences of making that change? 
More importantly (perhaps), is there any way to make the change in

[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?
 



We modified this on a few of our servers, without any noted ill-effect.

It's even user configurable in sip.conf:

useragent=My First SIP UA

Best regards,
Florian
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Re: [Asterisk-Users] DEFAULT_USERAGENT

2006-01-05 Thread Florian Overkamp


Hi,

Thczv F. Thczv wrote:

Would there be any other nasty consequences of making that change?
More importantly (perhaps), is there any way to make the change in
[EMAIL PROTECTED] without doing a recompile (and potentially screwing up
my system beyond my ability to repair it)?

We modified this on a few of our servers, without any noted ill-effect.

It's even user configurable in sip.conf:

useragent=My First SIP UA

Best regards,
Florian
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RE: [Asterisk-Users] Asterisk Debugging




I know 
plenty about scripting. Pick your interpreted language
However, if the functionality already existed in Asterisk, a script 
wouldn't be necessary. I'm not at the point yet where I want to start developing 
scripts for this.

  -Original Message-From: Alyed Tzompa 
  [mailto:[EMAIL PROTECTED]Sent: Thursday, January 05, 2006 
  12:38 PMTo: asterisk-users@lists.digium.comSubject: Re: 
  [Asterisk-Users] Asterisk DebuggingI do agree, plus even if you don't know anything about scripting there 
  are plenty of shell tutorials out thereAlyed 
  
  No but shell scripts are pretty easy and will cleanup your file for 
  you.On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote: 
  Not everyone is a C programmer extraordinairre. -Original 
  Message- From: Alyed Tzompa 
  [mailto:[EMAIL PROTECTED] Sent: Thursday, January 05, 2006 
  11:59 AM To: Douglas Garstang; asterisk-users@lists.digium.com 
  Subject: RE: [Asterisk-Users] Asterisk Debugging Then stop 
  looking for easy solutions and get your hands dirty  changing your c 
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[Asterisk-Users] PRI deadlock problem is 1.2.1

2006-01-05 Thread Johann

I thought this problem with PRI and channels getting out of sync was fixed in 
the 1.2.x release of Asterisk.  Here are the errors:


Jan  5 13:59:05 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jan  5 13:59:11 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jan  5 13:59:13 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.
Jan  5 13:59:14 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on 
channel 0/5 already in use on span 1.  Hanging up owner.
Jan  5 13:59:15 WARNING[1253]: chan_zap.c:8360 pri_dchannel: Ring requested on 
channel 0/2 already in use on span 1.  Hanging up owner.


Further testing and it shows the same patterns as before.  Using Queues, 
callback agents, and SIP phones.  Once a channel gets out of sync(always one 
with the queue) then no new calls can come in on the PRI line.  Existing calls 
are fine however.  In addition trying to do anything in the CLI dealing with 
queues results in no response and the refusal to do anything else in the CLI and 
it has to be killed and connected again.  Asterisk also can't restart itself 
cleanly in this state and has to be fixed.


We never tracked the problem down and people with similar reports said the 
problem didn't occur in 1.2...and it does us :(


Restarting Asterisk seems about the only way to prevent the deadlock and in a 
production PBX that accepts calls 24/7 that isn't acceptable...



--johann
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

2006-01-05 Thread bbench

On Thursday 05 January 2006 21:31, stotaro wrote:
  I have got one and and is working fine. It's exactly for
  cards lying around still inside their 12-month contracts..
  Actually it's full extend PnP-SIM-card-GSM-gateway(I forgot to unlock
   the pin:) so keep this in mind). There are 2 fxs ports, but I use just
   one; points to a SPA3000. The other could go to a phone set, too(I did
   test

 it)

  And that's it... pretty much .  Anything else you want to do is * job
   and

 dial

  plans. When one calls from outside, first is getting authenticated

 against

  CallerId and could then dial internal or any other destination.
  
  It's a week I have it and works no problem. It is a little big, but much
  cheaper than other solutions, I have checked around.
 
  Sounds pretty cool! Is the antenna detachable? Can you replace it with a
  longer antenna which can be stuck somewhere with decent GSM reception?
 
  Cheers,
  Jean-Michel.

 Can it be used to send SMS via asterisk?
Not by it self (It is rather cellsocket kind of thing),
but with an appropriate sms application, why not?
i.e. see http://tuxmobil.org/phones_linux_sms.html
for hints.
benchev

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Re: [Asterisk-Users] Problem with blind transfer and Polycom phones !! more info