Re: [asterisk-users] Changing CALLERIDNUM on the fly

[Peter Bowyer]

On 19/12/06, Doug Crompton [EMAIL PROTECTED] wrote:

Is what I am trying to do in this context possible. That is changing the
incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
preceeded by a 1 I want to add a 1. Often calls come in without the
preceeding 1 and this plays havoc with my redial if the 3 digit area
code matches a local 3 digit extension. All my outside calls are 10 digits
or 1+10 digits.

Doug


[from-pstn]
exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM})   if not add 1
exten = s,4,noop(${CALLERIDNUM})   and this still displays without


I tried no, one and two underscores with the CALLERIDNUM variable.


gonzales*CLI show function CALLERID
gonzales*CLI
 -= Info about function 'CALLERID' =-

[Syntax]
CALLERID(datatype)

[Synopsis]
Gets or sets Caller*ID data on the channel.

[Description]
Gets or sets Caller*ID data on the channel.  The allowable datatypes
are all, name, num, ANI, DNID, RDNIS.

--
Peter Bowyer
Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Inform callers on recorded/monitored number.

[Angel Heart]

Hi Paul  Eric,

Thank you for you information and quick response. I had enabled Monitoring in 
every SIP phone already. Made some Playback see below truncated config;

exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten = s,22,Goto(s-${DIALSTATUS},1)
exten = s,108,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
;Below was an added codes for the purpose of advising caller status of their 
call.
exten = s-NOANSWER,1,Playback(user)
exten = s-NOANSWER,n,Playback(is-curntly-unavail)
exten = s-NOANSWER,n,Hangup()

exten = s-ANSWER,1,Background(for-quality-purposes)
exten = s-ANSWER,n,Background(this-call-may-be)
exten = s-ANSWER,n,Background(recorded)
 
exten = s-CHANUNAVAIL,1,Playback(is-curntly-unavail)
exten = s-CHANUNAVAIL,n,Hangup()
 
exten = s-CONGESTION,1,PlayTones(congestion)
exten = s-CONGESTION,n,Wait(5)
exten = s-CONGESTION,n,StopPlayTones()
exten = s-CONGESTION,n,Hangup()

All the value of  DIALSTATUS are working except if its ANSWER, it not working 
neither the caller or callee doen't hear anything. I might inserted the message 
at the wrong .conf file. I just thought that somebody out there had tried doing 
these before.


Scenerio:

SIP phone (101) wanted to call out-side Asterisk via ISDN/PSTN (6320011). Upon 
answering by user 6320011, it hears sound like For Quality Assurance Purposes, 
this call might be monitored or recorded. It is more important for us that the 
called 6320011 should be informed about the recorded conversation and its up to 
him/her (called/6320011) to hangup or accept.

The same thing when some body called the SIP phone (101), from out-side 
Asterisk via ISDN/PSTN Trunk. The caller (from PSTN) should be informed about 
the recorded calls; Asterisk will send ringing tone then playback(For 
Quality...) continue with music(MOH) until SIP phone(101 will answer.

Hope you could provide me a little bit specific configuration on where to 
insert such scripts.

Thanks

Angel




- Original Message 
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Tuesday, December 19, 2006 1:02:34 PM
Subject: Re: [asterisk-users] Inform callers on recorded/monitored number.


exten = s,1,Answer
exten = 
s,n,Set(REC=${URIENCODE(${STRFTIME(,America/Toronto,%Y%m%d-%H%M%S)}-${CALLERID(number)}-TESTBOARD-${UNIQUEID})})
exten = s,n,MixMonitor(${REC}.wav)
exten = s,n,Playback(this-call-may-be-monitored-or-recorded)

Note that I intentionally start the recording BEFORE advising the user that the 
call may be monitored — that way the first thing on the recording is the user 
being advised of the recording.

-

With the playback command?

I think we are missing something here.

PaulH

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[asterisk-users] features.conf problems

[Lee]

Hi all,

I am having a couple of problems with features.conf I was hoping to get 
some help with.


#1.  If an outside caller is parked, when retrieved, that caller will 
now have the ability to transfer.  This only happens when they are put 
in call parking and then retrieved.


#2.  I cannot get any other keys to register for features.  For 
instance, I tried assigned blindxfer = *1, but both my grandstream and 
soft phones only react to # (pound) key.  Also, it's just the pound key. 
 As soon as it is pressed, the transferring message.


== features.conf =
transferdigittimeout = 2
courtesytone = beep
xfersound = beep
xferfailsound =beeperr
asdipark = yes
pickupexten = *8
parkingtime = 30
parkingpos = 701-720
context = parkedcalls
parkext = *70

[featuremap]
blindxfer =*1
atxfer =*2

Any help would be appreciated.

Thank you,

--

Warm Regards,

Lee

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[asterisk-users] AEL2 on Asterisk 1.2.4

[Lee]

Hey all,

I am very interested in using AEL2 (don't want to upgrade to 1.4 to get 
it though), but am having some problems upgrading/patching my asterisk 
system.  I am following the instructions on the wiki:


http://www.voip-info.org/wiki/view/Asterisk+AEL2#AEL2AnnouncementsandNews

But get the following error:

'http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/diffs.AEL2.patch' 
refers to a file, not a directory


This refers to the process of including the patch as described in the 
first portion of the wiki page.


Am am still new to linux so the problem could be just me, but I believe 
I followed the instructions.  They are pretty simple after all.


BTW, I tried both ways described and I could not get either to work.

Thanks for any help,

--

Warm Regards,

Lee

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Re: [asterisk-users] Inform callers on recorded/monitored number.

[Anselm Martin Hoffmeister]

Am Dienstag, den 19.12.2006, 01:13 -0800 schrieb Angel Heart:
 Hi Paul  Eric,
  
 Thank you for you information and quick response. I had enabled
 Monitoring in every SIP phone already. Made some Playback see below
 truncated config;
  
 exten = s,21,Dial(${pre_num:4}${the_num}${post_num},120,
 ${TRUNK_OPTIONS})
 exten = s,22,Goto(s-${DIALSTATUS},1)
 exten = s,108,Noop(max channels used up)
 exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
 exten = s-BUSY,2,Busy()
 exten = s-BUSY,3,Wait(60)
 exten = s-BUSY,4,NoOp()
 ;Below was an added codes for the purpose of advising caller status of
 their call.
 exten = s-NOANSWER,1,Playback(user)
 exten = s-NOANSWER,n,Playback(is-curntly-unavail)
 exten = s-NOANSWER,n,Hangup()
  
 exten = s-ANSWER,1,Background(for-quality-purposes)
 exten = s-ANSWER,n,Background(this-call-may-be)
 exten = s-ANSWER,n,Background(recorded)
  
 exten = s-CHANUNAVAIL,1,Playback(is-curntly-unavail)
 exten = s-CHANUNAVAIL,n,Hangup()
  
 exten = s-CONGESTION,1,PlayTones(congestion)
 exten = s-CONGESTION,n,Wait(5)
 exten = s-CONGESTION,n,StopPlayTones()
 exten = s-CONGESTION,n,Hangup()
  
 All the value of  DIALSTATUS are working except if its ANSWER, it not
 working neither the caller or callee doen't hear anything. I might
 inserted the message at the wrong .conf file. I just thought that
 somebody out there had tried doing these before.

I think the DIALSTATUS will be available only after the DIAL command
ends, which means the call either aborted (BUSY...) or one of the two
users hung up (and even then, you would have to set special flag to Dial
to continue in the dialplan instead of end connection).

You might want to test the A() or M() parameters to the dial command,
as mentioned in http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

Sorry I cannot offer dialplan excerpts. Perhaps you can write something
like on the wiki page mentioned. Possibly you would have to use A() for
outgoing calls and M() for incoming, or the other way around - I am not
sure wether those macros run on the bridged call (both sides listen to
the announcement) or to one side only.


Best regards
Anselm

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Re: [asterisk-users] Changing CALLERIDNUM on the fly

[Anselm Martin Hoffmeister]

Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
 Is what I am trying to do in this context possible. That is changing the
 incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
 preceeded by a 1 I want to add a 1. Often calls come in without the
 preceeding 1 and this plays havoc with my redial if the 3 digit area
 code matches a local 3 digit extension. All my outside calls are 10 digits
 or 1+10 digits.
 
 Doug
 
 
 [from-pstn]
 exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
 exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
 exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM})   if not add 1
 exten = s,4,noop(${CALLERIDNUM})   and this still displays without

Replace CALLERIDNUM with CALLERID(num) on all occasions, and you will
not need underscores because this is a special variable anyway.
CALLERIDNUM is obsolete.

You could get along with one line less:
exten = s,1,GotoIf($[A${CALLERID(num):0:1} = A1]?3:2)
exten = s,2,Set(CALLERID(num)=1${CALLERID(num)})
exten = s,3,NOOP(Continue in Dialplan)

Note that my GotoIf contains the two additional A letters which is
important to avoid syntax errors if the CALLERID(num) is empty for
whatever reason. I do not know what ends up in your CALLERID(num) if the
number of the caller is not available (like anonymous or withheld) -
anyway, with this statement it will end up being prepended by 1. You
migth want to have a special case for that.

If your phones happen to also display CALLERID(name) you can use this to
lookup the phone number in a phone book (here in Germany there is an
online service for number reverse lookup which works for about 50% of my
callers) and set the variable.

BR
Anselm

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[asterisk-users] 2 devices using same sip account

[rilawich ango]

Hi all,
 What will happen if 2 devices using the same set of sip account to
connect to the same asterisk?  Do they both can make call?  Can they
receive call as normal?
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Re: [asterisk-users] [Fwd: Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13]

[Jean-Yves Avenard]

Hi


On 12/19/06, Danny [EMAIL PROTECTED] wrote:

Hi Hermann !

I am using this script [ check the commented line ]


Can we please stay within the topic of this thread?

Thanks
JY
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[asterisk-users] Re: asterisk-users Digest, Vol 29, Issue 71

[sanchal . singh]

Hi,
   I want to unsubscribe from asterisk-users-request-lists, and donot
want to recieve mail any more.
   Kindly unsubscribe me...
sanchal singh  
On Mon, 2006-12-18 at 13:57, [EMAIL PROTECTED]
wrote:
 Send asterisk-users mailing list submissions to
   asterisk-users@lists.digium.com
 
 To subscribe or unsubscribe via the World Wide Web, visit
   http://lists.digium.com/mailman/listinfo/asterisk-users
 or, via email, send a message with subject or body 'help' to
   [EMAIL PROTECTED]
 
 You can reach the person managing the list at
   [EMAIL PROTECTED]
 
 When replying, please edit your Subject line so it is more specific
 than Re: Contents of asterisk-users digest...
 
 
 Today's Topics:
 
1. Re: Good Commercial Grade Service Provider? (William Piper)
2. Re: Good Commercial Grade Service Provider? (Al Bochter)
3. Re: sip peer name channel variable? (William Piper)
4. Re: Good Commercial Grade Service Provider? (William Piper)
5. Re: Good Commercial Grade Service Provider? (Al Bochter)
6. Re: BLF on GXP2000 (Andrew Joakimsen)
7. Re: Linux distro + Asterisk or Trixbox? (Andrew Joakimsen)
8. Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox?
   (Andrew Joakimsen)
9. Re: Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox?
   (Steve Sobol)
   10. Re: Multi Operator (Noc Phibee)
   11. Re: Linux distro + Asterisk or Trixbox? (Carla Schroder)
   12. Re: is it possible to use Asterisk voicemail as anouncement
   system only? (Wilson Pickett)
   13. zap sending fax congested (Ren? Enskat)
   14. Re: spandsp 0.0.3 RxFax fax reception crashes   bristuffed
   asterisk 1.2.13 (Jean-Yves Avenard)
   15. Re: Linux distro + Asterisk or Trixbox? (Vicky)
 
 
 --
 
 Message: 1
 Date: Sun, 17 Dec 2006 23:52:52 -0500
 From: William Piper [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Good Commercial Grade Service Provider?
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Message-ID:
   [EMAIL PROTECTED]
 Content-Type: text/plain; charset=iso-8859-1
 
 Check out www.cyberdyne-ip.com. Great rates, great quality, unlimited
 channels, and an easy to use GUI to manage your account.
 
 FYI, You may have more responses if you ask the -biz list.
 
 bp
 
 
 On 12/15/06, Paul Connolly [EMAIL PROTECTED] wrote:
 
   We currently have an Asterisk system with a PRI and 6 POTs lines for
  backup.  We are looking to add service such as Voicepulse Connect as an
  extra level of redundancy and a cost saving alternative to PRI calls.  VP
  Connect only allows 4 simultaneous calls; we are looking for 4 to 5 times
  that to support our call center.  Also, in looking through the archives, it
  seems like VP has had their share of outages and problems.  Can anyone
  suggest a good commercial grade package/provider?
 
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 Message: 2
 Date: Sun, 17 Dec 2006 23:58:42 -0500
 From: Al Bochter [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Good Commercial Grade Service Provider?
 To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=us-ascii
 
 I tried to setup an account with Cyberdyne-ip.com after filling out the 
 form all I get when I try to log in is
 
 Invalid User name and password please go back 
 javascript:window.history.back(); and try again
 
 If the login don't what about there service? :-\
 
 Best regards,
 
 Al Bochter
 Bochter Services
 http://www.BochterServices.com/?t=Email
 
 (VoIP PBX) 1-866-638-1254
 
 For Information on PBX Systems for SOHO
 http://www.bochterservices.com/?j=PBXt=email
 
 Need A Toll Free Number?
 http://www.bochterservices.com/?t=TFdidt=email
 
 For new and used security items
 http://www.bochterservices.com/?j=storet=email
 
 BUY Coins, Silver and Gold
 http://www.bochterservices.com/?j=goldt=email
 
 
 
 William Piper wrote:
 
  Check out www.cyberdyne-ip.com http://www.cyberdyne-ip.com. Great 
  rates, great quality, unlimited channels, and an easy to use GUI to 
  manage your account.
   
  FYI, You may have more responses if you ask the -biz list.
   
  bp
 
   
  On 12/15/06, *Paul Connolly* [EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED] wrote:
 
  We currently have an Asterisk system with a PRI and 6 POTs lines
  for backup.  We are looking to add service such as Voicepulse
  Connect as an extra level of 

RE: [asterisk-users] 2 devices using same sip account

[Gregory Duchatelet]

Hi,

It seems that they both can make calls, but only one can receive call: the
last registered...

Greg

 Hi all,
   What will happen if 2 devices using the same set of sip account to
 connect to the same asterisk?  Do they both can make call?  Can they
 receive call as normal?
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Re: [asterisk-users] Asterisk + Orion E1 GSM Gateway

[yusuf]

Hi Lex,

Ok, so I switched the Sangoma for a Digium Quad E1 card, but still now luck.
Here is my config, can you spot my mistake:

zaptel:

span=1,1,0,ccs,hdb3,crc4

bchan=1-15,17-31
dchan=16
loadzone=uk
defaultzone=uk

zapata:

immediate=no
switchtype=euroisdn
signalling=pri_cpe
group=1
callerid=asreceived
channel = 1-15,17-31

I just cant get the E1 sync light on the Orion to light up green(according to 
the manual)
I have tried crc on/off, pri_cpe/pri_net.   I'm kinda running out of ideas!  :)


Lex Lethol wrote:

Hi yusuf,

I am working right now on a similar setup.

If its the PRI type theres not so much on the syncing part.  You need
the PRI crossover rj45, theres info on voip-info on that and Orion has
software to configure via Serial cable the E1 PRI as NET/USER and Time
syncs.

I setup mine via zaptel using css,hdb3,crc on the span.
I am still debugging outogoing traffic but incoming is working OK.

Lex

On 12/18/06, yusuf [EMAIL PROTECTED] wrote:


Leo Ann Boon wrote:
 yusuf wrote:

 Hi,

 I just got hold on an Orion E1 30 port GSM Gateway, and I am having
 problems trying to get the E1 link to come up.  I am using Asteisk
 1.2.12 with a Sangoma A101 card.  I am quite familiar with E1's, both
 the Digium and Samgoma types, as I have successfully hooked up to many
 PBX's and such, but I just cant seem to get this one to work.

 None of the 30 channels 'come up'. What signailling, crc checking,
 should I be Master or slave?

 Sanity check: Have you read the fine manual :)?  I understand Orion
 makes both ISDN PRI/Q.SIG and MFC/R2 type E1 channel banks. If it's the
 PRI type, standard zaptel with the appropriate NET/CPE setting on 
the CB

 should be ok. If it's a MFC/R2, then you'll have to try unicall.

 Leo


Hi,

crazy thing is I dont have any manual or anything, just the Gateway.  
From reading the 'sales' doc
on the Orion site, this is a PRI/Q.SIg type.  But I dont have anything 
else besides that.  I dont
even know how to get the Serial cable to work to configure the Gateway 
(through
Minicom/Hyperterminal, there is a configuration on Orion, or so I'm 
told.)


Can you help?

--
thanks,
yusuf





--
thanks,
yusuf

--
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dangerous content by MailScanner, and is
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[asterisk-users] Asterisk-LDAP Integration?

[sandeep kalra]

Hi ,

 

Has anyone earlier tried integrating asterisk with LDAP.

I am interested to integrate LDAP for authentication purpose for any SIP
Incoming calls..

Pl. suggest pointers.

 

 

 

Thanks and Regards

--Sandeep Kalra



Ph: +91-120-4342000-X-2966

: +91-120-4342966 (direct)

M- 9810683168

visit: http://www.globalLogic.com

 

 

 

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[asterisk-users] Distinctive Ring detection and caller ID

[Phil Reynolds]

I have a line from BT (UK) connected to my asterisk system, on a
TDM400P.

I am able to see either distinctive ring cadences or caller ID but not
both. If I try to enable both, all drings show up as 0,0,0.

This is a pain because, if I make a call out over that line and the
number I call is busy, I can elect to camp on it (ringback), which
results in a different cadence of ring when the called number clears.

At the moment, if I try to do this, my IVR autoattendant picks up the
ringback, when ideally I want to be able to make every phone in the
house ring (distinctively in the case of the Zap devices but I have
mastered that).

I have read that this has been an issue for users in Argentina,
Australia and New Zealand, and have tried patching chan_zap.c and
recompiling, and enabling that patch, as people have had success with
there. However, it has not made any difference here.

It is more important for me to have the caller ID processing at present,
but I do need distinctive ring detection as well.

I use stable Debian with backports, meaning my asterisk is currently
1.2.10. All my packages are from there, patched as necessary.

I hope this can be resolved fairly quickly.

-- 
Phil Reynolds
 o   mail: [EMAIL PROTECTED]
|L_ \  / Web: http://www.tinsleyviaduct.com/phil/
(_)- \/  Waltham 66, Emley Moor 69, Droitwich 79, Windows 95
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Re: [asterisk-users] Distinctive Ring detection and caller ID

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 11:22:53AM +, Phil Reynolds wrote:
 I have a line from BT (UK) connected to my asterisk system, on a
 TDM400P.
 
 I am able to see either distinctive ring cadences or caller ID but not
 both. If I try to enable both, all drings show up as 0,0,0.
 
 This is a pain because, if I make a call out over that line and the
 number I call is busy, I can elect to camp on it (ringback), which
 results in a different cadence of ring when the called number clears.
 
 At the moment, if I try to do this, my IVR autoattendant picks up the
 ringback, when ideally I want to be able to make every phone in the
 house ring (distinctively in the case of the Zap devices but I have
 mastered that).
 
 I have read that this has been an issue for users in Argentina,
 Australia and New Zealand, and have tried patching chan_zap.c and
 recompiling, and enabling that patch, as people have had success with
 there. However, it has not made any difference here.
 
 It is more important for me to have the caller ID processing at present,
 but I do need distinctive ring detection as well.
 
 I use stable Debian with backports, meaning my asterisk is currently
 1.2.10. All my packages are from there, patched as necessary.
 
 I hope this can be resolved fairly quickly.

Don't know about distinctive ring. As for caller ID:

Have you set zapata.conf to use v23 signalling for callerid?

callerid=asreceived
cidsignalling=v23
cidstart=polarity

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] 2 devices using same sip account

[Johansson Olle E]

19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:


Hi,

It seems that they both can make calls, but only one can receive  
call: the

last registered...

Greg


Hi all,
  What will happen if 2 devices using the same set of sip account to
connect to the same asterisk?  Do they both can make call?  Can they
receive call as normal?
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In Asterisk, you should only have one phone per account. We do not  
support
multiple devices per account. The PBX core needs to know how many  
devices

that we are calling each time we access it.

/O
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Re: [asterisk-users] Distinctive Ring detection and caller ID

[Phil Reynolds]

On Tue, Dec 19, 2006 at 01:45:00PM +0200, Tzafrir Cohen wrote:
 Don't know about distinctive ring. As for caller ID:
 
 Have you set zapata.conf to use v23 signalling for callerid?
 
 callerid=asreceived
 cidsignalling=v23
 cidstart=polarity

Yes - and it works, but breaks distinctive ring detection, as indicated.

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

[Giedrius Augys]

2006/12/19, C F [EMAIL PROTECTED]:


Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you
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Mor and Mcc
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Re: [asterisk-users] IBM Server / USB Ports

[Matt]

So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!

I'm kind of at a loss with this machine, as I don't normally deal with
IBMs.  Here is the full output from the command.. can someone point
out where the Digium card is, because I don't see it.

[EMAIL PROTECTED] ~]# lspci -vb
00:00.0 Host bridge: Intel Corporation E7520 Memory Controller Hub (rev 0c)
   Subsystem: IBM: Unknown device 02dd
   Flags: bus master, fast devsel, latency 0
   Memory at ff00 (32-bit, non-prefetchable)
   Capabilities: [40] Vendor Specific Information

00:00.1 Class ff00: Intel Corporation E7525/E7520 Error Reporting
Registers (rev 0c)
   Subsystem: IBM: Unknown device 02dd
   Flags: fast devsel

00:02.0 PCI bridge: Intel Corporation E7525/E7520/E7320 PCI Express
Port A (rev 0c) (prog-if 00 [Normal decode])
   Flags: bus master, fast devsel, latency 0
   Bus: primary=00, secondary=02, subordinate=04, sec-latency=0
   I/O behind bridge: 4000-4fff
   Memory behind bridge: dd00-deff
   Capabilities: [50] Power Management version 2
   Capabilities: [58] Message Signalled Interrupts: 64bit-
Queue=0/1 Enable-
   Capabilities: [64] Express Root Port (Slot-) IRQ 0

00:04.0 PCI bridge: Intel Corporation E7525/E7520 PCI Express Port B
(rev 0c) (prog-if 00 [Normal decode])
   Flags: bus master, fast devsel, latency 0
   Bus: primary=00, secondary=05, subordinate=05, sec-latency=0
   Memory behind bridge: db00-dcff
   Capabilities: [50] Power Management version 2
   Capabilities: [58] Message Signalled Interrupts: 64bit-
Queue=0/1 Enable-
   Capabilities: [64] Express Root Port (Slot-) IRQ 0

00:05.0 PCI bridge: Intel Corporation E7520 PCI Express Port B1 (rev
0c) (prog-if 00 [Normal decode])
   Flags: bus master, fast devsel, latency 0
   Bus: primary=00, secondary=06, subordinate=06, sec-latency=0
   Capabilities: [50] Power Management version 2
   Capabilities: [58] Message Signalled Interrupts: 64bit-
Queue=0/1 Enable-
   Capabilities: [64] Express Root Port (Slot-) IRQ 0

00:06.0 PCI bridge: Intel Corporation E7520 PCI Express Port C (rev
0c) (prog-if 00 [Normal decode])
   Flags: bus master, fast devsel, latency 0
   Bus: primary=00, secondary=07, subordinate=09, sec-latency=0
   I/O behind bridge: 5000-
   Memory behind bridge: d900-daff
   Capabilities: [50] Power Management version 2
   Capabilities: [58] Message Signalled Interrupts: 64bit-
Queue=0/1 Enable-
   Capabilities: [64] Express Root Port (Slot-) IRQ 0

00:08.0 System peripheral: Intel Corporation E7525/E7520/E7320
Extended Configuration Registers (rev 0c)
   Subsystem: IBM: Unknown device 02dd
   Flags: fast devsel

00:1d.0 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB
UHCI Controller #1 (rev 02) (prog-if 00 [UHCI])
   Subsystem: IBM: Unknown device 02dd
   Flags: bus master, medium devsel, latency 0, IRQ 11
   I/O ports at 2200

00:1d.1 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB
UHCI Controller #2 (rev 02) (prog-if 00 [UHCI])
   Subsystem: IBM: Unknown device 02dd
   Flags: bus master, medium devsel, latency 0, IRQ 4
   I/O ports at 2600

00:1d.7 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB2
EHCI Controller (rev 02) (prog-if 20 [EHCI])
   Subsystem: IBM: Unknown device 02dd
   Flags: bus master, medium devsel, latency 0, IRQ 5
   Memory at f900 (32-bit, non-prefetchable)
   Capabilities: [50] Power Management version 2
   Capabilities: [58] Debug port

00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev c2)
(prog-if 00 [Normal decode])
   Flags: bus master, fast devsel, latency 0
   Bus: primary=00, secondary=01, subordinate=01, sec-latency=32
   I/O behind bridge: 3000-3fff
   Memory behind bridge: f800-f8ff
   Prefetchable memory behind bridge: f000-f7ff

00:1f.0 ISA bridge: Intel Corporation 82801EB/ER (ICH5/ICH5R) LPC
Interface Bridge (rev 02)
   Flags: bus master, medium devsel, latency 0

00:1f.1 IDE interface: Intel Corporation 82801EB/ER (ICH5/ICH5R) IDE
Controller (rev 02) (prog-if 8a [Master SecP PriP])
   Subsystem: IBM: Unknown device 02dd
   Flags: bus master, medium devsel, latency 0
   I/O ports at 01f0
   I/O ports at 03f4
   I/O ports at 0170
   I/O ports at 0374
   I/O ports at 0480
   Memory at 8000 (32-bit, non-prefetchable)

00:1f.3 SMBus: Intel Corporation 82801EB/ER (ICH5/ICH5R) SMBus
Controller (rev 02)
   Subsystem: IBM: Unknown device 02dd
   Flags: medium devsel, IRQ 7
   I/O ports at 0440

01:06.0 VGA compatible controller: ATI Technologies Inc Radeon RV100
QY [Radeon 7000/VE] (prog-if 00 [VGA])
   Subsystem: IBM: Unknown device 02c8
   Flags: bus master, stepping, medium devsel, latency 64, IRQ 3
   Memory at f000 

[asterisk-users] dtmf and ivr

[René Enskat]

hello,

i try to build a IVR for our company my problem is that the dtmf tones
are not recognized by the phones i tried several phones.
BUT when i call the voicemail i can navigate with all phones through the
menu. I use * 1.2

here is the context:

[ivr]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
;SAI menu - 1 for Sales, 2 for Support
exten = s,5,Background(say-menu1)
exten = s,6,Background(say-menu2)

; Sales
exten = 1,1,SetGlobalVar(ACCOUNTCODE=${callerid})
exten = 1,2,SetVar(callerid=${callerid})
exten = 1,3,Background(sai-welcome-sales)
exten = 1,4,Queue(sales)

; Tech Support
exten = 2,1,SetGlobalVar(ACCOUNTCODE=${callerid})
exten = 2,2,SetVar(callerid=${callerid})
exten = 2,3,Background(sai-welcome-support)
exten = 2,4,Queue(support)

; # =hangup
exten = #,1,Playback(thank-you-for-calling)
exten = #,2,Hangup

exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again

Somebody can give me a hint plz.

regards rene


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RE: [asterisk-users] BLF on GXP2000

[Ken Williams]

One thing I've noticed, is any time I make changes to Asterisk I have to
reboot the phones to keep BLF up to date.  Have you tried that?



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Monday, December 18, 2006 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF on GXP2000


Well, I am making some progress. I have made some changes as defined
below and now have a green line on the BLF, but it still does not
indicate when the extension receives a call or goes off hook. 

Here are the changes: 
the [ext-local-custom] context no longer exists
the subscribecontext in sip.con no longer exists

[internal]
exten = 101,1,Macro(voicemail,${polycom430})
exten = 101,hint,${polycom430}

Asterisk 1.4.0b3
*CLI show hints

-= Registered Asterisk Dial Plan Hints =-
[EMAIL PROTECTED]: SIP/101
State:Idle   Watchers  1

- 1 hints registered 
 


On 12/18/06, Ken Williams [EMAIL PROTECTED] wrote: 

Here's what I have, it's to early for me to think so hopefully
looking at mine helps :D
 
extensions.conf:
 
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701

sip.conf:
 
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/701
context=from-internal 
canreinvite=no
callerid=device 701
mailbox=701

If this doesn't help in some fashion let me know and I'll think
it through a little later...off to get some coffee.
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Johnson
Sent: Sunday, December 17, 2006 4:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] BLF on GXP2000



I am trying to set up the BLF on a GXP2000. 
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101

[internal] 
exten = 101,1,Macro(voicemail,${polycom430})


[macro-voicemail]
exten = s,1,Dial(${ARG1},10,tT)
exten = s,2,VoiceMail([EMAIL PROTECTED] )
exten = s,102,VoiceMail([EMAIL PROTECTED])

[ext-local-custom]
exten = 101,hint,${polycom430}


sip.conf:
[general]
subscribecontext=ext-local-custom

And have set up the key to Asterisk BLF with UserID101 

When I reload the phone, I get the following error:
[Dec 17 13:34:33] ERROR[2551]: chan_sip.c:14064
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED] from
192.168.1.248 http://192.168.1.248/ , but there is no hint for that
extension


Any help is greatly appreciated.

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Re: [asterisk-users] Remote Reboot of a Polycom

[Jerry Jones]

Or web into the phone and click any submit button - not a great idea  
though if you remotely provision, just make sure you do not change  
any settings as they will then over ride the remote file settings



On Dec 19, 2006, at 1:09 AM, Douglas Garstang wrote:


From the Asterisk console:


sip notify polycom-check-cfg ipaddr

Or you might have to switch the polycom-check-cfg and the ip. I  
forget the order. You also need to make sure that the phone has  
alwaysreboot=1 in the sip.cfg xml file.


Doug.


-Original Message-
From:   Klaverstyn, David C [mailto:[EMAIL PROTECTED]
Sent:   Mon 12/18/2006 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 
Subject:[asterisk-users] Remote Reboot of a Polycom

Does anyone know how to remotely reboot a PolyCom specifically 601
phone?






winmail.dat
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[asterisk-users] Re: IBM Server / USB Ports

[Steven]

Could the unknown device be a management card?

The newer dells have a management card built into the fist ethernet controller.

-- 
-- 
Steven

http://www.glimasoutheast.org



Matt [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
 So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' 
 followed!!

 I'm kind of at a loss with this machine, as I don't normally deal with
 IBMs.  Here is the full output from the command.. can someone point
 out where the Digium card is, because I don't see it.

 [EMAIL PROTECTED] ~]# lspci -vb
 00:00.0 Host bridge: Intel Corporation E7520 Memory Controller Hub (rev 0c)
Subsystem: IBM: Unknown device 02dd
Flags: bus master, fast devsel, latency 0
Memory at ff00 (32-bit, non-prefetchable)
Capabilities: [40] Vendor Specific Information

 00:00.1 Class ff00: Intel Corporation E7525/E7520 Error Reporting
 Registers (rev 0c)
Subsystem: IBM: Unknown device 02dd
Flags: fast devsel

 00:02.0 PCI bridge: Intel Corporation E7525/E7520/E7320 PCI Express
 Port A (rev 0c) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=02, subordinate=04, sec-latency=0
I/O behind bridge: 4000-4fff
Memory behind bridge: dd00-deff
Capabilities: [50] Power Management version 2
Capabilities: [58] Message Signalled Interrupts: 64bit-
 Queue=0/1 Enable-
Capabilities: [64] Express Root Port (Slot-) IRQ 0

 00:04.0 PCI bridge: Intel Corporation E7525/E7520 PCI Express Port B
 (rev 0c) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=05, subordinate=05, sec-latency=0
Memory behind bridge: db00-dcff
Capabilities: [50] Power Management version 2
Capabilities: [58] Message Signalled Interrupts: 64bit-
 Queue=0/1 Enable-
Capabilities: [64] Express Root Port (Slot-) IRQ 0

 00:05.0 PCI bridge: Intel Corporation E7520 PCI Express Port B1 (rev
 0c) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=06, subordinate=06, sec-latency=0
Capabilities: [50] Power Management version 2
Capabilities: [58] Message Signalled Interrupts: 64bit-
 Queue=0/1 Enable-
Capabilities: [64] Express Root Port (Slot-) IRQ 0

 00:06.0 PCI bridge: Intel Corporation E7520 PCI Express Port C (rev
 0c) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=07, subordinate=09, sec-latency=0
I/O behind bridge: 5000-
Memory behind bridge: d900-daff
Capabilities: [50] Power Management version 2
Capabilities: [58] Message Signalled Interrupts: 64bit-
 Queue=0/1 Enable-
Capabilities: [64] Express Root Port (Slot-) IRQ 0

 00:08.0 System peripheral: Intel Corporation E7525/E7520/E7320
 Extended Configuration Registers (rev 0c)
Subsystem: IBM: Unknown device 02dd
Flags: fast devsel

 00:1d.0 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB
 UHCI Controller #1 (rev 02) (prog-if 00 [UHCI])
Subsystem: IBM: Unknown device 02dd
Flags: bus master, medium devsel, latency 0, IRQ 11
I/O ports at 2200

 00:1d.1 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB
 UHCI Controller #2 (rev 02) (prog-if 00 [UHCI])
Subsystem: IBM: Unknown device 02dd
Flags: bus master, medium devsel, latency 0, IRQ 4
I/O ports at 2600

 00:1d.7 USB Controller: Intel Corporation 82801EB/ER (ICH5/ICH5R) USB2
 EHCI Controller (rev 02) (prog-if 20 [EHCI])
Subsystem: IBM: Unknown device 02dd
Flags: bus master, medium devsel, latency 0, IRQ 5
Memory at f900 (32-bit, non-prefetchable)
Capabilities: [50] Power Management version 2
Capabilities: [58] Debug port

 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev c2)
 (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=01, subordinate=01, sec-latency=32
I/O behind bridge: 3000-3fff
Memory behind bridge: f800-f8ff
Prefetchable memory behind bridge: f000-f7ff

 00:1f.0 ISA bridge: Intel Corporation 82801EB/ER (ICH5/ICH5R) LPC
 Interface Bridge (rev 02)
Flags: bus master, medium devsel, latency 0

 00:1f.1 IDE interface: Intel Corporation 82801EB/ER (ICH5/ICH5R) IDE
 Controller (rev 02) (prog-if 8a [Master SecP PriP])
Subsystem: IBM: Unknown device 02dd
Flags: bus master, medium devsel, latency 0
I/O ports at 01f0
I/O ports at 03f4
I/O ports at 0170
I/O ports at 0374
I/O ports at 0480
Memory at 8000 (32-bit, non-prefetchable)

 00:1f.3 SMBus: Intel Corporation 82801EB/ER (ICH5/ICH5R) SMBus
 Controller (rev 02)
Subsystem: 

Re: [asterisk-users] Asterisk-LDAP Integration?

[Steve Davies]

Google: asterisk ldap

http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+ldap.conf

Never done it myself though.

Steve

On 12/19/06, sandeep kalra [EMAIL PROTECTED] wrote:

Hi ,

Has anyone earlier tried integrating asterisk with LDAP.

I am interested to integrate LDAP for authentication purpose for any SIP
Incoming calls..

Pl. suggest pointers.


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[asterisk-users] G.279 license question

[Michel]

Hi,

I need to connect to a remote VOIP server that only uses G.729 codec.
From our Asterisk server,  we will then make several calls ( 1 but  
?? !!) in the same time to the remote VOIP server.


Do we need to purchase Asterisk G.279 license ? If yes, how many 
licenses must we buy?



Thanks you!

Michel



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Re: [asterisk-users] G.279 license question

[Jerry Jones]

OK with the remote server on one side doing G729, what will you be  
connecting to on the other side? If it does G729 then no license, if  
not then one license per active call. Also if * will be doing any  
voicemail etc then you will also need the license.



On Dec 19, 2006, at 8:31 AM, Michel wrote:


Hi,

I need to connect to a remote VOIP server that only uses G.729 codec.
From our Asterisk server,  we will then make several calls ( 1 but  
 ?? !!) in the same time to the remote VOIP server.


Do we need to purchase Asterisk G.279 license ? If yes, how many  
licenses must we buy?



Thanks you!

Michel



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[asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME

[Ex Vitorino]

 (1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)

-- Forwarded message --
From: Ex Vitorino [EMAIL PROTECTED]
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com



 Hello Asterisk Users,


 I guess the subject says the most of it; here goes some more
 detail:

 - Running Asterisk 1.2.14
 - Objective: record all calls managed by a specific queue
 - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}

 Facts:

 - If the UNIQUEID chan var is used in the MONITOR_FILENAME,
   before calling the Queue() application, the two legs of the call are
   not mixed and I end up with the two separate -in / -out files

 - If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
   then, the legs are mixed together...

 Note:

 - In my first attempt I never managed to get the legs mixed... Only
   after some experiment, I understood (well, not 100% clear why!)
   that I had to also to add to include recordagentcalls=yes and
   monitor-join=yes in agents.conf !


 Can anyone provide some insight into this ? Thanks in advance!

 (see below for config)
--
 Ex Vito



 queues.conf:

   [general]
   persistentmembers = yes

   [the_queue]
   musiconhold = default
   announce = the_announcement
   strategy = ringall
   servicelevel = 20
   context = the_context
   wrapuptime = 10
   announce-frequency = 30
   announce-holdtime = once
   monitor-format = wav
   monitor-join = yes
   eventwhencalled = yes
   eventmemberstatus = no
   reportholdtime = no
   member = SIP/sip0001


 agents.conf:

   [general]
   persistentagents=yes
   recordagencalls=yes
   monitor-join = yes
   [agents]

   (no agents declared, as they are directly configured in the
queues.conf file)


 extensions.conf:

   ...
   [globals]
   SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support

   [the_context]

   exten = 305,1,Answer()
   exten = 
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM})
exten = 305,n,Queue(the_queue,t)
exten = 305,n,Hangup()
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[asterisk-users] Automatic sip conference

[nik600]

Hi can i do that on asterisk?

- receive a call from h323
- call an internal sip extension AAA

AAA does meetme on

SIP/[EMAIL PROTECTED]
SIP/[EMAIL PROTECTED]

Finally, the caller from h323, user1 and user2 can speak together...

Is it possible?
Many thanks..
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Re: [asterisk-users] IBM Server / USB Ports

[Time Bandit]

So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!

I would try moving the Digium card to another slot. Your Ethernet
controlled must be onboard and it share its IRQ with the slot where
the Digium board is.

hth
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[asterisk-users] .Call files do not seem to work

[Lee]

Hi,

I was trying out call file just to see how they worked and my system 
does not seem to do anything with them, although asterisk *is* deleting 
the files that I put into /var/spool/asterisk/outgoing.


1. I nano'd a quick call file like so:

Channel: SIP/axVoice/910555
CallerID : Leebo 55
MaxRetries: 2
RetryTime: 30
WaitTime: 10
Context: main_menu
Extension: s
Priority: 1

2. And then mv'd to /var/spool/asterisk/outgoing

As I mentioned, Asterisk appears to be grabbing the file, but there is 
no call made.



Q. Do calls originated like this show up in CLI output?

Q. The context portion of the package refers to the context to place the 
call in after the remote person answers, right?  Or is it the context 
that the origination should dial out on?  I've tried both ways just in 
case, but no go.


Thanks for any help.


--

Warm Regards,

Lee

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Re: [asterisk-users] Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME

[James Fromm]

I spent hours debugging this a few weeks ago.

The ${UNIQUEID} contains a period (.).  Mine are something like 
.xx.  When soxmix is executed to mix the in and out files, the 
file types are not specified.  This causes soxmix to attempt to 
determine the file type by the filename's extension.  The routine in sox 
that looks for the filename's extension doesn't expect multiple periods 
in the filename.  So it finds the file type to be xx.wav (or xx.gsm) and 
that's not a format sox can handle.


You can add an AGI call to your dialplan immediately after the Queue 
application to join the files.


Ex Vitorino wrote:

 (1st attempt was rejected by postfix @lists.digium.com, here goes the 2nd)

-- Forwarded message --
From: Ex Vitorino [EMAIL PROTECTED]
Date: Dec 18, 2006 11:41 PM
Subject: Queue Monitor not mixing if using UNIQUEID in MONITOR_FILENAME
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com



 Hello Asterisk Users,


 I guess the subject says the most of it; here goes some more
 detail:

 - Running Asterisk 1.2.14
 - Objective: record all calls managed by a specific queue
 - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID}

 Facts:

 - If the UNIQUEID chan var is used in the MONITOR_FILENAME,
   before calling the Queue() application, the two legs of the call are
   not mixed and I end up with the two separate -in / -out files

 - If it isn't used (for example: using only ${TIMESTAMP]-${CALLERIDNUM})
   then, the legs are mixed together...

 Note:

 - In my first attempt I never managed to get the legs mixed... Only
   after some experiment, I understood (well, not 100% clear why!)
   that I had to also to add to include recordagentcalls=yes and
   monitor-join=yes in agents.conf !


 Can anyone provide some insight into this ? Thanks in advance!

 (see below for config)
--
 Ex Vito



 queues.conf:

   [general]
   persistentmembers = yes

   [the_queue]
   musiconhold = default
   announce = the_announcement
   strategy = ringall
   servicelevel = 20
   context = the_context
   wrapuptime = 10
   announce-frequency = 30
   announce-holdtime = once
   monitor-format = wav
   monitor-join = yes
   eventwhencalled = yes
   eventmemberstatus = no
   reportholdtime = no
   member = SIP/sip0001


 agents.conf:

   [general]
   persistentagents=yes
   recordagencalls=yes
   monitor-join = yes
   [agents]

   (no agents declared, as they are directly configured in the
queues.conf file)


 extensions.conf:

   ...
   [globals]
   SUPPORT_MONITOR_PATH=/var/spool/asterisk/monitor/support

   [the_context]

   exten = 305,1,Answer()
   exten = 
305,n,Set(MONITOR_FILENAME=${SUPPORT_MONITOR_PATH}/${TIMESTAMP}-${CALLERIDNUM}) 


exten = 305,n,Queue(the_queue,t)
exten = 305,n,Hangup()
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[asterisk-users] Polycom ring backs and CID

[J. Oquendo]

Hey all... Scenario

(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist

Now when the call goes back to the receptionist, how can I change either 
the ringer, the callerID or both?


   * TIA

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Re: [asterisk-users] .Call files do not seem to work

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 10:17:51AM -0500, Lee wrote:
 
 Hi,
 
 I was trying out call file just to see how they worked and my system 
 does not seem to do anything with them, although asterisk *is* deleting 
 the files that I put into /var/spool/asterisk/outgoing.
 
 1. I nano'd a quick call file like so:
 
 Channel: SIP/axVoice/910555
 CallerID : Leebo 55
 MaxRetries: 2
 RetryTime: 30
 WaitTime: 10
 Context: main_menu
 Extension: s
 Priority: 1
 
 2. And then mv'd to /var/spool/asterisk/outgoing
 
 As I mentioned, Asterisk appears to be grabbing the file, but there is 
 no call made.
 
 
 Q. Do calls originated like this show up in CLI output?

In the CLI:

sip show peer axVoice
show dialplan main_menu
set verbose 3


Then drop the call file

What is the CLI trace of the above?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Parsing Area Code from CallerID

[John French]

How would I parse the area code from this variable? Number=2515551212  
Sorry for the dense question, I don't seem to be able to find an 
appropriate function for parsing left to right.
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Re: [asterisk-users] IBM Server / USB Ports

[Matt]

Ok so that 'unknown' is infact the Digium card then?  I suspected that.

On 12/19/06, Time Bandit [EMAIL PROTECTED] wrote:

 So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' 
followed!!
I would try moving the Digium card to another slot. Your Ethernet
controlled must be onboard and it share its IRQ with the slot where
the Digium board is.

hth
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[asterisk-users] DTMF Tones A-B-C-D

[Al Bochter]

Ok does anyone know of any softphones that will dial DTMF tone keys A B 
C D

And do you know if Asterisk will take the DTMF Tones for A B C D

--
Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

-- For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBXt=email
-- Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdidt=email
-- Need Voice Mail?
http://www.bochterservices.com/?t=VMSt=email
--For new and used security items
http://www.bochterservices.com/?j=storet=email
--BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email

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Re: [asterisk-users] Remote Reboot of a Polycom

[Noah Miller]

 Does anyone know how to remotely reboot a PolyCom specifically 601
 phone?

sip notify polycom-check-cfg ipaddr

Or you might have to switch the polycom-check-cfg and the ip. I forget the 
order. You
also need to make sure that the phone has alwaysreboot=1 in the sip.cfg xml 
file.


You can also reboot by the sip device name (that you specified in sip.conf):

sip notify polycom-check-cfg sip-device-name

Note that you have to have a valid sip_notify.conf file.  The
default file from make samples has everything you need.

One caveat:  In the Polycom sip.cfg file, there is a setting:

specialEvent ... voIpProt.SIP.specialEvent.checkSync.alwaysReboot=1/

By default it is set to '0', which means the phone will only reboot on
a sip notify if the configuration files have changed.  I change this
to '1' so it will always reboot if I issue a sip notify command.


- Noah
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Re: [asterisk-users] IBM Server / USB Ports

[Gavin Hamill]

On Tue, 19 Dec 2006 10:35:32 -0500
Matt [EMAIL PROTECTED] wrote:

 Ok so that 'unknown' is infact the Digium card then?  I suspected
 that.

The Vendor ID is 'D161' which is supposed to look a bit like the first
four letters of 'Digium' :)

gdh
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Re: [asterisk-users] call from h323 to SIP

[nik600]

On 12/15/06, Thomas Kenyon [EMAIL PROTECTED] wrote:

nik600 wrote:
 Hi

 i am trying to do the same thing:
 receive a call from a cisco callmanager and forward it to a SIP user.

 Asterisk is compiled with h323 support, and is configured as a gateway
 in the cisco callmanager.

The incoming call is in the g.729 format, you should be able to fix this
in cisco call manager.

If not, make sure that the SIP target can accept a g.729 call.

I have resolved, it was a codec problem.

Enabling g711 on cisco callmanager has fixed the problem, many thanks.


Failing that buy a license for the codec.


 h323.conf:
 [general]
 port = 1720
 bindaddr = 193.x.x.x   ; this SHALL contain a single, valid IP
 address for this machine
 allow=all

 extension.conf:
 exten = 3298,1,Answer
 exten = 3298,2,Dial(SIP/[EMAIL PROTECTED])

 If a make a call to callamanager CISCO that forward to 3298 i read in
 asterisk console:

 Log:

 Verbosity is at least 20
-- Executing Answer(H323/ip$172.z.z.z:4836/14, ) in new stack
-- Executing Dial(H323/ip$172.z.z.z:4836/14,
 SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/[EMAIL PROTECTED] is ringing
 Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
 find a codec translation path from g729 to ulaw
 Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
 find a codec 
 ...
 translation path from g729 to slin
 Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
 find a codec translation path from g729 to ulaw
 Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to
 find a codec translation path from g729 to slin
 Dec 15 14:45:13 WARNING[19794]: translate.c:116
 ast_translator_build_path: No translator path from alaw to unknown
 Dec 15 14:45:13 WARNING[19794]: channel.c:1202 queue_frame_to_spies:
 Cannot build a path from g729 to slin
 Dec 15 14:45:13 WARNING[19794]: chan_h323.c:614 oh323_write: Asked to
 transmit frame type 64, while native formats is 256 (read/write =
 4/64)
 Dec 15 14:45:13 WARNING[19794]: chan_sip.c:2572 sip_write: Asked to
 transmit frame type 256, while native formats is 4 (read/write = 4/4)
 Dec 15 14:45:13 WARNING[19794]: translate.c:116
 ast_translator_build_path: No translator path from alaw to unknown
 Dec 15 14:45:13 WARNING[19794]: channel.c:2752
 ast_channel_make_compatible: No path to translate from
 H323/ip$172.z.z.z:4836/14(256) to SIP/193.x.x.x-40455d68(8)
 Dec 15 14:45:13 WARNING[19794]: app_dial.c:1602 dial_exec_full: Had to
 drop call because I couldn't make H323/ip$172.z.z.z:4836/14 compatible
 with SIP/193.x.x.x-40455d68
  == Spawn extension (default, 3298, 2) exited non-zero on
 'H323/ip$172.z.z.z:4836/14'

 Why? where am i wrong?
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Re: [asterisk-users] DTMF Tones A-B-C-D

[Zoa]

Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it).

Zoa


Al Bochter wrote:
Ok does anyone know of any softphones that will dial DTMF tone keys A 
B C D

And do you know if Asterisk will take the DTMF Tones for A B C D



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RE: [asterisk-users] Parsing Area Code from CallerID

[Jonathan k. Creasy]

${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end. 

-Jonathan

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of John French
 Sent: Tuesday, December 19, 2006 10:35 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Parsing Area Code from CallerID
 
 How would I parse the area code from this variable? Number=2515551212
 Sorry for the dense question, I don't seem to be able to find an
 appropriate function for parsing left to right.
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[asterisk-users] db.c: Unable to open Asterisk database

[DODO BUBU]

Dear asterisk users,
I am using Asterisk and I a m a new user.
Before it was working properly.
Since two days, users can not get registered : users registered timeout.
Those are the results of commands 
1. /var/log/asterisk#asterisk-rvv  
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk 1.2.1 currently running on ns1 (pid = 4244)
Verbosity is at least 46

2. var/log/asterisk# tail  -200  messages
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Database unavailable
Dec 19 09:45:59 WARNING[4245] chan_iax2.c: Unable to open IAX timing interface: 
No such file or directory
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:45:59 WARNING[4245] db.c: Database unavailable
Dec 19 09:45:59 WARNING[4245] db.c: Unable to open Asterisk database
Dec 19 09:54:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 09:54:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:02:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:02:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:02:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:10:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:10:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:10:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:19:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:19:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:19:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:27:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:27:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:27:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:35:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:35:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:35:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:44:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:44:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:44:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 10:52:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 10:52:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 10:52:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:00:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:00:59 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:00:59 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:09:19 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:09:19 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:09:19 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:17:39 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:17:39 WARNING[4249] res_musiconhold.c: Found no files in 
'/usr/share/asterisk/mohmp3'
Dec 19 11:17:39 WARNING[4249] res_musiconhold.c: Unable to spawn mp3player
Dec 19 11:25:59 NOTICE[4249] res_musiconhold.c: Request to schedule in the 
past?!?!
Dec 19 11:25:59 WARNING[4249] res_musiconhold.c: Found no files in 

Re: [asterisk-users] 2 devices using same sip account

[rilawich ango]

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue database showkey
SIP/Registry/sip account in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote:


19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:

 Hi,

 It seems that they both can make calls, but only one can receive
 call: the
 last registered...

 Greg

 Hi all,
   What will happen if 2 devices using the same set of sip account to
 connect to the same asterisk?  Do they both can make call?  Can they
 receive call as normal?
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In Asterisk, you should only have one phone per account. We do not
support
multiple devices per account. The PBX core needs to know how many
devices
that we are calling each time we access it.

/O
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Re: [asterisk-users] Need help getting started with asterisk

[Michael Sullivan]

Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all.  Is there some explicit thing I
need to put in to get the caller ID?

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Re: [asterisk-users] Re: IBM Server / USB Ports

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
 Could the unknown device be a management card?
 
 The newer dells have a management card built into the fist ethernet 
 controller.

D161:2400 is a Digium TDM2400P card.

Source: Debian has a nice 'update-pciids' command, which updates the
local pciids file from http://pciids.sourceforge.net/ . 
http://pci-ids.ucw.cz/iii/?p=d
http://pci-ids.ucw.cz/iii/?i=d161
http://pci-ids.ucw.cz/iii/?i=d1612400

-- 
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
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[asterisk-users] Re: AEL2 on Asterisk 1.2.4

[Steve Murphy]

On Tue, 2006-12-19 at 04:20:07 -0500, [EMAIL PROTECTED] wrote:
 
 Hey all,
 
 I am very interested in using AEL2 (don't want to upgrade to
 1.4 to get 
 it though), but am having some problems upgrading/patching my
 asterisk 
 system.  I am following the instructions on the wiki:
 
 http://www.voip-info.org/wiki/view/Asterisk
 +AEL2#AEL2AnnouncementsandNews
 
 But get the following error:
 
 
 'http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/diffs.AEL2.patch' 
 refers to a file, not a directory
 
 This refers to the process of including the patch as described
 in the 
 first portion of the wiki page.
 
 Am am still new to linux so the problem could be just me, but
 I believe 
 I followed the instructions.  They are pretty simple after
 all.
 
 BTW, I tried both ways described and I could not get either to
 work.
 
 Thanks for any help,
 
 -- 
 
 Warm Regards,
 
 Lee

Lee, everyone--

Sorry about that. I've created a patches subdir in the AEL2-1.2
repository, and put that patch down in there. So, now, you do:

svn co http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/patches

and then

patch -p0  patches/diffs.AEL2.patch

Assuming that you are in a 1.2 source directory...

Hopefully this sequence will work better. I've updated the voip-info
wiki.

murf

Again, you 1.2 users: It is not absolutely necessary to update your 1.2
installation to use AEL with 1.2; you can build a 1.4 somewhere, and use
the AEL in 1.4 to compile your extensions.ael into an extensions.conf
file via

aelparse -d -w

This will generate the file 'extensions.conf.aeldump', which you can
inspect and then copy into your appropriate /etc/asterisk directory on
the machine where your 1.2 installation resides.

The main thing to watch out for in this scenario, is that AEL uses a
slightly enhanced version of the $[] parser to do its thing. If you
don't use the new features, you should be quite OK.

murf


-- 
Steve Murphyaka 'codefreeze' or 'wyoming' on FreeNode IRC
Software Developer
Digium


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Re: [asterisk-users] 2 devices using same sip account

[Peder @ NetworkOblivion]

It doesn't have anything to do with hardphone versus softphone.  The 
issue is that it can only keep track of one registration per account. 
When the hardphone gets unplugged, it will not know about the softphone 
until it registers with asterisk.  It's initial registration was lost 
when the hardphone registered with the same info.


rilawich ango wrote:

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue database showkey
SIP/Registry/sip account in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote:


19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:

 Hi,

 It seems that they both can make calls, but only one can receive
 call: the
 last registered...

 Greg

 Hi all,
   What will happen if 2 devices using the same set of sip account to
 connect to the same asterisk?  Do they both can make call?  Can they
 receive call as normal?
 ___
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In Asterisk, you should only have one phone per account. We do not
support
multiple devices per account. The PBX core needs to know how many
devices
that we are calling each time we access it.

/O
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--

Network stuff you didn't know
http://www.networkoblivion.com

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Re: [asterisk-users] Parsing Area Code from CallerID

[Bruce Ferrell]

John French wrote:
How would I parse the area code from this variable? Number=2515551212  
Sorry for the dense question, I don't seem to be able to find an 
appropriate function for parsing left to right.

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NPA=${NUMBER:0:3}

--
One day at a time, one second if that's what it takes

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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

Do you, Gordon or Doug, happen to place international calls with 
early-dial enabled?  What kind of extensions.conf magic do you work to 
allow this?
I have been trying for some time to get this to work.  (My message from 
2006.11.03 regarding this is quoted just below)


On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to 
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1 
http://1.2.12.1

I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email



Gordon Henderson wrote:

On Sun, 5 Nov 2006, Doug Crompton wrote:

  

On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon

  

Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:



Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:

  

I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?


Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...

Gordon
  

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[asterisk-users] Is MOH Still Broken in Asterisk 1.4 (beta3)?

[Douglas Garstang]

I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, 
when a callee put a caller on hold, the musiconhold class that was played was 
not the one the callee wanted the caller to hear, but something else. Even 
after using mohsuggest in Asterisk 1.4, it still appears that this is not 
working correctly.

Here's the results of a simple test:

CASE CALLER CALLEE HOLDER HOLDER HEARS MOH
--
1325410132541023254101moh1
2325410132541023254102default

3325410232541013254102moh2
4325410232541013254101default

For each extension, I have mohsuggest set. Test cases 1 and 3, where the caller 
puts the callee on hold, yield the expected behaviour. However, test cases 2 
and 4 where the callee puts the caller on hold, do not yield the correct 
results.

Here's what the results SHOULD be.

CASE CALLER CALLEE HOLDER HOLDER HEARS MOH
--
1325410132541023254101moh1
2325410132541023254102moh2

3325410232541013254102moh2
4325410232541013254101moh1

Am I possibly doing something wrong with mohsuggest?

sip.conf:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[3254101]
type = friend 
context = CallStart
username = 3254101
accountcode = 3254101
qualify = yes
canreinvite = no
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang 3254101
secret = password
mohsuggest = moh1

[3254102]
type = friend 
context = CallStart
username = 3254102
accountcode = 3254102
qualify = yes
canreinvite = no
host = dynamic
dtmfmode = rfc2833
nat = no
callerid = Douglas Garstang 3254101
secret = password
mohsuggest = moh2
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Re: [asterisk-users] Polycom ring backs and CID

[Noah Miller]

Hi -


(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist

Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?


We'll need to see a little more info to help you out:

1) What mechanism are you using to transfer, built-in asterisk or the
Polycom transfer key(s)?
2) What does your dial plan look like - how is it that calls are
ringing back to your receptionist?

If you're looking for the technical aspects of how to do custom
ringtones, see here:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
(this page is for setting the phone to auto-answer, but changing
ringtones is the same procedure)

For setting the caller ID, see here:
http://www.voip-info.org/wiki/view/Setting+Callerid


- Noah
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Lee]

In the CLI:

sip show peer axVoice
show dialplan main_menu
set verbose 3


Then drop the call file

What is the CLI trace of the above?



Hi, thanks for responding.  Please see the output below.

Please note that moving a call file into /var/spool/asterisk/outgoing 
did not produce any CLI output.  The file was copied correctly, I 
believe and not present in the /outgoing directory when I checked with a 
simple ls command.


# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing



=== sip show peer axVoice ===
=
CLI

  * Name   : axVoice
  Secret   : Set
  MD5Secret: Not set
  Context  : incoming
  Subscr.Cont. : Not set
  Language :
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  FromUser : datatrak
  FromDomain   : 216.143.130.36
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : 555
  LastMsgsSent : -1
  Call limit   : 0
  Dynamic  : No
  Callerid :  
  Expire   : -1
  Insecure : port,invite
  Nat  : Always
  ACL  : No
  CanReinvite  : Yes
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   : 216.143.130.36
  Addr-IP : 216.143.130.36 Port 5060
  Defaddr-IP  : 216.143.130.36 Port 0
  Def. Username: set
  SIP Options  : (none)
  Codecs   : 0x4 (ulaw)
  Codec Order  : (ulaw)
  Status   : Unmonitored
  Useragent:
  Reg. Contact :

=== show dialplan main_after_hours ===
(I mistyped the name of the context in original post)

CLI show dialplan main_after_hours
[ Context 'main_after_hours' created by 'pbx_config' ]
  '1' =1. Playback(transfer) 
[pbx_config]
2. Macro(DialExtenVM|111|30|tm) 
[pbx_config]
3. Set(EXTEN=955) 
[pbx_config]
4. GoTo(Management|955|1) 
[pbx_config]
5. Playback(transfer) 
[pbx_config]
6. Macro(DialExtenVM|111|30|tr) 
[pbx_config]
7. Set(EXTEN=955) 
[pbx_config]
8. GoTo(Management|955|1) 
[pbx_config]
9. Playback(custom/no_tech_available) 
[pbx_config]
10. Voicemail(111) 
[pbx_config]
  '2' =1. 
Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
2. Goto(support_non_emergency|s|1) 
[pbx_config]
  '444' =  1. Set(LIMIT_PLAYAUDIO_CALLEE=yes) 
[pbx_config]
2. Dial(SIP/111|30|mgL(1:1:5000)) 
[pbx_config]
3. Wait(3) 
[pbx_config]
4. Goto(main_after_hours|s|1) 
[pbx_config]
  '9' =1. 
Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
2. Goto(main_branch|s|1) 
[pbx_config]
  'i' =1. GotoIf($[ ${FAIL_MENU} != ]|?2:3) 
[pbx_config]
2. Goto(${FAIL_MENU}|s|1) 
[pbx_config]
3. Goto(main_branch|s|1) 
[pbx_config]
  's' =1. Answer() 
[pbx_config]
2. Wait(1) 
[pbx_config]
3. Background(custom/after_hours) 
[pbx_config]
  't' =1. GotoIf($[ ${TIMEOUT_MENU} !=  ]|?2:3) 
[pbx_config]
2. Goto(${TIMEOUT_MENU}|s|1) 
[pbx_config]
3. Goto(main_branch|s|1) 
[pbx_config]
  '_ZZZ' = 1. 
Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m) 
[pbx_config]




--

Warm Regards,

Lee

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Re: [asterisk-users] 2 devices using same sip account

[Carla Schroder]

Your phones only register once, when they first start up. Seems to me that 
having multiple phones on the same account is asking for trouble- why not set 
up multiple accounts in the usual way, and create a ring group for all the 
phones you want to use? Like this example that rings two phones at the same 
time:

exten = 100,1,Dial(SIP/101SIP/102,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the 
same: one user with many phones, one extension, one voicemail box.

On Tuesday 19 December 2006 8:18 am, rilawich ango wrote:
 It seems that Greg is truth for the case.  Asterisk doesn't care how
 many devices register to the same account as it is a feature of sip
 protocol (please let me know if there is a method to restrict it).

 In my case, I use a soft phone an hard phone using the same sip
 account information to register to the same asterisk.  Soft phone
 register first and then hard phone register later.  I dial the number
 and hard phone ring.  Then I disconnect hard phone and expect soft
 phone will be ring after a couple of time.  However, soft phone didn't
 ring as the call is failed.  I issue database showkey
 SIP/Registry/sip account in CLI.  It displays the information which
 belongs to hard phone.  That's mean asterisk will keep the information
 of hard phone even it is disconnected with ignoring the soft phone
 registration.  Does asterisk can be set to refresh its registry in a
 couple of time to remove the old registry record?

 On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote:
  19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
   Hi,
  
   It seems that they both can make calls, but only one can receive
   call: the
   last registered...
  
   Greg
  
   Hi all,
 What will happen if 2 devices using the same set of sip account to
   connect to the same asterisk?  Do they both can make call?  Can they
   receive call as normal?
   ___
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  In Asterisk, you should only have one phone per account. We do not
  support
  multiple devices per account. The PBX core needs to know how many
  devices
  that we are calling each time we access it.
 
  /O

-- 
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/
best book for sysadmins and power users
~
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Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

how isit possible to get the VM there when one line is busy?

regards rene

On Tue, 19 Dec 2006 09:48:01 -0800
 Carla Schroder [EMAIL PROTECTED] wrote:
Your phones only register once, when they first start up. Seems to me that 
having multiple phones on the same account is asking for trouble- why not 
set 
up multiple accounts in the usual way, and create a ring group for all the 
phones you want to use? Like this example that rings two phones at the same 
time:


exten = 100,1,Dial(SIP/101SIP/102,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the 
same: one user with many phones, one extension, one voicemail box.


On Tuesday 19 December 2006 8:18 am, rilawich ango wrote:

It seems that Greg is truth for the case.  Asterisk doesn't care how
many devices register to the same account as it is a feature of sip
protocol (please let me know if there is a method to restrict it).

In my case, I use a soft phone an hard phone using the same sip
account information to register to the same asterisk.  Soft phone
register first and then hard phone register later.  I dial the number
and hard phone ring.  Then I disconnect hard phone and expect soft
phone will be ring after a couple of time.  However, soft phone didn't
ring as the call is failed.  I issue database showkey
SIP/Registry/sip account in CLI.  It displays the information which
belongs to hard phone.  That's mean asterisk will keep the information
of hard phone even it is disconnected with ignoring the soft phone
registration.  Does asterisk can be set to refresh its registry in a
couple of time to remove the old registry record?

On 12/19/06, Johansson Olle E [EMAIL PROTECTED] wrote:
 19 dec 2006 kl. 11.58 skrev Gregory Duchatelet:
  Hi,
 
  It seems that they both can make calls, but only one can receive
  call: the
  last registered...
 
  Greg
 
  Hi all,
What will happen if 2 devices using the same set of sip account to
  connect to the same asterisk?  Do they both can make call?  Can they
  receive call as normal?
  ___
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 In Asterisk, you should only have one phone per account. We do not
 support
 multiple devices per account. The PBX core needs to know how many
 devices
 that we are calling each time we access it.

 /O


--
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/

best book for sysadmins and power users
~
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Doug Crompton]

Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.

As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.

However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.

Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.

Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.

Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.

You just have to use it and fix things as they come up. I think I have
most all cases trapped now!

Doug



On Tue, 19 Dec 2006, Anthony Kepler wrote:

 Do you, Gordon or Doug, happen to place international calls with
 early-dial enabled?  What kind of extensions.conf magic do you work to
 allow this?
 I have been trying for some time to get this to work.  (My message from
 2006.11.03 regarding this is quoted just below)

  On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
  mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to
  place outgoing international calls from a
  GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1
  http://1.2.12.1
  I have the following extension line:
  exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
 
  When I attempt to place a call to a number in, for instance, Kenya, I
  dial 011254...etc.
  and I get this on the asterisk console:
  Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
 -- Called g1/0112
 
  It is attempting to dial out as soon as it receives a single digit to
  represent the .
  What I need is for it to wait a reasonable amount of time for additional
  digits.
  I have tried using set(TIMEOUT(digit)=5), and I see the following in the
  asterisk console:
 -- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
 -- Digit timeout set to 5
  However, this is printed far less than 5 seconds before the dial out
  attempt.
 
  I assume there must be something relatively obvious I'm missing here...
  if anyone can shed some light on this, it would be greatly appreciated.
 
 
  Thank you,
 - Anthony Kepler
  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email


 Gordon Henderson wrote:
  On Sun, 5 Nov 2006, Doug Crompton wrote:
 
 
  On the Budgetone 200 it is in the account tab settings of the web setup
  and it does work here with asterisk and my dialplans..
 
 
  On the GPX2000's it's via the web interface under each of the 4 Line
  configuration tabs. (so you'd have to set it on each account you were
  using on the phone)
 
  Gordon
 
 
  Doug
 
  On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:
 
 
  Hi,
 
  Where can I find that option?
 
  Thanks
  Jesus
 
  -Mensaje original-
  De: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] En nombre de Gordon
  Henderson
  Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
  Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
  Non-Commercial Discussion
  Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
 
  On Wed, 1 Nov 2006, Henry.L.Coleman wrote:
 
 
  I came to the same conclusion.
  There is one thing however that the GXP2000 needs in my opinion.
  There is no dial plan avaiable in the configuration, this means that when
  dialing a number there is a slight delay before it actually dials.
  With a dial plan the dialed number is sent immeadiately the pattern is
  match ed so it saves a second or two. Maybe they will fix this?
 
  Set the Early Dial option - it's on a per-line basis, then as soon
  as Asterisk gets a number it can dial, it will. No need to wait the 4
  seconds or press the send button...
 
  Gordon
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 




Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  

Re: [asterisk-users] Changing CALLERIDNUM on the fly

[Doug Crompton]

Thanks Anselm, That did it!

Doug

On Tue, 19 Dec 2006, Anselm Martin Hoffmeister wrote:

 Am Dienstag, den 19.12.2006, 01:11 -0500 schrieb Doug Crompton:
  Is what I am trying to do in this context possible. That is changing the
  incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not
  preceeded by a 1 I want to add a 1. Often calls come in without the
  preceeding 1 and this plays havoc with my redial if the 3 digit area
  code matches a local 3 digit extension. All my outside calls are 10 digits
  or 1+10 digits.
 
  Doug
 
 
  [from-pstn]
  exten = s,1,set(TEST=${CALLERIDNUM::1}) test for first digit1
  exten = s,2,GotoIf($[ ${TEST} = 1 ]?4:3)
  exten = s,3,set(__CALLERIDNUM=1${CALLERIDNUM})   if not add 1
  exten = s,4,noop(${CALLERIDNUM})   and this still displays without

 Replace CALLERIDNUM with CALLERID(num) on all occasions, and you will
 not need underscores because this is a special variable anyway.
 CALLERIDNUM is obsolete.

 You could get along with one line less:
 exten = s,1,GotoIf($[A${CALLERID(num):0:1} = A1]?3:2)
 exten = s,2,Set(CALLERID(num)=1${CALLERID(num)})
 exten = s,3,NOOP(Continue in Dialplan)

 Note that my GotoIf contains the two additional A letters which is
 important to avoid syntax errors if the CALLERID(num) is empty for
 whatever reason. I do not know what ends up in your CALLERID(num) if the
 number of the caller is not available (like anonymous or withheld) -
 anyway, with this statement it will end up being prepended by 1. You
 migth want to have a special case for that.

 If your phones happen to also display CALLERID(name) you can use this to
 lookup the phone number in a phone book (here in Germany there is an
 online service for number reverse lookup which works for about 50% of my
 callers) and set the variable.

 BR
 Anselm

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Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety.  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] Polycom ring backs and CID

[J. Oquendo]

Noah Miller wrote:

Hi -


(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist

Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?



If you're looking for the technical aspects of how to do custom
ringtones, see here:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
(this page is for setting the phone to auto-answer, but changing
ringtones is the same procedure)

For setting the caller ID, see here:
http://www.voip-info.org/wiki/view/Setting+Callerid

Not what I needed but thanks... I'm using the standard Asterisk 
transferring. I know there is a method to do so for parked calls:


exten = 7XX,1,SetVar(_ALERT_INFO=http://somewhere/alt.wav)
exten = 7XX,2,Set(CALLERID(name)=Parked Call)
exten = 7XX,n,ChanIsAvail(SIP/${EXTEN:1}|sj)
exten = 7XX,n,Dial(SIP/${EXTEN:1}|30)
exten = 7XX,n,Goto(default,${EXTEN},102)
exten = 7XX,102,Goto(main-aa,s,1)

I'm wondering if anyone has set it up differently


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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

I understand how early dial works (484 response and all that jazz), I 
also understand the NANP and how to keep my extensions from 
overlapping... but thank you for the tips.


My question was:  Do you place international calls from phones with 
early-dial enabled?
If so, might you be willing to share the relevant portions of your dial 
plan that are concerned with placing said international calls?


Thanks again,
   - Anthony Kepler
   [EMAIL PROTECTED] | SIP/Email

Doug Crompton wrote:

Early dial is a real nice feature BUT it requires that you carefully plan
and design your extensions. Each digit is accepeted by Asterisk and if a
match exists up to that point it will be accepted and dialed.

As an example, my internal extensions are 4xx and my internal special
extensions are 5xx. I chose those because they do not conflict with local
area codes or other first 3 digit sequences.

However if a call come in from, say, area code 512 (without the 1
prepended), and I have a local 512 extension, I would not be able to dial
that person back. It would instead go to the local 512, as this is
satisfied first.

Often callerID does not come in with the 1 before the area code. This is
what prompted me to put code in to append a 1 if none existed on the
incoming callerID. With the 1 appended there is no problem as 151 does not
match any local extension and I can use redial without problems.

Using 4 digit extensions would mostly eliminate this problem although you
still could not use 1xxx extensions.

Wildcard extension matches like X. or using the '.' anywhere in the
matches would not work.

You just have to use it and fix things as they come up. I think I have
most all cases trapped now!

Doug



On Tue, 19 Dec 2006, Anthony Kepler wrote:

  

Do you, Gordon or Doug, happen to place international calls with
early-dial enabled?  What kind of extensions.conf magic do you work to
allow this?
I have been trying for some time to get this to work.  (My message from
2006.11.03 regarding this is quoted just below)



On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email
  

Gordon Henderson wrote:


On Sun, 5 Nov 2006, Doug Crompton wrote:


  

On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon


  

Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:




Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:


  

I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?



Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...

Gordon

  

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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
 In the CLI:
 
 sip show peer axVoice
 show dialplan main_menu
 set verbose 3
 
 
 Then drop the call file
 
 What is the CLI trace of the above?
 
 
 Hi, thanks for responding.  Please see the output below.
 
 Please note that moving a call file into /var/spool/asterisk/outgoing 
 did not produce any CLI output.  The file was copied correctly, I 
 believe and not present in the /outgoing directory when I checked with a 
 simple ls command.
 
 # cp lee.call test.call
 # mv test.call /var/spool/asterisk/outgoing

Are both the current directory and /var/spool/asterisk/outgoing on the
same filesystem? If not, a 'mv' is implemented through a copy.

Anyway, you left out the CLI output of dropping trhe file.

Can Asterisk read that file? Write to it?

 
 
 
 === sip show peer axVoice ===
 =
 CLI
 
   * Name   : axVoice
   Secret   : Set
   MD5Secret: Not set
   Context  : incoming
   Subscr.Cont. : Not set
   Language :
   AMA flags: Unknown
   CallingPres  : Presentation Allowed, Not Screened
   FromUser : datatrak
   FromDomain   : 216.143.130.36
   Callgroup:
   Pickupgroup  :
   Mailbox  :
   VM Extension : 555
   LastMsgsSent : -1
   Call limit   : 0
   Dynamic  : No
   Callerid :  
   Expire   : -1
   Insecure : port,invite
   Nat  : Always
   ACL  : No
   CanReinvite  : Yes
   PromiscRedir : No
   User=Phone   : No
   Trust RPID   : No
   Send RPID: No
   DTMFmode : rfc2833
   LastMsg  : 0
   ToHost   : 216.143.130.36
   Addr-IP : 216.143.130.36 Port 5060
   Defaddr-IP  : 216.143.130.36 Port 0
   Def. Username: set
   SIP Options  : (none)
   Codecs   : 0x4 (ulaw)
   Codec Order  : (ulaw)
   Status   : Unmonitored
   Useragent:
   Reg. Contact :
 
 === show dialplan main_after_hours ===
 (I mistyped the name of the context in original post)
 
 CLI show dialplan main_after_hours
 [ Context 'main_after_hours' created by 'pbx_config' ]
   '1' =1. Playback(transfer) 
 [pbx_config]
 2. Macro(DialExtenVM|111|30|tm) 
 [pbx_config]
 3. Set(EXTEN=955) 
 [pbx_config]
 4. GoTo(Management|955|1) 
 [pbx_config]
 5. Playback(transfer) 
 [pbx_config]
 6. Macro(DialExtenVM|111|30|tr) 
 [pbx_config]
 7. Set(EXTEN=955) 
 [pbx_config]
 8. GoTo(Management|955|1) 
 [pbx_config]
 9. Playback(custom/no_tech_available) 
 [pbx_config]
 10. Voicemail(111) 
 [pbx_config]
   '2' =1. 
 Set(FAIL_MENU=main_after_hours|TIMEOUT_MENU=main_after_hours) [pbx_config]
 2. Goto(support_non_emergency|s|1) 
 [pbx_config]
   '444' =  1. Set(LIMIT_PLAYAUDIO_CALLEE=yes) 
 [pbx_config]
 2. Dial(SIP/111|30|mgL(1:1:5000)) 
 [pbx_config]
 3. Wait(3) 
 [pbx_config]
 4. Goto(main_after_hours|s|1) 
 [pbx_config]
   '9' =1. 
 Set(FAIL_MENU=main_branch|TIMEOUT_MENU=main_branch) [pbx_config]
 2. Goto(main_branch|s|1) 
 [pbx_config]
   'i' =1. GotoIf($[ ${FAIL_MENU} != ]|?2:3) 
 [pbx_config]
 2. Goto(${FAIL_MENU}|s|1) 
 [pbx_config]
 3. Goto(main_branch|s|1) 
 [pbx_config]
   's' =1. Answer() 
 [pbx_config]
 2. Wait(1) 
 [pbx_config]
 3. Background(custom/after_hours) 
 [pbx_config]
   't' =1. GotoIf($[ ${TIMEOUT_MENU} !=  ]|?2:3) 
 [pbx_config]
 2. Goto(${TIMEOUT_MENU}|s|1) 
 [pbx_config]
 3. Goto(main_branch|s|1) 
 [pbx_config]
   '_ZZZ' = 1. 
 Macro(DialExtenVM|${EXTEN}|${DEFAULT_RING_TIME}|${DEFAULT_CALLED_TRANS}${DEFAULT_CALLER_TRANS}m)
  
 [pbx_config]
 
 
 
 -- 
 
 Warm Regards,
 
 Lee
 
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-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Re: IBM Server / USB Ports

[Matt]

Ok.. so then for some reason the PCI slot that the Digium card is in
is following the IRQ of the Ethernet controller.  We will move the
Digium card and see what happens.

On 12/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

On Tue, Dec 19, 2006 at 09:32:00AM -0500, Steven wrote:
 Could the unknown device be a management card?

 The newer dells have a management card built into the fist ethernet 
controller.

D161:2400 is a Digium TDM2400P card.

Source: Debian has a nice 'update-pciids' command, which updates the
local pciids file from http://pciids.sourceforge.net/ .
http://pci-ids.ucw.cz/iii/?p=d
http://pci-ids.ucw.cz/iii/?i=d161
http://pci-ids.ucw.cz/iii/?i=d1612400

--
   Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] 2 devices using same sip account

[Noah Miller]

Hi Rene -


how isit possible to get the VM there when one line is busy?


If I understand your question correctly, the answer is you need two
incoming phone lines.


- Noah
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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Doug Crompton]

Sorry, I did not read the original message completely. The answer is no I
do not make international calls. I do not know anyone in any other country
to call! I do not have a rule for that but it should be easy to implement
as 01x would not match anything I currently have for early dial. Would you
always dial a 0 first for all international mumbers? Give me an example?

Are you outside the US? If so give me your number and I will try it!

Doug

On Tue, 19 Dec 2006, Anthony Kepler wrote:

 I understand how early dial works (484 response and all that jazz), I
 also understand the NANP and how to keep my extensions from
 overlapping... but thank you for the tips.

 My question was:  Do you place international calls from phones with
 early-dial enabled?
 If so, might you be willing to share the relevant portions of your dial
 plan that are concerned with placing said international calls?

 Thanks again,
 - Anthony Kepler
 [EMAIL PROTECTED] | SIP/Email


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RE: [asterisk-users] Parsing Area Code from CallerID

[Jonathan k. Creasy]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Bruce Ferrell
 Sent: Tuesday, December 19, 2006 12:00 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Parsing Area Code from CallerID
 
 
 John French wrote:
  How would I parse the area code from this variable?
Number=2515551212
  Sorry for the dense question, I don't seem to be able to find an
  appropriate function for parsing left to right.
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 NPA=${NUMBER:0:3}
 
 --
 One day at a time, one second if that's what it takes


That works if the number is always NPA-NXX-. If you end up with
+1NPANXX or 1NPANXX then you don't have the right data. 

-Jonathan

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Re: [asterisk-users] 2 devices using same sip account

[Carla Schroder]

Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:

http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
 how isit possible to get the VM there when one line is busy?

 regards rene

 On Tue, 19 Dec 2006 09:48:01 -0800

   Carla Schroder [EMAIL PROTECTED] wrote:
  Your phones only register once, when they first start up. Seems to me
  that having multiple phones on the same account is asking for trouble-
  why not set
  up multiple accounts in the usual way, and create a ring group for all
  the phones you want to use? Like this example that rings two phones at
  the same time:
 
  exten = 100,1,Dial(SIP/101SIP/102,30,t)
  exten = 100,2,VoiceMail([EMAIL PROTECTED])
 
  There are all kinds of fancy variations on this theme, but the idea is
  the same: one user with many phones, one extension, one voicemail box.

-- 
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/
best book for sysadmins and power users
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Lee]

Tzafrir Cohen wrote:

On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:

In the CLI:

sip show peer axVoice
show dialplan main_menu
set verbose 3


Then drop the call file

What is the CLI trace of the above?


Hi, thanks for responding.  Please see the output below.

Please note that moving a call file into /var/spool/asterisk/outgoing 
did not produce any CLI output.  The file was copied correctly, I 
believe and not present in the /outgoing directory when I checked with a 
simple ls command.


# cp lee.call test.call
# mv test.call /var/spool/asterisk/outgoing


Are both the current directory and /var/spool/asterisk/outgoing on the
same filesystem? If not, a 'mv' is implemented through a copy.

Anyway, you left out the CLI output of dropping trhe file.

Can Asterisk read that file? Write to it?



Hi,

As I mentioned above, the action of dropping a .call into the /outgoing 
directory did not produce any CLI output.  I did this through 2 putty 
sessions.  The first, we setup to watch the CLI output and the second 
was to use the commandline to move the .call into the /outgoing directory.


Asterisk must be doing *something* with the file (if just deleting it) 
because if I check the /outgoing directory after I move the file, there 
is no file there.  It's deleted.


--

Warm Regards,

Lee

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Re: [asterisk-users] STUN with one public and one private IP?

[David Thomas]

Are you kidding? Lighten up people!
Al made a friendly recommendation based on the comments regarding TrixBox.

Go have a beer... take a load off... enjoy the holidays.

Regards,
David
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Re: [asterisk-users] Re: AEL2 on Asterisk 1.2.4

[Lee]

Steve Murphy wrote:


Lee, everyone--

Sorry about that. I've created a patches subdir in the AEL2-1.2
repository, and put that patch down in there. So, now, you do:

svn co http://svn.digium.com/svn/asterisk/team/murf/AEL2-1.2/patches

and then


Excellent!  Thank you.  I will try this.


Again, you 1.2 users: It is not absolutely necessary to update your 1.2
installation to use AEL with 1.2; you can build a 1.4 somewhere, and use
the AEL in 1.4 to compile your extensions.ael into an extensions.conf
file via

aelparse -d -w

This will generate the file 'extensions.conf.aeldump', which you can
inspect and then copy into your appropriate /etc/asterisk directory on
the machine where your 1.2 installation resides.

The main thing to watch out for in this scenario, is that AEL uses a
slightly enhanced version of the $[] parser to do its thing. If you
don't use the new features, you should be quite OK.


Great idea.

Thanks again,

--

Warm Regards,

Lee

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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Anthony Kepler]

I am located on the west coast of the united states.
In order to dial an international number from within the US, we must 
first dial the special international access code that tells the PSTN 
the following call is an international one - in the US that is 011, 
followed by the country code, and then the actual number for our 
destination within that country.  (which would include whatever their 
concept of area code, prefix, and destination number are - which varies 
widely from country to country)


If you're generally interested in this, then you might find the 
following reading interesting as well:

http://en.wikipedia.org/wiki/North_American_Numbering_Plan
and
http://en.wikipedia.org/wiki/Area_code

  - Anthony Kepler
  [EMAIL PROTECTED] | SIP/Email

Doug Crompton wrote:

Sorry, I did not read the original message completely. The answer is no I
do not make international calls. I do not know anyone in any other 
country

to call! I do not have a rule for that but it should be easy to implement
as 01x would not match anything I currently have for early dial. Would 
you

always dial a 0 first for all international mumbers? Give me an example?

Are you outside the US? If so give me your number and I will try it!

Doug

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Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

nothing happen it only let ring all lines which are not in use but i want that 
the busy vm message is coming when one line is busy.


On Tue, 19 Dec 2006 10:55:34 -0800
 Carla Schroder [EMAIL PROTECTED] wrote:
Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:


http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:

how isit possible to get the VM there when one line is busy?

regards rene

On Tue, 19 Dec 2006 09:48:01 -0800

  Carla Schroder [EMAIL PROTECTED] wrote:
 Your phones only register once, when they first start up. Seems to me
 that having multiple phones on the same account is asking for trouble-
 why not set
 up multiple accounts in the usual way, and create a ring group for all
 the phones you want to use? Like this example that rings two phones at
 the same time:

 exten = 100,1,Dial(SIP/101SIP/102,30,t)
 exten = 100,2,VoiceMail([EMAIL PROTECTED])

 There are all kinds of fancy variations on this theme, but the idea is
 the same: one user with many phones, one extension, one voicemail box.


--
~
Carla Schroder
Linux geek and random computer tamer
check out my Linux Cookbook! 
http://www.oreilly.com/catalog/linuxckbk/

best book for sysadmins and power users
~
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Re: [asterisk-users] Polycom ring backs and CID

[Noah Miller]

Hi -


 (INTERNAL)
 1 Call comes in to receptionist and gets transferred to someone
 2 No one picks up that transfer
 3 Call goes back to receptionist

 Now when the call goes back to the receptionist, how can I change either
 the ringer, the callerID or both?

 If you're looking for the technical aspects of how to do custom
 ringtones, see here:
 http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
 (this page is for setting the phone to auto-answer, but changing
 ringtones is the same procedure)

 For setting the caller ID, see here:
 http://www.voip-info.org/wiki/view/Setting+Callerid

Not what I needed but thanks... I'm using the standard Asterisk
transferring. I know there is a method to do so for parked calls:


We'll still need to see more of your dialplan.  By your description,
it looks like the call is failing because the Dial() times out.

blindxfer and atxfer won't automatically return a caller to the
receptionist.  You have to have something in the dialplan to do that.
When we know what it is that is redirecting your failed transfers back
to the receptionist (probably the 't' extension), we can just insert a
Set(CALLERID=) or Set(_ALERT_INFO=).  You may also have
transfers fail because they get sent to an invalid extension.  The
calls go to the 'i' extension.  You can modify it accordingly, too.

- Noah
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Re: [asterisk-users] 2 devices using same sip account

[Aaron Daniel]

We use this function regularly (you should see my phone's
dialstring...).  If one phone responds that it's unavailable, the rest
of the phones will still ring through.  In the event that none of the
other phones are answered, the extension is considered unanswered, so
depending on how you program your dialplan, the call will go to the
unavailable voicemail.  If you watch the CLI in this situation, you'll
see Asterisk try all the devices in the group at the same time, and
it'll just bypass any devices that are unavailable.

Also, the problem with multiple phones registering with Asterisk at the
same name is that Asterisk only stores the information about the device
once, and is overwritten with each subsequent register.  If you have a
softphone and a hardphone both registered, whichever one has a faster
re-register rate will win out over the slower one.  The only way around
this is through the call groups, as several people have stated.

Aaron

On Tue, 2006-12-19 at 10:55 -0800, Carla Schroder wrote:
 Hmm, I don't know what happens when one of the lines is busy and none of the 
 lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
 then perhaps this is what you want:
 
 http://www.voip-info.org/wiki/view/Asterisk+tips+findme
 
 On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
  how isit possible to get the VM there when one line is busy?
 
  regards rene
 
  On Tue, 19 Dec 2006 09:48:01 -0800
 
Carla Schroder [EMAIL PROTECTED] wrote:
   Your phones only register once, when they first start up. Seems to me
   that having multiple phones on the same account is asking for trouble-
   why not set
   up multiple accounts in the usual way, and create a ring group for all
   the phones you want to use? Like this example that rings two phones at
   the same time:
  
   exten = 100,1,Dial(SIP/101SIP/102,30,t)
   exten = 100,2,VoiceMail([EMAIL PROTECTED])
  
   There are all kinds of fancy variations on this theme, but the idea is
   the same: one user with many phones, one extension, one voicemail box.

-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to work

[Doug Lytle]

Lee wrote:


As I mentioned above, the action of dropping a .call into the 
/outgoing directory did not produce any CLI output.  I did this 
through 2 putty sessions.  The first, we setup to watch the CLI output 
and the second was to use the commandline to move the .call into the 
/outgoing directory.


set verbose 50 and try again.

Doug

--

Ben Franklin quote:

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


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Re: [asterisk-users] 2 devices using same sip account

[René Enskat]

yes thats we i use 100,a,b,c etc.
do you can mail me your extensions how you do the dials and the vm?

regards rene

On Tue, 19 Dec 2006 13:15:53 -0600
 Aaron Daniel [EMAIL PROTECTED] wrote:

We use this function regularly (you should see my phone's
dialstring...).  If one phone responds that it's unavailable, the rest
of the phones will still ring through.  In the event that none of the
other phones are answered, the extension is considered unanswered, so
depending on how you program your dialplan, the call will go to the
unavailable voicemail.  If you watch the CLI in this situation, you'll
see Asterisk try all the devices in the group at the same time, and
it'll just bypass any devices that are unavailable.

Also, the problem with multiple phones registering with Asterisk at the
same name is that Asterisk only stores the information about the device
once, and is overwritten with each subsequent register.  If you have a
softphone and a hardphone both registered, whichever one has a faster
re-register rate will win out over the slower one.  The only way around
this is through the call groups, as several people have stated.

Aaron

On Tue, 2006-12-19 at 10:55 -0800, Carla Schroder wrote:
Hmm, I don't know what happens when one of the lines is busy and none of the 
lines get answered. It's easy enough to test. If it doesn't go to voicemail, 
then perhaps this is what you want:


http://www.voip-info.org/wiki/view/Asterisk+tips+findme

On Tuesday 19 December 2006 9:58 am, René Enskat wrote:
 how isit possible to get the VM there when one line is busy?

 regards rene

 On Tue, 19 Dec 2006 09:48:01 -0800

   Carla Schroder [EMAIL PROTECTED] wrote:
  Your phones only register once, when they first start up. Seems to me
  that having multiple phones on the same account is asking for trouble-
  why not set
  up multiple accounts in the usual way, and create a ring group for all
  the phones you want to use? Like this example that rings two phones at
  the same time:
 
  exten = 100,1,Dial(SIP/101SIP/102,30,t)
  exten = 100,2,VoiceMail([EMAIL PROTECTED])
 
  There are all kinds of fancy variations on this theme, but the idea is
  the same: one user with many phones, one extension, one voicemail box.


--
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] DTMF Tones A-B-C-D

[Al Bochter]

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF tones

Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but first 
I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

-- For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBXt=email
-- Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdidt=email
-- Need Voice Mail?
http://www.bochterservices.com/?t=VMSt=email
--For new and used security items
http://www.bochterservices.com/?j=storet=email
--BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
A B C D

And do you know if Asterisk will take the DTMF Tones for A B C D



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RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Colin Anderson]

If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document 
File Format  Unix Format. 

I ran into this same problem, and it turns out my Asterisk install would not
use Windows-formatted text files, it would just ignore them and delete them.


-Original Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
work


Tzafrir Cohen wrote:
 On Tue, Dec 19, 2006 at 12:44:39PM -0500, Lee wrote:
 In the CLI:

 sip show peer axVoice
 show dialplan main_menu
 set verbose 3


 Then drop the call file

 What is the CLI trace of the above?

 Hi, thanks for responding.  Please see the output below.

 Please note that moving a call file into /var/spool/asterisk/outgoing 
 did not produce any CLI output.  The file was copied correctly, I 
 believe and not present in the /outgoing directory when I checked with a 
 simple ls command.

 # cp lee.call test.call
 # mv test.call /var/spool/asterisk/outgoing
 
 Are both the current directory and /var/spool/asterisk/outgoing on the
 same filesystem? If not, a 'mv' is implemented through a copy.
 
 Anyway, you left out the CLI output of dropping trhe file.
 
 Can Asterisk read that file? Write to it?
 

Hi,

As I mentioned above, the action of dropping a .call into the /outgoing 
directory did not produce any CLI output.  I did this through 2 putty 
sessions.  The first, we setup to watch the CLI output and the second 
was to use the commandline to move the .call into the /outgoing directory.

Asterisk must be doing *something* with the file (if just deleting it) 
because if I check the /outgoing directory after I move the file, there 
is no file there.  It's deleted.

-- 

Warm Regards,

Lee

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Re: [asterisk-users] Polycom ring backs and CID

[J. Oquendo]

Noah Miller wrote:

Hi -
We'll still need to see more of your dialplan.  By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate... 
Call comes in receptionist answers. For some ungodly reason this client 
does not want voicemail, so when a call is xferred, the call goes 
through fine, if no one answers it rings back to the receptionist 
*SUCCESSFULLY*. However, what the client is complaining about is, it 
sounds idiotic to repeat the company mantra Thank you for calling 
Foobar Co. how can I xfer your call to a caller they just answered but 
failed to be xferred successfully. Before someone asks why identify the 
caller ID this customer also (for some ungodly reason) only wants his 
CID showing up in and out. (Don't ask)


So again:

Call comes in -- Receptionist (How can I direct your call)
Receptionist -- Transfers to extension
Extension -- No answer -- Back to receptionist
Receptionist (same call) -- Thank you for calling Foobar

Easier to comprehend?

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Re: [asterisk-users] AGI Help Please

[Jay Milk]

Does the script run from command-line?  Without taking a close look at 
this, the include statements in the function body of connect_db look 
potentially messy.


Also, any output to stdout is interpreted by asterisk as a command, so 
those fputs statements would be a problem -- do

fputs($stdout,VERBOSE \There have been\\n);
fputs($stdout,VERBOSE \$row_count calls made\\n);

instead.

William Piper wrote:

List,
 
I finally decided to break down  start playing with AGI scripts, but 
for the life of me, I can't figure out what I am doing wrong.
 
Below is a super simple script to run a query in mysql to see how many 
call records there are for the extension calling in, then print the 
total in the CLI.
 
This is all I get on the CLI:

-- Executing AGI(SIP/216-0baa, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
-- AGI Script test.php completed, returning 0
-- Executing Hangup(SIP/216-0baa, ) in new stack
 
 
Here is the script:

#!/usr/bin/php -q
?php
ob_implicit_flush(false);
set_time_limit(6);
$stdin  = fopen(php://stdin,r);
$stdout = fopen('php://stdout', 'w');

function read() {
  global $stdin, $debug;
  $input = str_replace(\n, , fgets($stdin, 4096));
  return $input;
}
function connect_db() {
$database=asteriskcdrdb;
 include(./common.php);
 include(./dbconnect.php);
 }

// parse agi headers into array
while ($env=read()) {
  $env = str_replace(\,,$env);
  $s = split(: ,$env);
  $agi[str_replace(agi_,,$s[0])] = trim($s[1]);
 if (($env == ) || ($env == \n)) {
 break;
  }
}

// main program
$clid = $agi[callerid];
connect_db();

$query1 = SELECT * FROM cdr WHERE dst = '$clid' ;
$query_result1 = @mysql_query($query1);
$row_count = mysql_num_rows($query_result1);
$row1 = @mysql_fetch_array ($query_result1);

fputs($stdout,There have been\n);
fputs($stdout,$row_count calls made\n);

fflush($stdout);
fclose($stdin);
fclose($stdout);
exit;
?

There are no debug errors and the query is going through just fine... 
and yes, I chmod 755.

Does anyone have a clue what I am doing wrong?

Thanks,


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Re: [asterisk-users] DTMF Tones A-B-C-D

[Bob Chiodini]

The free version 1.31 has all 16 keys on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF 
tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but 
first I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

-- For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBXt=email
-- Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdidt=email
-- Need Voice Mail?
http://www.bochterservices.com/?t=VMSt=email
--For new and used security items
http://www.bochterservices.com/?j=storet=email
--BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
A B C D

And do you know if Asterisk will take the DTMF Tones for A B C D



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Re: [asterisk-users] DTMF Tones A-B-C-D

[Zoa]

Hmm, if the latest free version does not have all 16 keys, email 
[EMAIL PROTECTED], there should not be a difference in the amount 
of DTMF keys between biz and free version.


Zoa

Bob Chiodini wrote:

The free version 1.31 has all 16 keys on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 
DTMF tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but 
first I must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

-- For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBXt=email
-- Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdidt=email
-- Need Voice Mail?
http://www.bochterservices.com/?t=VMSt=email
--For new and used security items
http://www.bochterservices.com/?j=storet=email
--BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will 
accept it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys 
A B C D

And do you know if Asterisk will take the DTMF Tones for A B C D



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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Maxim Veksler]

On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
 Hi list,

 It's been a while since I've done asterisk stuff, and I'm wondering if
 there any news in the field.

 What do you people use today for http management of debian based Asterisk
 setup?
 Preferably something with the proven .deb extension.

destar has a tar extension, but a prefix quite similar to Debian.
Availble in Etch.

FreePBX is not yet availble, though a Sarge-based CD that includes it
and generally works could be downloaded from
http://updates.xorcom.com/iso/rapid-current.iso and the package is
availble from

  deb http://updates.xorcom.com/rapid future main



Hi Tzafrir,
Thank you!

I am aware of both of these tools, I don't like them!
They make absolute changes in your /etc/asterisk/* files, they assume
that they are the only thing you will be using for managing your
asterisk pbx and they are both totally unfriendly to 3rd party
changes.

Are there any other tools then, perhaps some that has not been debianized yet?
I'd like something that could be more cooperative with user hand made changes.

Maybe not web based GUI then?

Thanks,
Maxim.


--
   Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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--
Cheers,
Maxim Veksler

Free as in Freedom - Do u GNU ?
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Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Lee]

Colin Anderson wrote:

If you are using Windows to generate the .call files, make sure they are in
Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use
Crimson Editor www.crimsoneditor.com to make the file, and click Document 
File Format  Unix Format. 


I ran into this same problem, and it turns out my Asterisk install would not
use Windows-formatted text files, it would just ignore them and delete them.




Hi Colin,

Thanks for responding.  I've run into the problem elsewhere myself. 
Alas, I wrote the call file using nano on the linux box through ssh/putty.


--

Warm Regards,

Lee

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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Michiel van Baak]

On 23:33, Tue 19 Dec 06, Maxim Veksler wrote:
 Are there any other tools then, perhaps some that has not been debianized 
 yet?
 I'd like something that could be more cooperative with user hand made 
 changes.

Thanks to the folks at SpeakUp (http://www.speakup.nl) I
have a nice webtool that puts everything in seperate files
you can #include in your normal asterisk configs.
We use it on one of our PBX boxen next to several custom
made stuff.

Speakup granted me the rights to release the version in GPL.
I've been too busy to prepare a release but with holidays
coming etc it will be available somewhere Q1 2007 (I hope)

As soon as I have something I'll let the list know...

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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[asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

Anyone know if there's a way to match a dialplan extension, execute some code, 
say set a variable, and then continue with the dialplan?

I want to set a variable when the dialplan flows beyond a certain context. This 
would be a great feature.

Doug.
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[asterisk-users] No music on hold?

[Phil Finkler]

Hi all,

 

I've got Asterisk 1.2.10 up and running on Debian using the back ports.
I noticed that it didn't come with mpg123 or depend on it and I believe
I read somewhere that asterisk now handles it's own mp3 playback?  Is
this true?  If so I must have a problem, because I hear no music when
putting someone on hold.  When looking at the console when putting
someone on hold, I see the following:

 

-- Started music on hold, class 'default', on channel
'IAX2/voicepulse01-3'

-- Stopped music on hold on IAX2/voicepulse01-3

 

It says music starts and then it instantly stops.  Any ideas?

 

Thanks,

Phil 

 

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Re: [asterisk-users] Match a Numer - then continue with dialplan

[David Thomas]

On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote:

Anyone know if there's a way to match a dialplan extension, execute some code, 
say set a variable, and then continue with the dialplan?

I want to set a variable when the dialplan flows beyond a certain context. This 
would be a great feature.

Doug.


Have you tried using the SetVar cmd? I haven't tried it but it sounds
like it might work for this.

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

Regards,
David
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Re: [asterisk-users] DTMF Tones A-B-C-D

[Steve Edwards]

I downloaded the free 1.37 version. The slide out keypad displays ABCD 
buttons but they do not respond to clicks. You can enter ABCD using your 
keyboard and they will be sent to Asterisk.


There appears to be some funkyness with case though.

If you enter  and click dial,  is sent to Asterisk.

If you then enter A, Aaaa is displayed, matching the previous entry 
ignoring case. If you continue typing AAA and click dial, Aaaa is sent 
to Asterisk.


On Tue, 19 Dec 2006, Zoa wrote:



Hmm, if the latest free version does not have all 16 keys, email 
[EMAIL PROTECTED], there should not be a difference in the amount of 
DTMF keys between biz and free version.


Zoa

Bob Chiodini wrote:

The free version 1.31 has all 16 keys on the keypad.

Bob...

Al Bochter wrote:

Are you sure IdsFisk will do all 16 DTMF tones?
I have the free ver of IdeFisk and that one does only the base 12 DTMF 
tones


Base 12 DTMF are
1 2 3 4 5 6 7 8 9 0 # *

The 16 DTMF are
1 2 3 4 5 6 7 8 9 0 # * A B C D

If the paid ver of IdeFisk has that may have to pay the money but first I 
must know for sure. :-)

I want to use A B C D for control IVR's
Not Everyone knows about the 16 tones like the Hams do 8-)

Best regards,

Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email

(VoIP PBX) 1-563-773-6610 EXT: 250

-- For Information on PBX Systems for SOHO
http://www.bochterservices.com/?j=PBXt=email
-- Need A Toll Free Number?
http://www.bochterservices.com/?t=TFdidt=email
-- Need Voice Mail?
http://www.bochterservices.com/?t=VMSt=email
--For new and used security items
http://www.bochterservices.com/?j=storet=email
--BUY Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email



Zoa wrote:



Idefisk will do that - www.asteriskguru.com . (And asterisk will accept 
it).


Zoa


Al Bochter wrote:

Ok does anyone know of any softphones that will dial DTMF tone keys A B 
C D

And do you know if Asterisk will take the DTMF Tones for A B C D



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Inbound (clean). Database: 0660-0, 12/19/2006 - 12/19/2006 11:16:05 AM





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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] T1 Pri Question

[Rob Schall]

When setting up the zaptel, etc Is it necessary to have a seperate
group for incoming and one for outgoing calls? Either way, does asterisk
always know which channels are open, and does it always clear up a
channel for use once a call completes?

Reason for asking... After dialing into our system over a few DID
numbers, I noticed you can only call 2-3 times before getting a busy
signal. However we have a full 24 channel PRI. During this time, you are
more than able to make outgoing calls over that same PRI. After hanging
up on the incoming call, (outside into the PRI), it can take upto a few
minutes to clear up for the next person to call in.

Any thoughts,
Rob

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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

I just know someone is going to ask 'why would you ever want to do that?'. 
Here's my answer.

We have two companies, each with a dialplan similar to what's below. In the 
event that the number being dialled does not match any number within our OWN 
company, we want to set the caller id to be a generic one for the company, NOT 
one for the user. This is a pretty normal requirement that most companies want. 
So, in the event that the logic flows beyond coo1_OnNet, we want to reset the 
caller id of say, 3254001 Doug, to 3254000 Widgets Inc. If there was a way 
to match against a number in the dialplan, and then continue execution after 
that point, we could put this statement at the end of the coo1_OnNet context 
and it would all be sweet. Without that, I don't have a clue how to do this... 
unless we stick with out current 3,000 line python script.

[coo1_CallStart]
include = coo1_OnNet
include = syst_OnNet
include = syst_OffNet

[coo1_OnNet]

exten = 3254101,1,Dial(SIP/3254101,20,tr)
exten = 3254102,1,Dial(SIP/3254102,20,tr)
exten = 3254103,1,Dial(SIP/3254103,20,tr)

exten = 1000,1,Answer
exten = 1000,2,Wait(1)
exten = 1000,3,NoOp(${FOO})

[syst_OnNet]
include = coo1_OnNet
include = coo2_OnNet

[syst_OffNet]
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],180,tr)



~  


 -Original Message-
 From: Douglas Garstang 
 Sent: Tuesday, December 19, 2006 2:46 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Match a Numer - then continue with dialplan
 
 
 Anyone know if there's a way to match a dialplan extension, 
 execute some code, say set a variable, and then continue with 
 the dialplan?
 
 I want to set a variable when the dialplan flows beyond a 
 certain context. This would be a great feature.
 
 Doug.
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RE: [asterisk-users] No music on hold?

[Kevin Trumbull]

I had this same problem.  I also read that mpg123 was not required, but it 
actually is if you wish to use mp3 files.  I just decided to go with RAW files 
because I had problems converting some mp3's to the appropriate bit rate.

http://www.voip-info.org/wiki/index.php?page=Asterisk+mpg123+faking+it

--
Kevin Trumbull

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Finkler
Sent: Tuesday, December 19, 2006 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No music on hold?


Hi all,

I've got Asterisk 1.2.10 up and running on Debian using the back ports.  I 
noticed that it didn't come with mpg123 or depend on it and I believe I read 
somewhere that asterisk now handles it's own mp3 playback?  Is this true?  If 
so I must have a problem, because I hear no music when putting someone on hold. 
 When looking at the console when putting someone on hold, I see the following:

-- Started music on hold, class 'default', on channel 'IAX2/voicepulse01-3'
-- Stopped music on hold on IAX2/voicepulse01-3

It says music starts and then it instantly stops.  Any ideas?

Thanks,
Phil
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RE: [asterisk-users] Match a Numer - then continue with dialplan

[Douglas Garstang]

 -Original Message-
 From: David Thomas [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, December 19, 2006 3:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Match a Numer - then continue with
 dialplan
 
 
 On 12/19/06, Douglas Garstang [EMAIL PROTECTED] wrote:
  Anyone know if there's a way to match a dialplan extension, 
 execute some code, say set a variable, and then continue with 
 the dialplan?
 
  I want to set a variable when the dialplan flows beyond a 
 certain context. This would be a great feature.
 
  Doug.
 
 Have you tried using the SetVar cmd? I haven't tried it but it sounds
 like it might work for this.
 
 http://www.voip-info.org/wiki/view/Asterisk+variables
 
 Regards,
 David

David,

If I call setvar, my variable will be set, but dialplan processing will stop...
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RE: [asterisk-users] T1 Pri Question

[Colin Anderson]

Short answer: a single group should be fine. Long answer: It depends. 

Your Dial() command determines the order in which Asterisk plucks channels
from your PRI. Most north american system call inbound channel 1 first, then
2, etc. It makes sense, then for you to take channels from the topmost
first, for outbound calls. This is dictacted by how you format your Dial()
command: 

g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group). 
G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group). 
r: use a round-robin search, starting at the next highest channel than last
time (aka. ascending rotary hunt group). 
R: use a round-robin search, starting at the next lowest channel than last
time (aka. descending rotary hunt group). 

(from voip-info.org)

So, a dial command like:

exten = 1,Dial(Zap/g0/5551212)

would take the channel from the bottom of your group. 

This may have bearing on your situation. However, from what you are
descibing, this seems to be a symptom of a larger problem, that of Asterisk
not correctly hanging up the zap channel and that has nothing to do with
channel selection preferences. Anything weird on your CLI during this
period? 

-Original Message-
From: Rob Schall [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] T1 Pri Question


When setting up the zaptel, etc Is it necessary to have a seperate
group for incoming and one for outgoing calls? Either way, does asterisk
always know which channels are open, and does it always clear up a
channel for use once a call completes?

Reason for asking... After dialing into our system over a few DID
numbers, I noticed you can only call 2-3 times before getting a busy
signal. However we have a full 24 channel PRI. During this time, you are
more than able to make outgoing calls over that same PRI. After hanging
up on the incoming call, (outside into the PRI), it can take upto a few
minutes to clear up for the next person to call in.

Any thoughts,
Rob

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Re: [asterisk-users] Need help getting started with asterisk

[Time Bandit]

Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all.  Is there some explicit thing I
need to put in to get the caller ID?

callerid=asreceived
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RE: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to wo rk

[Colin Anderson]

The only other thing that comes to mind is that .call files are very
sensitive to whitespace; you may have unintentially padded the .call file
with whitespace or tabs that it does not like. 

The attached .call file works on my 1.0.9 server. Maybe it can give you some
insight. 

-Original Message-
From: Lee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 19, 2006 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: SPAM-LOW: Re: [asterisk-users] .Call files do not seem to
wo rk


Colin Anderson wrote:
 If you are using Windows to generate the .call files, make sure they are
in
 Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files.
Use
 Crimson Editor www.crimsoneditor.com to make the file, and click Document

 File Format  Unix Format. 
 
 I ran into this same problem, and it turns out my Asterisk install would
not
 use Windows-formatted text files, it would just ignore them and delete
them.
 
 

Hi Colin,

Thanks for responding.  I've run into the problem elsewhere myself. 
Alas, I wrote the call file using nano on the linux box through ssh/putty.

-- 

Warm Regards,

Lee

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614.call
Description: Binary data
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Re: [asterisk-users] No music on hold?

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 04:49:36PM -0500, Phil Finkler wrote:
 Hi all,
 
  
 
 I've got Asterisk 1.2.10 up and running on Debian using the back ports.

Debian does not include the default MoH files that come with Debian for
legal reasons. Get some sound files in the moh directory, basically, and
use the naitve moh.

Grab
http://updates.xorcom.com/rapid/pool/main/f/freepbx/asterisk-sounds-moh-freepbx_2.1.3.dfsg-1.2902_all.deb
 
or an equivalent.

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

[Michael Sullivan]

OK.  My basic asterisk install seems to be working.  I can get caller
ID.  My dialplan says:

[incoming]
; incoming calls from the FXO port are directed to this context from
zapata.conf
exten = s/9185415897,1,Set(CALLERID(name)=Michael Sullivan)
exten = s/9185415897,1,HANGUP(1)
exten = s,1,Set(CALLERID(name)=Someone Else)

This is for testing.  It's supposed to check the caller ID to see who is
calling, and if it's my cell phone, hang up, but let any other number
through to ring on our handsets.  Instead it rewrites the caller ID name
to Michael Sullivan and allows the call to pass through.  Here is the
output of the call to the CLI:

-- Starting simple switch on 'Zap/1-1'
-- Executing Set(Zap/1-1, CALLERID(name)=Michael Sullivan) in
new stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Dec 19 16:51:37 NOTICE[7413]: chan_zap.c:6073 ss_thread: Got event 18
(Ring Begin)...
-- Executing Set(Zap/1-1, CALLERID(name)=Someone Else) in new
stack
  == Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'

In fact, the hang up on my number clause isn't even shown in the
dialplan in the CLI:

camille*CLI show dialplan incoming
[ Context 'incoming' created by 'pbx_config' ]
  's' =1. Set(CALLERID(name)=Michael Sullivan)
[pbx_config]
  's' =1. Set(CALLERID(name)=Someone Else)
[pbx_config]
camille*CLI 
-= 2 extensions (2 priorities) in 1 context. =-
camille*CLI 


What am I doing wrong?





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Re: [asterisk-users] What web interfaces are available today for debian based Asterisk installation?

[Tzafrir Cohen]

On Tue, Dec 19, 2006 at 11:33:55PM +0200, Maxim Veksler wrote:
 On 12/17/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 On Sun, Dec 17, 2006 at 07:07:07PM +0200, Maxim Vexler wrote:
  Hi list,
 
  It's been a while since I've done asterisk stuff, and I'm wondering if
  there any news in the field.
 
  What do you people use today for http management of debian based Asterisk
  setup?
  Preferably something with the proven .deb extension.
 
 destar has a tar extension, but a prefix quite similar to Debian.
 Availble in Etch.
 
 FreePBX is not yet availble, though a Sarge-based CD that includes it
 and generally works could be downloaded from
 http://updates.xorcom.com/iso/rapid-current.iso and the package is
 availble from
 
   deb http://updates.xorcom.com/rapid future main
 
 
 Hi Tzafrir,
 Thank you!
 
 I am aware of both of these tools, I don't like them!
 They make absolute changes in your /etc/asterisk/* files, they assume
 that they are the only thing you will be using for managing your
 asterisk pbx and they are both totally unfriendly to 3rd party
 changes.
 
 Are there any other tools then, perhaps some that has not been debianized 
 yet?
 I'd like something that could be more cooperative with user hand made 
 changes.

You refer to direct addition of content to the cnfig files. But this is
not the only way to affect the output.

If you consider those interfaces to be code generators, you can
manipulate their inputs (the mysql database in the case of FreePBX, the
set of pthon ojects for DeStar) or the generator itself (patch the
code).

Sometimes patching the code is trivial (e.g: disabling a line from being
generaed). Sometimes it's more complicated.


I'm quite pessimistic regarding tools that are supposed to tolerate any
sort of manual override with Asterisk's configuration. At least tools
that are aimed to give a simplified/more powerful user interface rather
than a glorified text editor with macros.

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

[Carlos Rojas]

a2billing

Is very good

On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote:




2006/12/19, C F [EMAIL PROTECTED]:

 Can anyone recommend a call accounting solution with rating for post
 paid billing that works well with asterisk using the account code or
 any other info from the CDR?

 I don't want the billing software to any phone calls for me, therefore
 any solution that modifies my extensions.conf is out, nor does it have
 to allow for customers the ability to log in to check their
 usage/balances.
 I have looked at astbill but it looks to be way overcomplicated for
 what I want, as well as it requires realtime.
 Thank you
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Mor and Mcc

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Re: [asterisk-users] 2 devices using same sip account

[Gordon Henderson]

On Tue, 19 Dec 2006, Carla Schroder wrote:


Your phones only register once, when they first start up. Seems to me that
having multiple phones on the same account is asking for trouble- why not set
up multiple accounts in the usual way, and create a ring group for all the
phones you want to use? Like this example that rings two phones at the same
time:

exten = 100,1,Dial(SIP/101SIP/102,30,t)
exten = 100,2,VoiceMail([EMAIL PROTECTED])

There are all kinds of fancy variations on this theme, but the idea is the
same: one user with many phones, one extension, one voicemail box.


I've been setting up a few systems recently with a SIP account and an IAX 
account (same passwords, CLI, etc.) and having the users use a SIP 
hardphone for the office desk, and an IAX (idefisk) softphone for 
out-of-office calls. (My Dial() calls both accounts, so both phones ring)


It saves hassles with NAT, etc. for remote SIP phones too.

No good if you only have SIP phones though!

Gordon
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Re: [asterisk-users] AGI Help Please

[William Piper]

I am running console. I'm a newbie for AGI's but not that new.

Thanks,

bp


On 12/19/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:


On Tue, Dec 19, 2006 at 09:35:56AM +0200, yusuf wrote:

 to see debug output for AGI's, you *must* be connected to the first Ast
 terminal.  So start Asterisk like 'asterisk -cvv', then you
 will see output from your AGI.

Actually, if you have not started Asterisk in a console, it will be sent
to the first remote client. Thus you won't exit Asterisk if you
accidentally press Ctrl-C, as you're used to ;-)

--
  Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] NAT and Dial to two channels at once

[Brad Templeton]

On Mon, Dec 11, 2006 at 06:48:18PM -0500, Andrew Joakimsen wrote:
 You need to understand how NAT works, if you can chan2 and chan2 is behind a
 NAT and suddenly someone else is invited to chan2's IP address port 5060
 chan2's router willl say WTF I dont have an estabished connection on port
 5060 (to the client being reinvited to chan2) and it wont work. You need to
 have the media path go through asterisk in that case.

Actually, it's more complex than that.

If the NAT box has had a hole poked (in its config) for the RTP port (SIP
port is only used by Asterisk) then any machine can send it RTP on that
port.

In addition, if the NAT is of the full cone type, any host can send to
your port once you have sent a packet out that port.

With Restricted cone and Port restricted cones, it also works as long as
the Natted IP phone is sending packets out to the other host already.
Which it should be if we have symmetric RTP.

Symmetric NATs, which are rare, will change the port number when they
start talking to a different host for RTP.  This will screw up all but
the cleverest implementations.  (Though there are endpoints that notice
if the RTP is coming from a port other than they were told, and start
sending to that instead of the one in the SDP)

What doesn't work is assymetric RTP with NAT.   In this case we have
the audio going through asterisk in one direction, and directly in
the other direction.  That will fail if the direct direction tries
to go into a nat (it should work if it's only leaving a nat)

Asterisk currently does assymetric RTP if it thinks it only has to
listen to one end of the audio path.  That's a good idea in
general -- but not one that works through anything but a
manually opened NAT.


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Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

[Gordon Henderson]

On Tue, 19 Dec 2006, Anthony Kepler wrote:

Do you, Gordon or Doug, happen to place international calls with early-dial 
enabled?  What kind of extensions.conf magic do you work to allow this?
I have been trying for some time to get this to work.  (My message from 
2006.11.03 regarding this is quoted just below)


Not me ( I'm in the UK FWIW).

I'm trying to get my users into thinking of the phones in the same terms 
as they'd treat their mobiles - so get them to dial the full area code 
starting with a zero (no 9 for outside line here, although I do support it 
in addition to zero), and then pushing the send key after they have 
entered the number... My reasoning for this is that it then mimics the way 
they use their mobiles, (and who doesn't have a mobile these days?) and 
you can dial the full number in the UK anyway without incuring any cost or 
call routing issues (just time to dial the 4 or 5 digit prefix)


Gordon


 
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:I am trying to allow users to 
place outgoing international calls from a
GXP-2000 with early dial enabled, connected to Asterisk 1.2.12.1 
http://1.2.12.1

I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

When I attempt to place a call to a number in, for instance, Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new stack
   -- Called g1/0112

It is attempting to dial out as soon as it receives a single digit to
represent the .
What I need is for it to wait a reasonable amount of time for additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the following in the
asterisk console:
   -- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5) in new stack
   -- Digit timeout set to 5
However, this is printed far less than 5 seconds before the dial out
attempt.

I assume there must be something relatively obvious I'm missing here...
if anyone can shed some light on this, it would be greatly appreciated.


Thank you,
   - Anthony Kepler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] | SIP/Email



Gordon Henderson wrote:

On Sun, 5 Nov 2006, Doug Crompton wrote:



On the Budgetone 200 it is in the account tab settings of the web setup
and it does work here with asterisk and my dialplans..



On the GPX2000's it's via the web interface under each of the 4 Line
configuration tabs. (so you'd have to set it on each account you were
using on the phone)

Gordon



Doug

On Sun, 5 Nov 2006, [iso-8859-1] Jes?s M?ndez Rom?n wrote:



Hi,

Where can I find that option?

Thanks
Jesus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Gordon
Henderson
Enviado el: Jueves, 02 de Noviembre de 2006 11:44 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

On Wed, 1 Nov 2006, Henry.L.Coleman wrote:



I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that 
when

dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the pattern is
match ed so it saves a second or two. Maybe they will fix this?


Set the Early Dial option - it's on a per-line basis, then as soon
as Asterisk gets a number it can dial, it will. No need to wait the 4
seconds or press the send button...

Gordon


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