[asterisk-users] Generate cdr on Hangup

[Deepesh D]

Hello,

I have the following dialplan

exten = _X.,1,Set(CDR(userfield)=test)
exten = _X.,n,Do some checks and hangup if checks fail
exten = _X.,n,Dial(SIP/${EXTEN})
exten = _X.,n,Hangup

1. If the Dial fails with a busy, noanswer or congestion then a cdr is
generated.
2. If the call fails before Dial (if the checks fail) then no cdr is
generated.

I would like to generate a cdr in the second case also. Is there a way to do
this?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Create Conference and exit myself

[Olivier]

2010/6/21 RSCL Mumbai rscl.mum...@gmail.com

 Hi,

 I am using Trixbox trixbox CE 2.6.2.3 (Stable) using Asterisk 1.4.22-4

 I am looking for the following functionality:

 ``
 I receive a call from Mr. A.
 I put Mr. A on hold.
 I dial Mr. B
 I connect Mr. A's call (which was on hold) to Mr. B and I get out of the
 call.
 Mr. A  Mr. B are in conversation, while my line is free to accept a new
 calls.

 

 What is the simplest way to achieve this ??

an attended transfer


 Thx
 Sans

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] features.conf - parkedcalls - transfer

[Olivier]

2010/6/21 Aksel Celasun ak...@abacus-it.no

  Hello dear list.





 I am having issues on parkedcalls.



 I am using a Cisco SPA525G as a test phone, and I have the transfer button
 there when I am in a call,

 But when I want to transfer the current call I am in, I push the transfer
 button, and onscreen I se “Enter Number”, and if I enter ex sip 200, I have
 to wait

Don't you have an OK button somewhere available when dialing ?
To my knowledge, most SIP phone can both use a timer and a dedicated key to
send a dialed number.

Cheers

 Almost 10 seconds, before the transfer to sip 200 is made, can I reduce
 that timer?

 And I can’t see any button on the Cisco phone which will function like a
 “direct transfer now”, do I have to wait…?



 And, secondly, is there a another way to do transfer/send to another sip
 phone?

 Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401
 and the Sip phone will directly call SIP401?





 Default features.conf context.





 Thank you.





 Med vennlig hilsen / Best regards

 Abacus IT AS

 - din Visma Software Partner

 - your Visma Software Partner



 *Tor Aksel **Celasun*

 Mobilnummer/cell phone: (+47) 900 15 103

 Sentralbord/Support 4000 1850

 ak...@abacus-it.no



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How do I access the Dialstatus numeric code received?

[Olivier]

2010/6/21 CDR vene...@gmail.com

 I need to access number received after a I dial a SIP or H323 call?
 suppose I get one of these:

 *404 Not found
 **486 Busy here
 **408 Request Timeout
 **480 Temporarily unavailable
 **480 Temporarily unavailable
 **403 Forbidden (+) **
 410 Gone
 **301 Moved Permanently
 **410 Gone **
 404 Not Found (=)
 **502 Bad Gateway
 **484 Address incomplete*

 How do I get the 404, 486, etc.
 F.A.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


For curiosity sake, why would need such data in a dialplan ?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Generate cdr on Hangup

[Olivier]

2010/6/22 Deepesh D deep.d2...@gmail.com

 Hello,

 I have the following dialplan

 exten = _X.,1,Set(CDR(userfield)=test)
 exten = _X.,n,Do some checks and hangup if checks fail
 exten = _X.,n,Dial(SIP/${EXTEN})
 exten = _X.,n,Hangup

 1. If the Dial fails with a busy, noanswer or congestion then a cdr is
 generated.
 2. If the call fails before Dial (if the checks fail) then no cdr is
 generated.

 I would like to generate a cdr in the second case also. Is there a way to
 do this?

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Have you tried with an h priority ?
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Unable to set callerid for incoming skype calls

[Matteo Piazza]

HI,

I'm using the usual Set(Callerid(num) function to change the incoming 
from skype callerid but it's not working.

Asterisk 1.4.31 and last release of skype channels


This is the dialplan

exten = _0X.,1,NoOP(${CALLERID(num)} - ${CALLERID(name)})
exten = _0X.,n,Set(STRINGA=Skype)
exten = _0X.,n,NoOP(${STRINGA})
exten = _0X.,n,Set(CALLERID(num) = ${STRINGA})
exten = _0X.,n,NoOP(${CALLERID(num)} - ${CALLERID(name)})

and is the output
NoOp(Skype/lab.trentinonetwork.it-08a32278, mapiazza - Matteo 
Piazza) in new stack
 -- Executing [0461020...@dial-to-openser:2] 
Set(Skype/lab.trentinonetwork.it-08a32278, STRINGA=Skype) in new stack
 -- Executing [0461020...@dial-to-openser:3] 
NoOp(Skype/lab.trentinonetwork.it-08a32278, Skype) in new stack
 -- Executing [0461020...@dial-to-openser:4] 
Set(Skype/lab.trentinonetwork.it-08a32278, CALLERID(num) = Skype) in 
new stack
 -- Executing [0461020...@dial-to-openser:5] 
NoOp(Skype/lab.trentinonetwork.it-08a32278, mapiazza - Matteo 
Piazza) in new stack

As you can see the incoming callerid(num) mapiazza doesn't change.

Is there any limitation on skype channel ?

Matteo


-- 
==
Ing. Matteo Piazza
Trentino Network s.r.l.
Area Ricerca  Sviluppo
Via Gilli, 2 - 38121 TRENTO
Tel (+39) 0461.020224
Mob (+39) 335.5378482
Fax (+39) 0461.020201
Cap. Soc. sottoscritto  7.573.248,00 - i. v.
REG. IMP. C.F. e P. IVA 01904880224
Società soggetta a direzione e controllo da parte della Provincia
Autonoma di Trento. C.F. e P. IVA 00337460224
==


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] NO ANSWER before playback or background function?

[Zhang Shukun]

hi,all

i find in asterisk 1.6.2.1, before play a sound file use playback or
background, it will answer the channel first.

but i want to answer the channel when dial someone and pick up the
phone.not play a file.

i know there are some params such as 'noanswer' for playback or 'n'
for background can not answer before play a file.

but it is not always take effect on my tests.as it said:

noanswer: Play the sound file, but don't answer the channel first (if
hasn't been answered already). Not all channels support playing
messages while still on hook.

Could you help me ?

Thanks!

-- 
Thanks for your supporting,
have a nice day.
Sucan

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Local channel usage

[Harel Cohen]

Hi All,
I’m trying to do “things” after my Dial application terminates (e.g. play IVR 
to called party, calling party, etc.). I’m trying to use the local channel for 
this purpose but so far with no success. I’m using 1.6.1.18 and this is my 
extensions.conf:

[Internal]
exten = _22,1,Dial(Local/${ext...@cw/n) ; 22 is test number
exten = _22,2,Noop(After Hangup)

[CW]
exten = _x.,1,Dial(SIP/307)
exten = _x.,2,Noop(After Hangup)

The call never reaches neither of the Noop applications. Consol:
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
-- Executing [...@internal:1] Dial(SIP/309-00a5, Local/2...@cw/n) 
in new stack
-- Called 2...@cw/n
-- Executing [...@cw:1] Dial(Local/2...@cw-af6f;2, SIP/307) in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
-- Called 307
-- SIP/307-00a6 is ringing
-- Local/2...@cw-af6f;1 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 is ringing
-- SIP/307-00a6 answered Local/2...@cw-af6f;2
-- Local/2...@cw-af6f;1 answered SIP/309-00a5
  == Spawn extension (CW, 22, 1) exited non-zero on 'Local/2...@cw-af6f;2'
  == Spawn extension (Internal, 22, 1) exited non-zero on 'SIP/309-00a5'
If I use the ‘g’ option in my Dial() both Noop will be run only if the called 
party hang-up first. I need a simple continuation in the dial plan regardless 
of who disconnected the call.
Thanks in advance
Harel

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] storing DTMF inputs

[nikhil singhania]

Thanks a lot Danny.
   I have done the part of playing a file by creating a context in my
dialplan. Now I   am puzzled as i wish to store the DTMF inputs done by the
users who is listening to the playback. I found there are ways, but some
specific way by which it is not stored in file but conveyed directly to the
asterisk server.
  When the call landed up on the softphone, i pressed keys the softphone
detects pressing of the keys but how the server will know which key is
pressed and CLI shows no such message of key pressing. Is it supposed to
show the message??
  There may be other ways too, what ever would be implemented easily.

-- 
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DAHDI: Inbound BRI call, DDI not presented

[Tzafrir Cohen]

On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote:
 Hello-
 
 I have a system with one D410P and one B200P (both OpenVox).  All is 
 well with the D410P, inbound and outbound, and I can initiate calls on 
 the B200P  BRI span, but there may be something wrong with my inbound 
 BRI setup:  there is no indication of an inbound call when I dial in to 
 it from the PSTN.
 
 When I run pri intense debug and make a call to the BRI span, I can 
 see a message containing the DDI that I'm dialing, in this case 336027 
 (BT supplies only the last 6 digits of a delivered number).  See debug 
 output below...

Is there anything you see in the dialplan trace itself?

Also, 'intense debug' shows a lot of noise of ISDN layer 2 (Q.921). But
that is normally not interesting. Do you see anything on a simple 'pri
debug span 1' (only layer 3 debug)?

 
 Have I neglected to set up some needed parameter?  This all worked on 
 older boards when using bristuff, but now I want to use dahdi.   My 
 client is in the UK, connected to BT, and I have specified euroisdn as 
 the switch type.
 
 many thanks
 
 -
 (snippet during inbound call to 336027)
 
  Supervisory frame:
  SAPI: 00  C/R: 0 EA: 0
   TEI: 000EA: 1
  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
  N(R): 008 P/F: 1
  0 bytes of data
 -- ACKing all packets from 8 to (but not including) 8
 -- Stopping T200 timer
 -- Starting T203 timer

Shouldn't an RR be sent back?

 Handling message for SAPI/TEI=0/0
 TEI: 0 State 7
 V(S) 8 V(A) 8 V(R) 8
 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0
 T200 0, N200 3, T203 1
 
  [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 
 a3 70 07 81 33 33 36 30 32 37 ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 127EA: 1
M3: 0   P/F: 0 M2: 0 11: 3  [ UI (unnumbered information) ]
  30 bytes of data
 Handling message for SAPI/TEI=0/127
 TEI: 0 State 7
 V(S) 8 V(A) 8 V(R) 8
 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0
 T200 0, N200 3, T203 1
 
  [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21 
 a3 70 07 81 33 33 36 30 32 37 ]
 
  Unnumbered frame:
  SAPI: 00  C/R: 1 EA: 0
   TEI: 127EA: 1
M3: 0   P/F: 0 M2: 0 11: 3  [ UI (unnumbered information) ]
  30 bytes of data
 Handling message for SAPI/TEI=0/127
 -

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 71, Issue 36

[Dharmesh Garg]

Hi,
I tried to use PLAYTONES with tonename and tonlist both but none of them 
worked.



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

[Tzafrir Cohen]

On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote:
 Hi Guys,
 
 An 8 channel 

FXO?

 Astribank is connected to Trixbox 2.8 and I ran
 freepbx-module-zapauto but I get the following when running these
 commands and can't make calls out:
 
 [Trixbox]# dahdi_genconf xpporder
 /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
 '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc

Which one, exactly? Trixbox originally had rather dates DAHDI drivers. I
believe you should now be able to find much newer ones. At least in
their repos.

 
 pbx*CLI dahdi show channels
 Chan Extension Context Language MOH Interpret Blocked State
 
 pseudo default default In Service
 1 from-pstn default In Service
 2 from-pstn default In Service
 3 from-pstn default In Service
 4 from-pstn default In Service
 5 from-pstn default In Service
 6 from-pstn default In Service
 
 7 from-pstn default In Service
 8 from-pstn default In Service
 
 pbx*CLI dahdi show status
 Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
 Xorcom XPD #00/00: FXO OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
 
 
 pbx*CLI dahdi show regdump 1
 Unable to get registers on channel 1
 Unable to get stats on channel 1

I believe that regdump uses some specific interface to the Digium card.
If you want a bunch of technical information you don't really
understand, look under /proc/xpp/ . I'm not going to start explaining it
beyond
http://docs.tzafrir.org.il/dahdi-linux/README.Astribank.html#_proc_interface

 
 [Trixbox]# dahdi_hardware -v
 /usr/sbin/dahdi_hardware: warning - OLD DRIVER: missing
 '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc
 
 usb:002/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
 LABEL=[usb:X1044207] connect...@usb-:00:1d.7-4
 XBUS-00/XPD-00: FXO (8) Span 1 DAHDI-SYNC
 
 
 Note: This Astribank is deployed in United Arab Emirates and I am not
 sure what the line type is in terms of Ground or Loop start and
 wondering if that makes a difference with the Astribank and the fact
 that it can't how the voltage using show regdump

IIRC they use LS (That is: no power denial is used at the end of a
call).

 
 And I am definitly not sure what that warning of OLD DRIVER is about.
 Any help is appreciated.

At the time we wrote it, we relied heavily on procfs. However procfs is
not something that can be safely used when the module is loaded or
removed. This is a fine way to get panics. Thus we gradually moved many
things from /proc to /sys .

Typically such a message would mean a combination of older dahdi-linux
(loaded) and newer dahdi-tools. Though it's a warning, and not an error.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Local channel usage

[Tiago Geada]

Hi,

After a Dial, the call is hung up. It doesn't carry on with dialplan unless
you specify the appropriate dial option.

Check wiki voip-info for cmd Dial, I think the option is g

2010/6/22 Harel Cohen ha...@easycall.gi

 Hi All,

 I’m trying to do “things” after my Dial application terminates (e.g. play
 IVR to called party, calling party, etc.). I’m trying to use the local
 channel for this purpose but so far with no success. I’m using 1.6.1.18 and
 this is my extensions.conf:



 [Internal]

 exten = _22,1,Dial(Local/${ext...@cw/n) ; 22 is test number

 exten = _22,2,Noop(After Hangup)



 [CW]

 exten = _x.,1,Dial(SIP/307)

 exten = _x.,2,Noop(After Hangup)



 The call never reaches neither of the Noop applications. Consol:

   == Using SIP RTP CoS mark 5

   == Using UDPTL CoS mark 5

 -- Executing [...@internal:1] Dial(SIP/309-00a5, Local/2...@cw/n)
 in new stack

 -- Called 2...@cw/n

 -- Executing [...@cw:1] Dial(Local/2...@cw-af6f;2, SIP/307) in new
 stack

   == Using SIP RTP CoS mark 5

   == Using UDPTL CoS mark 5

 -- Called 307

 -- SIP/307-00a6 is ringing

 -- Local/2...@cw-af6f;1 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 answered Local/2...@cw-af6f;2

 -- Local/2...@cw-af6f;1 answered SIP/309-00a5

   == Spawn extension (CW, 22, 1) exited non-zero on 'Local/2...@cw-af6f;2'

   == Spawn extension (Internal, 22, 1) exited non-zero on
 'SIP/309-00a5'

 If I use the ‘g’ option in my Dial() both Noop will be run only if the
 called party hang-up first. I need a simple continuation in the dial plan
 regardless of who disconnected the call.

 Thanks in advance

 Harel



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[Tiago Geada]

Hi!

If it was me, I would create a bash script calling asterisk -vrx core show
commands

something like:

for chan in $(asterisk -vrx core show channels concise);
do
asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
native;
done

On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:

 Hi Everyone,

 I want to know if a specific codec type is used at least one. For example,
 I want to know if out of the 100 calls on the system if there is a 1 channel
 that is running G.729 codec right now. If using dial-plan and I dial in, I
 can use this to obtain information about CURRENT channel. But it won't allow
 me to obtain information about OTHER channels and that is what I want to do.
 I want a search for all channels and an output spit out as g729 or TRUE or
 FALSE if there is a g729 channel.

 exten = s,1,Answer()
 exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
 exten = s,n,NoOp(${foo})

 Above  NoOp spits out g729 if I call in with a g729 codec. But I want 
 that to be about other channels and not the one I am calling into.

 Thanks,

 Bruce


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] UDPTL T38 via NAT

[Remco Bressers]

Dear list,

I've got the following setup :

[FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
PBX WAN, i see the following in udptl debug :

Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

This means my outgoing udptl traffic is correctly translated, but
somehow i'm sending 172.16.0.156 instead of my public IP address on the
firewall.

On the LAN PBX, i've got the following config :

[general]
t38pt_udptl=yes

[202]
type=friend
secret=***
username=202
regexten=202
host=dynamic
canreinvite=yes
allow=alaw
context=local
qualify=yes

On the WAN PBX, the config for the trunk is the following :

[general]
t38pt_udptl=yes

[trunk]
type=peer
context=trunk-in
host=62.180.xxx.xxx
port=5070
disallow=all
allow=alaw
allow=ulaw
qualify=yes
nat=no


Can anybody tell me how to change this behaviour? Fax isn't working
ofcourse.

-- 
Kind regards,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[Tzafrir Cohen]

On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
 Hi!
 
 If it was me, I would create a bash script calling asterisk -vrx core show
 commands
 
 something like:
 
 for chan in $(asterisk -vrx core show channels concise);
 do
 asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
 native;
 done

The overhead of each 'asterisk -rx' command is noticable. If you have 10
calls or more, this can have an odd effect.

Not to mention that the fact that it is so slow exposes its raciness[1].

 
 On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:
 
  Hi Everyone,
 
  I want to know if a specific codec type is used at least one. For example,
  I want to know if out of the 100 calls on the system if there is a 1 channel
  that is running G.729 codec right now. If using dial-plan and I dial in, I
  can use this to obtain information about CURRENT channel. But it won't allow
  me to obtain information about OTHER channels and that is what I want to do.
  I want a search for all channels and an output spit out as g729 or TRUE or
  FALSE if there is a g729 channel.
 
  exten = s,1,Answer()
  exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
  exten = s,n,NoOp(${foo})
 
  Above  NoOp spits out g729 if I call in with a g729 codec. But I want 
  that to be about other channels and not the one I am calling into.
 
  Thanks,
 
  Bruce

[1] Which should naturally be fixed using locks :-)

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] [AGI] What scripting language for embedded hardware?

[Gilles]

On Mon, 21 Jun 2010 16:10:12 +, Edwin Quijada
listas_quij...@hotmail.com wrote:
Uhmmm.. remember for each channel you run perl or php interpreter so with that 
amount of memory maybe this can be a problem.
 For that kind of project I'd use C or java as fastagi protocol

Thanks Edwin. In my case, the hardware will only handle one or two
channels at a time (SOHO user), so it's OK if the interpreter takes
about 2-3MB, especially if it can be launched once to handle AGI
scripts for each channel.

As a middle-of-the-road solution, I'm thinking Lua, as an easier to
use solution than C while keeping things tidy.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Generate cdr on Hangup

[Zeeshan Zakaria]

Yes, generate CDR in h extension, i.e. add the following to your context:

exten = h,1,ResetCDR(vw)

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-22 2:25 AM, Olivier oza_4...@yahoo.fr wrote:



2010/6/22 Deepesh D deep.d2...@gmail.com

 
  Hello,
 
  I have the following dialplan
 
  exten = _X.,1,Set(CDR(userfield)=test)
  exten = ...
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Have you tried with an h priority ?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] NO ANSWER before playback or background function?

[Philipp von Klitzing]

Hi!

 but i want to answer the channel when dial someone and pick up the
 phone.not play a file.

Search this list for early media and maybe also for progress.

Philipp


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[A J Stiles]

Is anybody else using the following combination:

* a TE410P card  (wct4xxp driver)
* a BT ISDN connection
* DAHDI 2.3.0.1
* Asterisk 1.6.2.9

I'm trying to configure a new box to replace a legacy system  (same hardware; 
some old version of Asterisk with Zaptel; works lovely but hopelessly 
out-of-date)  and not having much joy.  Specifically, I couldn't get it to 
see a D-channel on channel 16 of span 1.  And without a D-channel, there is 
no way I'm going to be able to get a call in or out.

This could well be because the syntax of modern /etc/dahdi/system.conf 
and /etc/asterisk/chan_dahdi.conf is slightly incompatible with the old 
zapata.conf and zaptel.conf files.

So I guess the first question should be, has anybody else managed to make this 
combination work?

(I'm new here and I may have missed some important information, so please 
ask.)

-- 
AJS

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Local channel usage

[Zeeshan Zakaria]

'g' option continues the dial plan after the call has been answered, not
after it is hung up. Depending upon what you are trying to do, first try to
use h extension, i.e. in the example you gave, you should replace '_22,2'
with 'h,1'.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-22 6:23 AM, Tiago Geada tiago.ge...@gmail.com wrote:

Hi,

After a Dial, the call is hung up. It doesn't carry on with dialplan unless
you specify the appropriate dial option.

Check wiki voip-info for cmd Dial, I think the option is g

2010/6/22 Harel Cohen ha...@easycall.gi

 
  Hi All,
 
  I’m trying to do “things” after my Dial application terminates (e.g. play
 IVR to cal...
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[Tzafrir Cohen]

On Tue, Jun 22, 2010 at 12:26:28PM +0100, A J Stiles wrote:
 Is anybody else using the following combination:
 
 * a TE410P card  (wct4xxp driver)
 * a BT ISDN connection
 * DAHDI 2.3.0.1
 * Asterisk 1.6.2.9
 
 I'm trying to configure a new box to replace a legacy system  (same hardware; 
 some old version of Asterisk with Zaptel; works lovely but hopelessly 
 out-of-date)  and not having much joy.  Specifically, I couldn't get it to 
 see a D-channel on channel 16 of span 1.  And without a D-channel, there is 
 no way I'm going to be able to get a call in or out.
 
 This could well be because the syntax of modern /etc/dahdi/system.conf 
 and /etc/asterisk/chan_dahdi.conf is slightly incompatible with the old 
 zapata.conf and zaptel.conf files.

The old ones should work just as well. Apart from 'echocanceller' lines
in system.conf. Those may prevent you from having a working echo
canceller, but nothing worse.

What do you have in those files?

What's the output of lsdahdi ?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Necati Demir]

Thanks for your answers.
I think i still have questions.

Now without a ISDN PRI card, i can connect to SIP server and do what i want.
The card that i mentioned has a RJ45 port, so i think i still did not
understand the advantage of it.

On 21 June 2010 22:04, Necati Demir nde...@demir.web.tr wrote:

 This is a really rookie question: when should i use TE110P ISDN PRI Card?

 --
 Necati DEMİR
 ---




-- 
Necati DEMİR
http://demir.web.tr

Pi Bilişim Teknolojileri
http://www.pibilisim.com.tr
--
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[A J Stiles]

On Tuesday 22 Jun 2010, Tzafrir Cohen wrote:
 The old ones should work just as well. Apart from 'echocanceller' lines
 in system.conf. Those may prevent you from having a working echo
 canceller, but nothing worse.

 What do you have in those files?

 What's the output of lsdahdi ?

Files attached.

Note I commented out all but the first span, to try to make things easier to 
work with.  Unfortunately, it's a live system; which means I can't test it 
for real by plugging into the ISDN until everyone else in the office has gone 
home .

-- 
AJS
[channels]

context=bt-isdn
signalling=pri_cpe
switchtype=euroisdn
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
echocancel=no
echotraining=no
rxgain=0.0
txgain=0.0
immediate=no
musiconhold=default
busydetect=no
busycount=8
usecallingpres=yes
pridialplan=unknown

; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 
group=1
context=bt-isdn
switchtype = euroisdn
signalling = pri_cpe
channel = 1-15,17-31
;;context = default
;;group = 63

;; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 
;group=2
;context=bt-isdn
;switchtype = euroisdn
;signalling = pri_cpe
;channel = 32-46,48-62
;;;context = default
;;;group = 63
;
;; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 
;group=3
;context=milgram
;switchtype = euroisdn
;signalling = pri_cpe
;channel = 63-77,79-93
;;;context = default
;;;group = 63
;
;; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 
;group=4
;context=bluecheese
;switchtype = euroisdn
;signalling = pri_cpe
;channel = 94-108,110-124
;;;context = default
;;;group = 63

### Span  1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4
  1 PRIClear   (In use) (SWEC: MG2)
  2 PRIClear   (In use) (SWEC: MG2)
  3 PRIClear   (In use) (SWEC: MG2)
  4 PRIClear   (In use) (SWEC: MG2)
  5 PRIClear   (In use) (SWEC: MG2)
  6 PRIClear   (In use) (SWEC: MG2)
  7 PRIClear   (In use) (SWEC: MG2)
  8 PRIClear   (In use) (SWEC: MG2)
  9 PRIClear   (In use) (SWEC: MG2)
 10 PRIClear   (In use) (SWEC: MG2)
 11 PRIClear   (In use) (SWEC: MG2)
 12 PRIClear   (In use) (SWEC: MG2)
 13 PRIClear   (In use) (SWEC: MG2)
 14 PRIClear   (In use) (SWEC: MG2)
 15 PRIClear   (In use) (SWEC: MG2)
 16 PRIHDLCFCS (In use)
 17 PRIClear   (In use) (SWEC: MG2)
 18 PRIClear   (In use) (SWEC: MG2)
 19 PRIClear   (In use) (SWEC: MG2)
 20 PRIClear   (In use) (SWEC: MG2)
 21 PRIClear   (In use) (SWEC: MG2)
 22 PRIClear   (In use) (SWEC: MG2)
 23 PRIClear   (In use) (SWEC: MG2)
 24 PRIClear   (In use) (SWEC: MG2)
 25 PRIClear   (In use) (SWEC: MG2)
 26 PRIClear   (In use) (SWEC: MG2)
 27 PRIClear   (In use) (SWEC: MG2)
 28 PRIClear   (In use) (SWEC: MG2)
 29 PRIClear   (In use) (SWEC: MG2)
 30 PRIClear   (In use) (SWEC: MG2)
 31 PRIClear   (In use) (SWEC: MG2)
### Span  2: TE4/0/2 T4XXP (PCI) Card 0 Span 2
 32 PRI
 33 PRI
 34 PRI
 35 PRI
 36 PRI
 37 PRI
 38 PRI
 39 PRI
 40 PRI
 41 PRI
 42 PRI
 43 PRI
 44 PRI
 45 PRI
 46 PRI
 47 PRI
 48 PRI
 49 PRI
 50 PRI
 51 PRI
 52 PRI
 53 PRI
 54 PRI
 55 PRI
 56 PRI
 57 PRI
 58 PRI
 59 PRI
 60 PRI
 61 PRI
 62 PRI
### Span  3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
 63 PRI
 64 PRI
 65 PRI
 66 PRI
 67 PRI
 68 PRI
 69 PRI
 70 PRI
 71 PRI
 72 PRI
 73 PRI
 74 PRI
 75 PRI
 76 PRI
 77 PRI
 78 PRI
 79 PRI
 80 PRI
 81 PRI
 82 PRI
 83 PRI
 84 PRI
 85 PRI
 86 PRI
 87 PRI
 88 PRI
 89 PRI
 90 PRI
 91 PRI
 92 PRI
 93 PRI
### Span  4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
 94 PRI
 95 PRI
 96 PRI
 97 PRI
 98 PRI
 99 PRI
100 PRI
101 PRI
102 PRI
103 PRI
104 PRI
105 PRI
106 PRI
107 PRI
108 PRI
109 PRI
110 PRI
111 PRI
112 PRI
113 PRI
114 PRI
115 PRI
116 PRI
117 PRI
118 PRI
119 PRI
120 PRI
121 PRI
122 PRI
123 PRI
124 PRI

# Autogenerated by /usr/sbin/dahdi_genconf on Thu Jun 17 21:56:01 2010
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#

# Global data

loadzone= uk
defaultzone = uk

# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31

## Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 
#span=2,0,0,ccs,hdb3,crc4
## termtype: te
#bchan=32-46,48-62
#dchan=47
#echocanceller=mg2,32-46,48-62
#
## Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 
#span=3,0,0,ccs,hdb3,crc4
## termtype: te
#bchan=63-77,79-93
#dchan=78
#echocanceller=mg2,63-77,79-93
#
## Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 
#span=4,0,0,ccs,hdb3,crc4
## 

[asterisk-users] Unregister and register SIP phones by using num pad on phones?

[Aksel Celasun]

Hello dear list.


A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, 
and we had the opportunity
to unregister user  by typing *-a number and -* again, ex * 99 *, and then the 
phone number/sip extension was unavailable, and
all of the calls to that extension was redirected to the receptionist.

When the user came back and wanted to register her sip account/extension, the 
user typed in a similar code ex * 99 * and internal sip, and voila,
Extension is online again. This was very useful regarding when users changed 
offices and so on, they didn't have to carry their phones, they just 
unregistered and
Later on registered themselves on the other office.

Are there any similar options on Asterisk, or is this more or less HW related?
Currently testing SNOM m300,Cisco spa525, Cisco spa520, and grandstrem gxp 3000.

Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.nomailto:ak...@abacus-it.no

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Zeeshan Zakaria]

If you are doing pure VoIP and no PRIs or PSTN, you don't need these cards.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-22 7:57 AM, Necati Demir nde...@demir.web.tr wrote:

Thanks for your answers.
I think i still have questions.

Now without a ISDN PRI card, i can connect to SIP server and do what i want.
The card that i mentioned has a RJ45 port, so i think i still did not
understand the advantage of it.



On 21 June 2010 22:04, Necati Demir nde...@demir.web.tr wrote:

 This is a really rookie quest...
-- 
Necati DEMİR
http://demir.web.tr

Pi Bilişim Teknolojileri
http://www.pibilisim.com.tr
--

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[Tzafrir Cohen]

On Tue, Jun 22, 2010 at 12:26:28PM +0100, A J Stiles wrote:
 Is anybody else using the following combination:
 
 * a TE410P card  (wct4xxp driver)
 * a BT ISDN connection
 * DAHDI 2.3.0.1
 * Asterisk 1.6.2.9
 
 I'm trying to configure a new box to replace a legacy system  (same hardware; 
 some old version of Asterisk with Zaptel; works lovely but hopelessly 
 out-of-date)  and not having much joy.  Specifically, I couldn't get it to 
 see a D-channel on channel 16 of span 1.  And without a D-channel, there is 
 no way I'm going to be able to get a call in or out.

What version of libpri? What does it mean 'no D-Channel'? What is the
output of pri show span 1'?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How do I access the Dialstatus numeric code received?

[CDR]

Tilghman Lesher wrote
Not available in anything other than trunk (to be 1.8).  It depends upon a
new
feature, so it's not something you can easily backport.  After dialling, the
SIP code is available in ${HASH(SIP_CAUSE,channel-name)}
In a real dialplan, how do I get a variable with channel-name? I mean: My
app is Sip2Sip. I get one call inbound, get the number dialed ${EXTEN} and
proceed to try many carriers. If the carriers send me something different
than 503 Service Unavailable or 404 Not Found, I need to close the call and
send back whatever SIP code I got, exactly. There is no way for me to do
that now. Unless I am missing something, I can only play with ${DIALSTATUS}
and do Hangup(Code), but my Code variable is never the same that I got
from the second leg. I would like to be able to do
Hangup( ${HASH(SIP_CAUSE,channel-name)}, where channel-name is the last
channel used to dial-out. How do I do this in trunk? I will have to start
using trunk in production. Another issues is the the function Hangup(Code)
takes a decimal, not related to the SIP code I just got. How would you
design your 1.8 or 1.62 dialplan around this issue?
Thanks in advance.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk T.38 Gateway code testing

[marek cervenka]

asterisk t38 gw patch updated to 1.6.2.9
https://issues.asterisk.org/view.php?id=13405

 i made page for Asterisk T.38 Gateway code testing
 http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway

 Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later 
 BUT Asterisk 1.8 is too far and we need t.38 gw now (for testing etc)

 if you would like help/test current code(last patch from 
 https://issues.asterisk.org/view.php?id=13405), please contact me
 i have 2 public testing machines connected over E1

 my jabber is cerv...@njs.netlab.cz


---
Marek Cervenka
===


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Johann Steinwendtner]

On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.



Did you try t38pt_usertpsource=yes ?

Best regards

Hans

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Remco Bressers]

On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.

 
 
 Did you try t38pt_usertpsource=yes ?
 

Hi,

Yes, i tried adding that to the SIP peer configuration for the FAX ATA.
Should i put it on the PBX trunk configuration also??

Remco


-- 
Met vriendelijke groet,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl
altijd online? www.signet.nl

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] PRI span problem - no D channel

[Mike]

Hi,

 

I have the following happen to me after the restart of one of my servers:
out of my 3 PRIs (all configured with the same technical settings), the last
one isn't coming back.  It's underutilized (chances it didn't get a call
since my reboot), if it makes a difference .

 

The PRI goes from provisioned to unprovisioned, and I get this regularly:

[Jun 22 09:03:48] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No
D-channels available!  Using Primary channel 72 as D-channel anyway!

 

Here is my PRI debug span:

 

-- Timeout occured, restarting PRI

q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED

Sending Set Asynchronous Balanced Mode Extended

q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH

[Jun 22 09:03:48] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No
D-channels available!  Using Primary channel 72 as D-channel anyway!

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Timeout occured, restarting PRI

q921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED

Sending Set Asynchronous Balanced Mode Extended

q921.c:211 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH

[Jun 22 09:04:19] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No
D-channels available!  Using Primary channel 72 as D-channel anyway!

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

Sending Set Asynchronous Balanced Mode Extended

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

-- Got SABME from network peer.

Sending Unnumbered Acknowledgement

q921.c:858 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED

q921.c:805 q921_dchannel_up: q921_state now is
Q921_LINK_CONNECTION_ESTABLISHED

edison*CLI pri no debug span 3

Disabled debugging on span 3

 [Jun 22 09:04:51] WARNING[30723]: chan_dahdi.c:2790 pri_find_dchan: No
D-channels available!  Using Primary channel 72 as D-channel anyway!

 

 

 

Any clue, anyone?

 

Mike

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com 

Re: [asterisk-users] Unregister and register SIP phones by using num pad on phones?

[Philipp von Klitzing]

Hi!

 A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip
 phones, and we had the opportunity to unregister user by typing *-a
 number and -* again, ex * 99 *, and then the phone number/sip extension
 was unavailable

It is entirely up to you to design the Asterisk dialplan this way; many 
implementations have created such a roving user one way or another. 
Some more complex solutions also re-provision the phones (e.g. 
Gemeinschaft) accordingly, while others simply make the extension 
available/unavailable.

I am sure that the Wiki at voip-info.org has some examples.

There are also phones that support hot-desking (the SNOMs do, for 
example), but usually I find that to be too complicated for the end user.

Philipp


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ISP down internal phones become unavailable

[Mr Shunz]

Hi all,

 I have a PRI, and when the Internet connection goes out so do my
 phones.  I suspect it is some type of DNS issue.  I do have a SIP
 trunk, and it appears that if I lose DNS to the SIP trunk, the entire
 PBX is offline.  I have no actual proof of any of this, and have not
 done any extensive testing to prove or disprove this.

well, we have various asterisk installations, ranging from 1.4.25
to (upgraded today) 1.4.33 (we don't use 1.6.X yet) and two
of them show this behaviour...
one is upgraded to 1.4.33, the other is 1.4.30, they have similar configuration
to all the other machines (which work flawlessy even when connection
is down), and the phones are the same brand/model we use everywhere,
with almost the same configuration.

I'm not sure about a DNS issue because all our customers have local
DNS/cache servers and we configure all the phones (and sip trunks
on asterisks) with ip addresses and not FQDNs just to be sure...

what we see is when the trunk goes down, i.e.
'Registration for ...@yy.yy.yy.yy timed out, trying again (Attempt #ZZ)'
we have also 'Peer XXX is now UNREACHABLE (internal phones), even if they
are pingable/accessibile on the LAN...

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 

Daniele Santi   .o.
dani...@santi.vr.it ..o () ascii ribbon campaign
Linux User #415108  ooo /\  www.asciiribbon.org


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] storing DTMF inputs

[Danny Nicholas]

DTMF inputs are detected from the dialplan (usually, there are some
exceptions) from a waitexten or read command - let's say your play back said
press 1 for apples, 2 for cherries or enter a count of bananas to ship.
Here's a dialplan to let you handle that situation

Exten = 100,1,playback(myprompt)

Exten = 100,2,Goto(get-fruit,s,1)

 

[get-fruit]

Exten = s,1,waitexten(5)

Exten = s,2,read(bananas,,100,5)

Exten = s,3,verbose(you want ${bananas} bananas)

Exten = s,4,hangup

Exten = 1,1,playback(apples)

Exten = 1,2,hangup

Exten = 2,1,playback(cherries)

Exten = 2,1,hangup

Exten = t,1,playback(whatdidyouwant)

Exten = t,n,hangup

Exten = *,1,playback(badinput)

Exten = *,n,hangup

 

If you press 1, you get the apples message, 2 the cherries message.  If you
wait 5 seconds, your dtmf input becomes a banana order.  If you didn't wait
5 seconds, bananas is invalid input.  If you enter nothing, you just
timeout.

 

  _  

From: nikhil singhania [mailto:niksingha...@gmail.com] 
Sent: Tuesday, June 22, 2010 3:52 AM
To: da...@debsinc.com
Cc: asterisk-users@lists.digium.com
Subject: storing DTMF inputs

 

Thanks a lot Danny.

   I have done the part of playing a file by creating a context in my
dialplan. Now I   am puzzled as i wish to store the DTMF inputs done by the
users who is listening to the playback. I found there are ways, but some
specific way by which it is not stored in file but conveyed directly to the
asterisk server. 

  When the call landed up on the softphone, i pressed keys the softphone
detects pressing of the keys but how the server will know which key is
pressed and CLI shows no such message of key pressing. Is it supposed to
show the message??

  There may be other ways too, what ever would be implemented easily.

-- 
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
 niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Necati Demir]

As you know, sending fax over ip is not very stable. So do these cards help
to make this situation stable?

On 22 June 2010 15:18, Zeeshan Zakaria zisha...@gmail.com wrote:

 If you are doing pure VoIP and no PRIs or PSTN, you don't need these cards.

 Zeeshan A Zakaria

 --
 www.ilovetovoip.com

 On 2010-06-22 7:57 AM, Necati Demir nde...@demir.web.tr wrote:

 Thanks for your answers.
 I think i still have questions.

 Now without a ISDN PRI card, i can connect to SIP server and do what i
 want. The card that i mentioned has a RJ45 port, so i think i still did not
 understand the advantage of it.



 On 21 June 2010 22:04, Necati Demir nde...@demir.web.tr wrote:
 
  This is a really rookie quest...

 --
 Necati DEMİR
 http://demir.web.tr

 Pi Bilişim Teknolojileri
 http://www.pibilisim.com.tr
 --

 --

 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Necati DEMİR
http://demir.web.tr

Pi Bilişim Teknolojileri
http://www.pibilisim.com.tr
--
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[A J Stiles]

On Tuesday 22 Jun 2010, Tzafrir Cohen wrote:
 On Tue, Jun 22, 2010 at 12:26:28PM +0100, A J Stiles wrote:
  Is anybody else using the following combination:
 
  * a TE410P card  (wct4xxp driver)
  * a BT ISDN connection
  * DAHDI 2.3.0.1
  * Asterisk 1.6.2.9
 
  I'm trying to configure a new box to replace a legacy system  (same
  hardware; some old version of Asterisk with Zaptel; works lovely but
  hopelessly out-of-date)  and not having much joy.  Specifically, I
  couldn't get it to see a D-channel on channel 16 of span 1.  And without
  a D-channel, there is no way I'm going to be able to get a call in or
  out.

 What version of libpri?

Libpri version is 1.4.11.1.  Sorry.

 What does it mean 'no D-Channel'?

The message I get is

WARNING[3502]: chan_dahdi.c:4160 pri_find_dchan: No D-channels available!  
Using Primary channel 16 as D-channel anyway!

which is the same message I get if I attempt to start up Asterisk with the 
ISDN disconnected.  (I transferred the cable from one machine to the other, 
so that eliminates that as a cause.  Could the card be the problem?)

 What is the  
 output of pri show span 1'?

With the ISDN disconnected, I get:

Primary D-channel: 16
Status: Provisioned, Down, Active
Switchtype: EuroISDN
Type: CPE
Overlap Dial: 0
Logical Channel Mapping: 0
Timer and counter settings:
  N200: 3
  N202: 3
  K: 7
  T200: 1000
  T202: 1
  T203: 1
  T303: 4000
  T305: 3
  T308: 4000
  T309: 6000
  T313: 4000
  T-HOLD: 4000
  T-RETRIEVE: 4000
  T-RESPONSE: 4000
Overlap Recv: No

On the working box  (connected to ISDN and calls going in and out; running 
older versions of all software),  the same command produces:

Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3

-- 
AJS

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Why asterisk down when inet server down?

[Theo Band]

Steve Edwards wrote:
 On Fri, 18 Jun 2010, sean darcy wrote:

 (Sean has a problem and several posters suspect it is DNS related.)

 On Fri, 18 Jun 2010, Zeeshan Zakaria wrote:
   
 Did you check /etc/resolv? Does it point to any DNS by domain name?
 

 If you mean /etc/resolv.conf and the nameserver option, an IP address 
 is required -- otherwise all attempts to use the resolver library fail.
   
And what if you have two servers in /etc/resolv.conf?
I have two but if the first fails Asterisk does not resolve on the
second one. Open calls continue but registrations start to fail.

I probably gonna try the caching name server, but it feels like a bug in
Asterisk.

Theo

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[Scott Stingel]

On 6/22/2010 4:26 AM, A J Stiles wrote:
 Is anybody else using the following combination:

 * a TE410P card  (wct4xxp driver)
 * a BT ISDN connection
 * DAHDI 2.3.0.1
 * Asterisk 1.6.2.9

 I'm trying to configure a new box to replace a legacy system  (same hardware;
 some old version of Asterisk with Zaptel; works lovely but hopelessly
 out-of-date)  and not having much joy.  Specifically, I couldn't get it to
 see a D-channel on channel 16 of span 1.  And without a D-channel, there is
 no way I'm going to be able to get a call in or out.

 This could well be because the syntax of modern /etc/dahdi/system.conf
 and /etc/asterisk/chan_dahdi.conf is slightly incompatible with the old
 zapata.conf and zaptel.conf files.

 So I guess the first question should be, has anybody else managed to make this
 combination work?

 (I'm new here and I may have missed some important information, so please
 ask.)


Hi-
I've been going through the same upgrade process recently, and had the 
same error (shown in your other message).  I had forgotten that the 
equipment I was plugged in to was CPE, so I had to change my new setting 
for that span to NET rather than CPE.  I notice in your old zapata 
files that you had CPE for two spans and NET for the other two, and your 
dahdi_chan setup is set up the same.  But I'm thinking perhaps during 
testing you plugged a CPE on your new setup to a CPE on the other, which 
would produce the symptoms you see.

-Scott


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Tim Nelson]

- Necati Demir nde...@demir.web.tr wrote: 
 As you know, sending fax over ip is not very stable. So do these cards help 
 to make this situation stable? 
 

Stopping the top posting parade... 

Using an analog digitial TDM connection will be far superior if you're using it 
for your connectivity to the PSTN for faxing. Faxing over VoIP is a headache 
and sometimes only slightly less of a headache when using T.38. 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Tim Nelson]

- Zeeshan Zakaria zisha...@gmail.com wrote: 
If you are doing pure VoIP and no PRIs or PSTN, you don't need these cards. 


Zeeshan A Zakaria 




Un-top-posting... 

The cards aren't needed unless of course you want a stable hardware timing 
source for Meetme() or IAX trunk mode as an alternative to dahdi_dummy. 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 -- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Johann Steinwendtner]

On 2010-06-22 15:16, Remco Bressers wrote:
 On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.



 Did you try t38pt_usertpsource=yes ?


 Hi,

 Yes, i tried adding that to the SIP peer configuration for the FAX ATA.
 Should i put it on the PBX trunk configuration also??

 Remco

Yes.



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[marek cervenka]

 On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.

try asterisk 1.6.2.9


---
Marek Cervenka
===


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Remco Bressers]

On 06/22/2010 04:35 PM, Johann Steinwendtner wrote:
 On 2010-06-22 15:16, Remco Bressers wrote:
 On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.



 Did you try t38pt_usertpsource=yes ?


 Hi,

 Yes, i tried adding that to the SIP peer configuration for the FAX ATA.
 Should i put it on the PBX trunk configuration also??

 Remco

 Yes.
 

This results in the very same problem :

Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 101, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 102, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 103, len 32)
Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)


-- 
Kind regards,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Remco Bressers]

On 06/22/2010 04:38 PM, marek cervenka wrote:
 On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.
 
 try asterisk 1.6.2.9

What would be the reason to do that? Is there any change on this in 1.6.2.9?

-- 
Regards,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] DAHDI: Inbound BRI call, DDI not presented

[Scott Stingel]

On 6/22/2010 2:03 AM, Tzafrir Cohen wrote:
 On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote:

 Hello-

 I have a system with one D410P and one B200P (both OpenVox).  All is
 well with the D410P, inbound and outbound, and I can initiate calls on
 the B200P  BRI span, but there may be something wrong with my inbound
 BRI setup:  there is no indication of an inbound call when I dial in to
 it from the PSTN.

 When I run pri intense debug and make a call to the BRI span, I can
 see a message containing the DDI that I'm dialing, in this case 336027
 (BT supplies only the last 6 digits of a delivered number).  See debug
 output below...
  
 Is there anything you see in the dialplan trace itself?

 Also, 'intense debug' shows a lot of noise of ISDN layer 2 (Q.921). But
 that is normally not interesting. Do you see anything on a simple 'pri
 debug span 1' (only layer 3 debug)?


 Have I neglected to set up some needed parameter?  This all worked on
 older boards when using bristuff, but now I want to use dahdi.   My
 client is in the UK, connected to BT, and I have specified euroisdn as
 the switch type.

 many thanks

 -
 (snippet during inbound call to 336027)

   Supervisory frame:
   SAPI: 00  C/R: 0 EA: 0
TEI: 000EA: 1
   Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
   N(R): 008 P/F: 1
   0 bytes of data
 -- ACKing all packets from 8 to (but not including) 8
 -- Stopping T200 timer
 -- Starting T203 timer
  
 Shouldn't an RR be sent back?


 Handling message for SAPI/TEI=0/0
 TEI: 0 State 7
 V(S) 8 V(A) 8 V(R) 8
 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0
 T200 0, N200 3, T203 1

   [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21
 a3 70 07 81 33 33 36 30 32 37 ]

   Unnumbered frame:
   SAPI: 00  C/R: 1 EA: 0
TEI: 127EA: 1
 M3: 0   P/F: 0 M2: 0 11: 3  [ UI (unnumbered information) ]
   30 bytes of data
 Handling message for SAPI/TEI=0/127
 TEI: 0 State 7
 V(S) 8 V(A) 8 V(R) 8
 K 1, RC 0, l3initiated 0, reject_except 0 ack_pend 0
 T200 0, N200 3, T203 1

   [ 02 ff 03 08 01 01 05 a1 04 03 80 90 a3 18 01 89 1e 02 84 83 6c 02 21
 a3 70 07 81 33 33 36 30 32 37 ]

   Unnumbered frame:
   SAPI: 00  C/R: 1 EA: 0
TEI: 127EA: 1
 M3: 0   P/F: 0 M2: 0 11: 3  [ UI (unnumbered information) ]
   30 bytes of data
 Handling message for SAPI/TEI=0/127
 -
  

Thanks, will try the less intense debug.  I thought it was interesting 
however that the incoming DDI was in the message, but not showing up in 
the dialplan trace..
-Scott


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RES: AMD

[Tetra Informatica]

Thanks a lot, John.

It’s all working well now.

 

  _  

De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de John Rose
Enviada em: segunda-feira, 21 de junho de 2010 21:15
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Assunto: Re: [asterisk-users] AMD

 

Sometimes you have to play some audio before calling AMD or other audio
functions for whatever reason... Like play 100ms of silence in a .wav file
immediately after answer. This causes RTP to be sent out to the carrier.

 

John

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tetra
Informatica
Sent: Monday, June 21, 2010 3:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMD

 

Hi

 

I am using the AMD application in a power dialing.

All works well when I use an internal extension but when I try to use an
external number, the AMD every times returns non human status. Also the
AMDCAUSE returns Total-Time-5500. I am using the default parameters in
AMD.CONF.

Anybody has some idea?

Thanks

 

Sergio

 

Nenhum vírus encontrado nessa mensagem recebida.
Verificado por AVG - www.avgbrasil.com.br
Versão: 9.0.829 / Banco de dados de vírus: 271.1.1/2953 - Data de
Lançamento: 06/21/10 03:36:00


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ISP down internal phones become unavailable

[Ryan Wagoner]

On Tue, Jun 22, 2010 at 9:33 AM, Mr Shunz mrsh...@gmail.com wrote:
 Hi all,

 I have a PRI, and when the Internet connection goes out so do my
 phones.  I suspect it is some type of DNS issue.  I do have a SIP
 trunk, and it appears that if I lose DNS to the SIP trunk, the entire
 PBX is offline.  I have no actual proof of any of this, and have not
 done any extensive testing to prove or disprove this.

 well, we have various asterisk installations, ranging from 1.4.25
 to (upgraded today) 1.4.33 (we don't use 1.6.X yet) and two
 of them show this behaviour...
 one is upgraded to 1.4.33, the other is 1.4.30, they have similar 
 configuration
 to all the other machines (which work flawlessy even when connection
 is down), and the phones are the same brand/model we use everywhere,
 with almost the same configuration.

 I'm not sure about a DNS issue because all our customers have local
 DNS/cache servers and we configure all the phones (and sip trunks
 on asterisks) with ip addresses and not FQDNs just to be sure...

 what we see is when the trunk goes down, i.e.
 'Registration for ...@yy.yy.yy.yy timed out, trying again (Attempt #ZZ)'
 we have also 'Peer XXX is now UNREACHABLE (internal phones), even if they
 are pingable/accessibile on the LAN...

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




 --
 
 Daniele Santi       .o.
 dani...@santi.vr.it ..o () ascii ribbon campaign
 Linux User #415108  ooo /\  www.asciiribbon.org
 


It is interesting that you are seeing this on different machines with
the same Asterisk version. There must be something different in the
configuration or DNS. However Asterisk should gracefully handle no DNS
or a SIP provider issue without affecting the phones. I haven't been
able to troubleshoot this much since I can't just take the Internet
connection down.

Ryan

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] FIXED: DAHDI: Inbound BRI call, DDI not presented

[Scott Stingel]

Thanks to the OpenVox engineer for picking this up:

I had bri_cpe for my signaling type, should be bri_cpe_ptmp.  The 
BRI circuit on the B200P works fine now in both directions.

-Scott


On 6/22/2010 7:58 AM, Scott Stingel wrote:

 On 6/22/2010 2:03 AM, Tzafrir Cohen wrote:

 On Mon, Jun 21, 2010 at 09:08:02AM -0700, Scott Stingel wrote:

  
 Hello-

 I have a system with one D410P and one B200P (both OpenVox).  All is
 well with the D410P, inbound and outbound, and I can initiate calls on
 the B200P  BRI span, but there may be something wrong with my inbound
 BRI setup:  there is no indication of an inbound call when I dial in to
 it from the PSTN.

 When I run pri intense debug and make a call to the BRI span, I can
 see a message containing the DDI that I'm dialing, in this case 336027
 (BT supplies only the last 6 digits of a delivered number).  See debug
 output below...




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How do I access the Dialstatus numeric code received?

[Tilghman Lesher]

On Tuesday 22 June 2010 07:32:00 CDR wrote:
 Tilghman Lesher wrote
  Not available in anything other than trunk (to be 1.8).  It depends upon
  a new
  feature, so it's not something you can easily backport.  After dialling,
  the SIP code is available in ${HASH(SIP_CAUSE,channel-name)}

 In a real dialplan, how do I get a variable with channel-name?

${HASHKEYS(SIP_CAUSE)} will deliver a list of all channel names which have set
a SIP cause.

 I mean: My 
 app is Sip2Sip. I get one call inbound, get the number dialed ${EXTEN} and
 proceed to try many carriers. If the carriers send me something different
 than 503 Service Unavailable or 404 Not Found, I need to close the call and
 send back whatever SIP code I got, exactly.

I don't think you can send back the same cause code, necessarily.  It would
depend upon the state of your calling channel.  Certainly if the calling
channel is already answered, the only thing you really can do is to drop the
call.  In any case, it's the PRI cause code that you would pass to the Hangup
function that would get mapped back to a SIP cause code.  The SIP cause in the
dialplan is really only useful for dialplan logic, not for passing back to the
calling channel.

 There is no way for me to do 
 that now. Unless I am missing something, I can only play with ${DIALSTATUS}
 and do Hangup(Code), but my Code variable is never the same that I got
 from the second leg. I would like to be able to do
 Hangup( ${HASH(SIP_CAUSE,channel-name)}, where channel-name is the
 last channel used to dial-out. How do I do this in trunk? I will have to
 start using trunk in production. Another issues is the the function
 Hangup(Code) takes a decimal, not related to the SIP code I just got. How
 would you design your 1.8 or 1.62 dialplan around this issue?
 Thanks in advance.

No, actually, the Hangup code is directly mapped to and from SIP codes.  There
are some less-specific cause codes (codes that get mapped from more than one
SIP code), but that's the best that you can get without using a real proxy.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Video not working with PortSIP SDK

[list mail]

Hi, I'm setup the Asterisk 1.4.33 and try test it with the PortSIP SDK(
www.portsip.com), but seems the video does not works.
When I make the call from PortSIP SDK Demo to GrandStream GXV3140, it's
working fine if no video codec selected.

If make call with H.264 codec, the PortSIP got 503 service unavailable
response from Asterisk, why?

Thanks
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Local channel usage

[Harel Cohen]

Zeeshan:
1. g option continues the dial plan after the called party hangup, and only the 
called party. See the manual or check for yourself...
2. h extension is no good for me because the voice path is already closed at 
this point therefore I cannot play IVR (Im getting Warnings like: file.c:750 
ast_readaudio_callback: Failed to write frame).
Tiago:
There is no Dial() option to simply continue dial-plan after Dial(). See above 
regarding g option.

Can anyone think of a way to play IVR after conversation initiated by Dial() 
terminates?

Harel
--

Message: 9
Date: Tue, 22 Jun 2010 07:27:42 -0400
From: Zeeshan Zakaria zisha...@gmail.com
Subject: Re: [asterisk-users] Local channel usage
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
aanlktilo2hzyq4jp7rb_iguzaq-n2chxnhq96grg0...@mail.gmail.com
Content-Type: text/plain; charset=windows-1252

'g' option continues the dial plan after the call has been answered, not
after it is hung up. Depending upon what you are trying to do, first try to
use h extension, i.e. in the example you gave, you should replace '_22,2'
with 'h,1'.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-06-22 6:23 AM, Tiago Geada tiago.ge...@gmail.com wrote:

Hi,

After a Dial, the call is hung up. It doesn't carry on with dialplan unless
you specify the appropriate dial option.

Check wiki voip-info for cmd Dial, I think the option is g

2010/6/22 Harel Cohen ha...@easycall.gi

 
  Hi All,
 
  I?m trying to do ?things? after my Dial application terminates (e.g. play
 IVR to cal...
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Sangoma - how to show channels in use?

[Jeff LaCoursiere]

Hi,

I have several 1.4.29 installations with Sangoma AFT101d cards.  Normally 
we have been collecting the raw data and then graphing channel use for 
these customers with:

asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l

Then I recently noticed that there were some zombie calls in this list 
that were not actually active anymore.  They go away if I restart 
asterisk, but in the meantime channel use appears artificially inflated.

I am wondering if there is a better method, perhaps with Sangoma CLI 
tools, to show which channels are ACTUALLY in use?  I played around with 
wanpipemon but that doesn't really give channel specific info.

Any clues?  I posted on the Sangoma forums also...

Thanks!

j



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[A J Stiles]

On Tuesday 22 Jun 2010, Scott Stingel wrote:
 Hi-
 I've been going through the same upgrade process recently, and had the
 same error (shown in your other message).  I had forgotten that the
 equipment I was plugged in to was CPE, so I had to change my new setting
 for that span to NET rather than CPE.  I notice in your old zapata
 files that you had CPE for two spans and NET for the other two, and your
 dahdi_chan setup is set up the same.  But I'm thinking perhaps during
 testing you plugged a CPE on your new setup to a CPE on the other, which
 would produce the symptoms you see.

On the current machine, spans 1 and 2 are the ISDN exchange lines  (they go to 
the box on the wall labelled NTE2D);  span 3 is connected to an Eicon Diva 
server card for fax sending  (but that's for another day .);  and span 4 
is available to use as though it was another exchange line  (used to be used 
for something once).  I'm not certain that span 2 actually does anything; it 
may have been turned off as a money-saving measure.  But the cable is still 
plugged in anyway.

I unplugged the cables from spans 1 and 2 of the old machine, and transferred 
them to the new machine, leaving 3 and 4 alone for the time being.

Next live testing I'll have to do tonight, once nobody else needs the phones.

-- 
AJS

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[marek cervenka]

 On 06/22/2010 04:38 PM, marek cervenka wrote:
 On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
 On 2010-06-22 12:36, Remco Bressers wrote:
 Dear list,

 I've got the following setup :

 [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]

 On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general].
 The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
 PBX WAN, i see the following in udptl debug :

 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
 Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
 Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)

 This means my outgoing udptl traffic is correctly translated, but
 somehow i'm sending 172.16.0.156 instead of my public IP address on the
 firewall.

 try asterisk 1.6.2.9

 What would be the reason to do that? Is there any change on this in 1.6.2.9?

yes
1.6.2.x branch is a lot better in T.38 area

---
Marek Cervenka
===


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma - how to show channels in use?

[Danny Nicholas]

Since you are already grepping, just add a grep -e zombie (you should
probably go ahead and do core show channels instead of show channels
since this will bite you at some time in the future).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, June 22, 2010 11:41 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Sangoma - how to show channels in use?


Hi,

I have several 1.4.29 installations with Sangoma AFT101d cards.  Normally 
we have been collecting the raw data and then graphing channel use for 
these customers with:

asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l

Then I recently noticed that there were some zombie calls in this list 
that were not actually active anymore.  They go away if I restart 
asterisk, but in the meantime channel use appears artificially inflated.

I am wondering if there is a better method, perhaps with Sangoma CLI 
tools, to show which channels are ACTUALLY in use?  I played around with 
wanpipemon but that doesn't really give channel specific info.

Any clues?  I posted on the Sangoma forums also...

Thanks!

j



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] when to use e1/t1 card?

[Steve Edwards]

On Tue, 22 Jun 2010, Zeeshan Zakaria wrote:

 If you are doing pure VoIP and no PRIs or PSTN, you don't need these 
 cards.

Just to clarify the acronyms...

PRI is Primary Rate Integrated Services Digital Network (usually 
delivering 24 (T1) or 31 (E1) channels over a 4 wire connection with 
connectors that look like RJ45 but are really RJ48).

POTS is Plain Old Telephone Service (usually delivering a single circuit 
over a 2 wire connection with connectors commonly known as RJ11). POTS 
circuits can be combined (or split up) into a T1 (or E1) using a device 
known as a channel bank.

PSTN is the Public Switched Telephone Network.

So, wouldn't it be more accurate to say no PRIs or POTS since both are 
used to connect to the PSTN?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Call file structure and syntax

[Mike Ely]

Hi there,

I¹ve been looking to do an outbound dialer for systems alerting, etc. and
have in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
Madsen,  Smith can I find a detailed discussion of what goes into a
callfile, how to get it to do things like interact with the shell (I¹d like
³Press 2² in my outbound call to do something of value), etc.  I¹ve googled
around but haven¹t found what I¹m looking for, just other people¹s ³Hello
World² callfiles.  As of now, I can make outbound calls well enough, but
want more...

Can someone point me in the right direction for this?

Thanks,
Mike
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Local channel usage

[Philipp von Klitzing]

Hi!

 Can anyone think of a way to play IVR after conversation initiated by
 Dial() terminates?

You will most probably have to prevent the hangup to happen in the first 
place: 

You could, for example, join the two callers by the help of a dynamic 
MeetMe room, and then take action when the other parties leaves, i.e. 
kick the remaining user out of the room and continue in the dialplan.

Here's an example for Voicemail live that uses such a technique:
http://www.voip-info.org/wiki/view/Asterisk+tips+voicemail+live

Another way *might* be to involve a local channel for the calling party 
with the /n option to prevent it from optimizing themselves away: For 
example: The caller's SIP channel hangs up, but the local channel that it 
is connected with then continues in the dialplan? Not sure if there is a 
way to make this work - could be that you need to twist things badly so 
that also the caller is in fact a callee to the local channel...

Finally: Put a SIP proxy in between that catches the hangup and then 
takes action like a redirect (transfer).

Philipp


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UDPTL T38 via NAT

[Miguel Amez]

Hi guys,

Has anyone of you configured succesfully T38 asterisk faxing with hylafax
and t38modem?
I'm very interested on this kind of configuration, because I have installed
it with asterisk and I haven't been able to get it work.

Any kind of info on this issue would be appreciated.

Regards,
Miguel Amez

2010/6/22 marek cervenka cerv...@fpf.slu.cz

  On 06/22/2010 04:38 PM, marek cervenka wrote:
  On 06/22/2010 02:51 PM, Johann Steinwendtner wrote:
  On 2010-06-22 12:36, Remco Bressers wrote:
  Dear list,
 
  I've got the following setup :
 
  [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP]
 
  On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in
 [general].
  The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the
  PBX WAN, i see the following in udptl debug :
 
  Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32)
  Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
  Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32)
  Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32)
  Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32)
  Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29)
 
  This means my outgoing udptl traffic is correctly translated, but
  somehow i'm sending 172.16.0.156 instead of my public IP address on
 the
  firewall.
 
  try asterisk 1.6.2.9
 
  What would be the reason to do that? Is there any change on this in
 1.6.2.9?

 yes
 1.6.2.x branch is a lot better in T.38 area

 ---
 Marek Cervenka
 ===


 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[Scott Stingel]

On 6/22/2010 9:44 AM, A J Stiles wrote:
 On Tuesday 22 Jun 2010, Scott Stingel wrote:

 Hi-
 I've been going through the same upgrade process recently, and had the
 same error (shown in your other message).  I had forgotten that the
 equipment I was plugged in to was CPE, so I had to change my new setting
 for that span to NET rather than CPE.  I notice in your old zapata
 files that you had CPE for two spans and NET for the other two, and your
 dahdi_chan setup is set up the same.  But I'm thinking perhaps during
 testing you plugged a CPE on your new setup to a CPE on the other, which
 would produce the symptoms you see.
  
 On the current machine, spans 1 and 2 are the ISDN exchange lines  (they go to
 the box on the wall labelled NTE2D);  span 3 is connected to an Eicon Diva
 server card for fax sending  (but that's for another day .);  and span 4
 is available to use as though it was another exchange line  (used to be used
 for something once).  I'm not certain that span 2 actually does anything; it
 may have been turned off as a money-saving measure.  But the cable is still
 plugged in anyway.

 I unplugged the cables from spans 1 and 2 of the old machine, and transferred
 them to the new machine, leaving 3 and 4 alone for the time being.

 Next live testing I'll have to do tonight, once nobody else needs the phones.


Yes, it sounds like you've configured it correctly, ie the same as the 
old machine, but just for fun you might try pri_net on one of the spans, 
stop and start the dahdi service and asterisk and see what happens!

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Internal timing bad for Fax?

[Kristijan Vrban]

Hello, i just made the reproducible watching:
I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via
T.38 - Audiocodes Mediant 2000 (FW 5.60.43.5) - PSTN Fax
With Internal timing Enabled, the Fax break after the first quarter
from the first page is transfered.
With Internal timing Disabled, the fax is transferred flawless.

Both test with pthread timing module on a QEMU Virtual maschine

So, is internal timing bad for Fax?

Kristijan

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Running SIP on non-standard ports

[Stephen Brown]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I want to have the ability to have anonymous SIP calls hit my server but
I want to run it on different ports and create an SRV record for my
target domain.

My understanding of SIP is limited, but in a nutshell I want to
accomplish the following:

- - run SIP signaling on port 6200
- - create RTP ports on 6201-62XX

Do I really need 10k ports open for RTP!!?? I don't plan on doing more
than a 5-10 calls simultaneously, maybe less than that. Does each RTP
port represent one channel, or does it take two, one for each end
perhaps? Just trying to come up with a number...

And my assumption is also that I will only need to create an SRV record
for the SIP signaling, will I need to create SRV record(s) for the RTP
ports as well? I'm assuming the SIP signaling handles that instead?

Thanks in advance...

Stephen
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (Darwin)

iEYEARECAAYFAkwg8nkACgkQ3sJXNEncx7i1LQCfcgefEgyBb4QC96dBe46dK6DA
EYUAoNcVLAt/lr4EzUslnXEzIJMTVt9h
=eis2
-END PGP SIGNATURE-

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call file structure and syntax

[Danny Nicholas]

#1 - once you've got to this point, AMI would be a better option than a call
file

#2 -  using AMI or a call file, you are going to want to use the
context-based method instead of application to get the most bang for your
buck

 

I use a bigger instance of this to play a message and accept 1 or 2 from the
user

; this context is used by AMI to play a message

[accept]

exten = s,1,Answer

exten = s,n,Background(important)

exten = s,n,WaitExten(5,m)

exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))

exten = 1,n,Background(${Data})

exten = 1,n,Background(repeatmsg)

exten = 1,n,WaitExten(5,m)

exten = 1,n,Hangup

exten = 2,1,Background(calllater)

exten = 2,n,ForkCDR(v,s(reject=${Data}))

exten = 2,n,Hangup

exten = 3,1,Goto(accept|1|2)

exten = *,1,Goto(accept|s|1)

exten = i,1,Goto(accept|s|1)

exten = t,1,Goto(accept|s|1)

 

here's the call file

Action = 'Originate',

  Channel = DAHDI/1,

  Variable = Data=/tmp/test.gsm,

  Exten = 'SIP/170',

  Context = 'accept',

  priority = 1,

  Number = 5551212

Using the accept context, 5551212 is called on DAHDI/1 and user hears
important.gsm.  then they press 1 to hear test.gsm or 2 to hear it later.

 

Hope this is helpful.

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely
Sent: Tuesday, June 22, 2010 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call file structure and syntax

 

Hi there,

I've been looking to do an outbound dialer for systems alerting, etc. and
have in large part followed the recipe here:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

That and the associated pages at voip-info give a basic set of recipes for
callfiles, but nowhere there or in my copy of the O'Reilly book by Meggelen,
Madsen,  Smith can I find a detailed discussion of what goes into a
callfile, how to get it to do things like interact with the shell (I'd like
Press 2 in my outbound call to do something of value), etc.  I've googled
around but haven't found what I'm looking for, just other people's Hello
World callfiles.  As of now, I can make outbound calls well enough, but
want more...

Can someone point me in the right direction for this?

Thanks,
Mike 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SkypeKit

[Jay R. Worthington]

http://www.engadget.com/2010/06/22/skypekit-beta-sdk-adds-skype-to-any-application-or-device/

Great! Finally a change to get a chan_skype without beeing a**-raped by the
copyprotection (which is the sole reason i didn't buy it), and maybe even
more than the absolute basic features (like Silk and video)...

J.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Running SIP on non-standard ports

[Danny Nicholas]

Can't really answer the rest of this, but you only need 40 ports open for 10
RTP calls (4 per call).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Sent: Tuesday, June 22, 2010 12:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Running SIP on non-standard ports

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I want to have the ability to have anonymous SIP calls hit my server but
I want to run it on different ports and create an SRV record for my
target domain.

My understanding of SIP is limited, but in a nutshell I want to
accomplish the following:

- - run SIP signaling on port 6200
- - create RTP ports on 6201-62XX

Do I really need 10k ports open for RTP!!?? I don't plan on doing more
than a 5-10 calls simultaneously, maybe less than that. Does each RTP
port represent one channel, or does it take two, one for each end
perhaps? Just trying to come up with a number...

And my assumption is also that I will only need to create an SRV record
for the SIP signaling, will I need to create SRV record(s) for the RTP
ports as well? I'm assuming the SIP signaling handles that instead?

Thanks in advance...

Stephen
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (Darwin)

iEYEARECAAYFAkwg8nkACgkQ3sJXNEncx7i1LQCfcgefEgyBb4QC96dBe46dK6DA
EYUAoNcVLAt/lr4EzUslnXEzIJMTVt9h
=eis2
-END PGP SIGNATURE-

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call file structure and syntax

[Mike Ely]

That¹s a good start.  In my case, I want it to dial a round-robin queue (set
up separately) and if the user presses 2, stop dialing the queue and log
which user acknowledged the alarm.  If the user presses 1, repeat the
message, if no key is pressed before a timeout, hang up and dial the next
user in the queue.  Or something like that.  One of the things I also want
to be able to do with this is echo out something to the shell, either a
textfile or an actual command so that I can trigger some other actions not
necessarily related to Asterisk.

It¹s a fun project except for the knowledge that successful completion is
going to mean it wakes me up some night at 3am.



On 6/22/10 10:31 AM, Danny Nicholas da...@debsinc.com wrote:

 #1 ­ once you¹ve got to this point, AMI would be a better option than a call
 file
 #2 -  using AMI or a call file, you are going to want to use the context-based
 method instead of application to get the most ³bang for your buck²
  
 I use a bigger instance of this to play a message and accept 1 or 2 from the
 user
 ; this context is used by AMI to play a message
 [accept]
 exten = s,1,Answer
 exten = s,n,Background(important)
 exten = s,n,WaitExten(5,m)
 exten = 1,1,ForkCDR(v,s(fullcmd=${Data}))
 exten = 1,n,Background(${Data})
 exten = 1,n,Background(repeatmsg)
 exten = 1,n,WaitExten(5,m)
 exten = 1,n,Hangup
 exten = 2,1,Background(calllater)
 exten = 2,n,ForkCDR(v,s(reject=${Data}))
 exten = 2,n,Hangup
 exten = 3,1,Goto(accept|1|2)
 exten = *,1,Goto(accept|s|1)
 exten = i,1,Goto(accept|s|1)
 exten = t,1,Goto(accept|s|1)
  
 here¹s the call file
 Action = 'Originate',
  Channel = DAHDI/1,
  Variable = Data=/tmp/test.gsm²,
  Exten = 'SIP/170',
  Context = 'accept',
  priority = 1,
  Number = 5551212
 Using the accept context, 5551212 is called on DAHDI/1 and user hears
 important.gsm.  then they press 1 to hear test.gsm or 2 to hear it later.
  
 Hope this is helpfulŠ
 
 
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Ely
 Sent: Tuesday, June 22, 2010 12:02 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Call file structure and syntax
  
 Hi there,
 
 I¹ve been looking to do an outbound dialer for systems alerting, etc. and have
 in large part followed the recipe here:
 http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
 
 That and the associated pages at voip-info give a basic set of recipes for
 callfiles, but nowhere there or in my copy of the O¹Reilly book by Meggelen,
 Madsen,  Smith can I find a detailed discussion of what goes into a callfile,
 how to get it to do things like interact with the shell (I¹d like ³Press 2² in
 my outbound call to do something of value), etc.  I¹ve googled around but
 haven¹t found what I¹m looking for, just other people¹s ³Hello World²
 callfiles.  As of now, I can make outbound calls well enough, but want more...
 
 Can someone point me in the right direction for this?
 
 Thanks,
 Mike 
 
 
 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver

[bruce bruce]

Thanks for the tips.

This is an 8 FXO channel Astribank. My understanding is that Trixbox 2.8
already had everything dahdi related installed so there is no driver from
Astribank. I followed this page and did rpm -Uvh for freepbx-module-zapauto

http://www.xorcom.com/downloads/astribank2-dahdi.html

What other steps do I have to take to complete the installation if you think
I have not finished. Or where can I look to find the problem?

As you could see from my last e-mail, everything seemed fine except for the
voltage part. I am SURE that PSTN lines are connected to the box.

Thanks

http://www.xorcom.com/downloads/astribank2-trixbox-ce-drivers.html#trixboxce2.8

On Tue, Jun 22, 2010 at 5:25 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote:
  Hi Guys,
 
  An 8 channel

 FXO?

  Astribank is connected to Trixbox 2.8 and I ran
  freepbx-module-zapauto but I get the following when running these
  commands and can't make calls out:
 
  [Trixbox]# dahdi_genconf xpporder
  /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
  '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc

 Which one, exactly? Trixbox originally had rather dates DAHDI drivers. I
 believe you should now be able to find much newer ones. At least in
 their repos.

 
  pbx*CLI dahdi show channels
  Chan Extension Context Language MOH Interpret Blocked State
 
  pseudo default default In Service
  1 from-pstn default In Service
  2 from-pstn default In Service
  3 from-pstn default In Service
  4 from-pstn default In Service
  5 from-pstn default In Service
  6 from-pstn default In Service
 
  7 from-pstn default In Service
  8 from-pstn default In Service
 
  pbx*CLI dahdi show status
  Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
  Xorcom XPD #00/00: FXO OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1)
 
 
  pbx*CLI dahdi show regdump 1
  Unable to get registers on channel 1
  Unable to get stats on channel 1

 I believe that regdump uses some specific interface to the Digium card.
 If you want a bunch of technical information you don't really
 understand, look under /proc/xpp/ . I'm not going to start explaining it
 beyond

 http://docs.tzafrir.org.il/dahdi-linux/README.Astribank.html#_proc_interface

 
  [Trixbox]# dahdi_hardware -v
  /usr/sbin/dahdi_hardware: warning - OLD DRIVER: missing
  '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc
 
  usb:002/002 xpp_usb+ e4e4:1162 Astribank-modular FPGA-firmware
  LABEL=[usb:X1044207] connect...@usb-:00:1d.7-4
  XBUS-00/XPD-00: FXO (8) Span 1 DAHDI-SYNC
 
 
  Note: This Astribank is deployed in United Arab Emirates and I am not
  sure what the line type is in terms of Ground or Loop start and
  wondering if that makes a difference with the Astribank and the fact
  that it can't how the voltage using show regdump

 IIRC they use LS (That is: no power denial is used at the end of a
 call).

 
  And I am definitly not sure what that warning of OLD DRIVER is about.
  Any help is appreciated.

 At the time we wrote it, we relied heavily on procfs. However procfs is
 not something that can be safely used when the module is loaded or
 removed. This is a fine way to get panics. Thus we gradually moved many
 things from /proc to /sys .

 Typically such a message would mean a combination of older dahdi-linux
 (loaded) and newer dahdi-tools. Though it's a warning, and not an error.

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[James Lamanna]

If you've used Linksys phones against recent Asterisk 1.4.x you may
have noticed
that they may drop registration for a quick bit and then go back to being ok
if your phone is behind NAT.
If you turn Asterisk's sip debug information on, you'll probably find
errors like these in your logs:

NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
received from '11 sip:999...@208.90.186.10'

I believe I have determined that this is caused by a bug in the
Linksys firmware that is related to the NAT Keep-Alive packets.
Because recent Asterisk 1.4.x's do not establish a SIP dialog for
NOTIFY requests, the 489 Bad Event
replies were going back to the wrong address if your phone was behind NAT.
This lack of reply would cause the next REGISTER message to use the
same nonce as the previous REGISTER,
resulting in the stale nonce errors and temporarily dropping
registration. I've also seen the lack of response to
the NAT keep-alive cause the phone to stop being able to register
completely as well.

Below I've posted a patch that responds with a 200 OK to these
keep-alive requests, and I believe
also solves the temporary loss of registration problem, though more
testing in different environments
for those who experience this problem would be greatly appreciated.

The patch is against 1.4.32.

-- James


keep_alive_fix.diff
Description: Binary data
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk distribution for a Call Center

[Alejandro Cabrera Obed]

Dear all, I need to build a PBX based on Asterisk for a call center. I
have worked with raw Asterisk but it's hard to work for big
implementations think.

Also I have worked with Trixbox CE for a small bussines and it was
prette good, but I have not have many features like ACD. I know there
is another  version called Trixbox PRO -specially Call Center edition-
that's not free but has got more features like ACD and billing.

I've heart about AsteriskNow and I know it's free.

What distribution/version do you recommend to me in order to implement
a call center and taking into account I'm not an expert in programming
from Asterisk CLI ???

Thanks a lot

Alejandro

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Endless loop with asterisk directory

[Douglas Mortensen]

Every so often, I have an asterisk 1.4.22-4 system that goes into an endless 
loop with the following:

[Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch' 
(escape_digits=) (sample_offset 0)
[Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing 'dir-intro' (language 'en')
[Jun 1 13:30:45] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch' 
(escape_digits=) (sample_offset 0)
[Jun 1 13:30:45] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:45] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:45] VERBOSE[13160] logger.c: -- Playing 'dir-intro' (language 'en')
[Jun 1 13:30:45] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch' 
(escape_digits=) (sample_offset 0)
[Jun 1 13:30:45] WARNING[13160] file.c: Failed to write frame
[Jun 1 13:30:45] WARNING[13160] file.c: Failed to write frame

Etc...

This happens until the file system fills up with a huge asterisk/full log, and 
then asterisk crashes.

A lot more detail on the issue is here:
http://www.freepbx.org/forum/freepbx/users/asterisk-directory-looping

The FreePBX project-lead has said that this is clearly an asterisk issue, and 
has nothing to do with them 
(http://www.freepbx.org/forum/freepbx/beta-program-issues/loop-when-the-call-is-not-answered-by-the-extension#comment-27032).

Anyone here seen this before? Any ideas as to how I can get this issue 
resolved, short of just entirely disabling the app_directory.so ?

Other people seeming to have the same problem here 
(http://www.trixbox.org/forums/trixbox-forums/help/trixbox-2-6-0-7-directory-not-working-looping),
 but none of the suggested solutions from either thread has worked for me.


Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCP, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Internal timing bad for Fax?

[Kevin P. Fleming]

On 06/22/2010 12:24 PM, Kristijan Vrban wrote:
 Hello, i just made the reproducible watching:
 I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via
 T.38 - Audiocodes Mediant 2000 (FW 5.60.43.5) - PSTN Fax
 With Internal timing Enabled, the Fax break after the first quarter
 from the first page is transfered.
 With Internal timing Disabled, the fax is transferred flawless.
 
 Both test with pthread timing module on a QEMU Virtual maschine
 
 So, is internal timing bad for Fax?

No, but you've chosen a very bad combination to provide timing;
res_timing_pthread uses a great deal of CPU to do its job (much more
than the other options), and you are running in a virtual machine on top
of that. If you are running on Linux and can run a kernel that supports
res_timing_timerfd, you'll have better results, but running in a virtual
machine will always means that you are subject to random
scheduling-related problems.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[Stefan Schmidt]

James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James
   
Hello,

you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY 
to $OPTIONS and make this extension in your default context:
exten = s,1,hangup

and you also would get a 200 ok for the keep alive package.

IMHO a stale nonce would only occur when a user tries to register faster 
than 3600s cause of the register timeout used in asterisk. Maybe you 
should also try to set a higher register timeout on your phones. but i 
dont have an 1.4 system running, only around 2k of linksys phones on a 
1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

best regards.

steve

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Edwin Quijada]

The best option JUST ASTERISK without anything else.

Maybe you need hire somebody with expereince with callcenter.

*---* 
*-Edwin Quijada 
*-Developer DataBase 
*-JQ Microsistemas 
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*




 
 Date: Tue, 22 Jun 2010 15:21:18 -0300
 From: aco1...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Asterisk distribution for a Call Center
 
 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.
 
 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.
 
 I've heart about AsteriskNow and I know it's free.
 
 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???
 
 Thanks a lot
 
 Alejandro
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
  
_

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[bruce bruce]

Thanks Tiago and Tzafrir. I agree with the heavy load that Tzafrir
mentioned. I already made a phpagi that does a system() for asterisk -rx and
it's not very responsive at time.

So what is the solution guys?

You see, I only want to know if g729 is being used because I want to
determine if a trunk is being used or not. Now, don't be hasty and suggest
GROUP_COUNT to me as I can not use that because I can only see the calls by
sip show peers or core show channels and group show channels doesn't
show me any channels because I do not have control over the calls place as
they are placed by A2Billing.

Any more Gurus want to weigh in more?


On Tue, Jun 22, 2010 at 6:42 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
  Hi!
 
  If it was me, I would create a bash script calling asterisk -vrx core
 show
  commands
 
  something like:
 
  for chan in $(asterisk -vrx core show channels concise);
  do
  asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
  native;
  done

 The overhead of each 'asterisk -rx' command is noticable. If you have 10
 calls or more, this can have an odd effect.

 Not to mention that the fact that it is so slow exposes its raciness[1].

 
  On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:
 
   Hi Everyone,
  
   I want to know if a specific codec type is used at least one. For
 example,
   I want to know if out of the 100 calls on the system if there is a 1
 channel
   that is running G.729 codec right now. If using dial-plan and I dial
 in, I
   can use this to obtain information about CURRENT channel. But it won't
 allow
   me to obtain information about OTHER channels and that is what I want
 to do.
   I want a search for all channels and an output spit out as g729 or TRUE
 or
   FALSE if there is a g729 channel.
  
   exten = s,1,Answer()
   exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
   exten = s,n,NoOp(${foo})
  
   Above  NoOp spits out g729 if I call in with a g729 codec. But I
 want that to be about other channels and not the one I am calling into.
  
   Thanks,
  
   Bruce

 [1] Which should naturally be fixed using locks :-)

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Andrew Latham]

oops, missed part...

make  make install  make samples  make install-logrotate; cd ..
perl -pi -e s/exit 0/\/usr\/sbin\/safe_asterisk\nexit 0/g /etc/rc.local
cd addons-1.6.2  ./configure  make  make install; cd ..
cd gui-2.0  ./configure  make  make install; cd ..
mkdir /usr/src/asterisk-sound
cd /usr/src/asterisk-sound


~
Andrew lathama Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Tue, Jun 22, 2010 at 3:20 PM, Andrew Latham lath...@gmail.com wrote:
 debian lenny

 aptitude install openssh-server
 aptitude install -y build-essential subversion autoconf
 linux-headers-`uname -r` /
 ncurses-dev mc libgmime2-dev libsnmp-dev libiksemel-dev /
 vim-full libxml2-dev libmysqlclient15-dev tcpdump unzip mysql-client ntp rsync

 #!/bin/bash
 cd /usr/src
 svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 asterisk-1.6.2
 svn co http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4
 svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 gui-2.0
 svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 addons-1.6.2
 svn co http://svn.digium.com/svn/dahdi/linux/branches/2.3 dahdi-linux-2.3
 svn co http://svn.digium.com/svn/dahdi/tools/branches/2.3 dahdi-tools-2.3
 cd dahdi-linux-2.3  make  make install; cd ..
 cd dahdi-tools-2.3  ./configure  make  make install  make config; cd 
 ..
 cd libpri-1.4  make  make install; cd ..
 cd asterisk-1.6.2  ./configure


 should be a good start...

 edit http.conf and manager.conf and you have the fantastic GUI!
 ~
 Andrew lathama Latham
 lath...@gmail.com

 * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
 * Learn more about Linux http://en.wikipedia.org/wiki/Linux
 * Learn more about Tux http://en.wikipedia.org/wiki/Tux



 On Tue, Jun 22, 2010 at 2:21 PM, Alejandro Cabrera Obed
 aco1...@gmail.com wrote:
 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Andrew Latham]

debian lenny

aptitude install openssh-server
aptitude install -y build-essential subversion autoconf
linux-headers-`uname -r` /
ncurses-dev mc libgmime2-dev libsnmp-dev libiksemel-dev /
vim-full libxml2-dev libmysqlclient15-dev tcpdump unzip mysql-client ntp rsync

#!/bin/bash
cd /usr/src
svn co http://svn.digium.com/svn/asterisk/branches/1.6.2 asterisk-1.6.2
svn co http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4
svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 gui-2.0
svn co http://svn.digium.com/svn/asterisk-addons/branches/1.6.2 addons-1.6.2
svn co http://svn.digium.com/svn/dahdi/linux/branches/2.3 dahdi-linux-2.3
svn co http://svn.digium.com/svn/dahdi/tools/branches/2.3 dahdi-tools-2.3
cd dahdi-linux-2.3  make  make install; cd ..
cd dahdi-tools-2.3  ./configure  make  make install  make config; cd ..
cd libpri-1.4  make  make install; cd ..
cd asterisk-1.6.2  ./configure


should be a good start...

edit http.conf and manager.conf and you have the fantastic GUI!
~
Andrew lathama Latham
lath...@gmail.com

* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux



On Tue, Jun 22, 2010 at 2:21 PM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] joining 2 conferences together

[Daniel Knoll]

Is it possible to join 2 meetme conferences (each on different server) 
together, that if i load balance the callers, they can see altogether 
something like a inter system communikation ?

Thanx for your help.
Daniel
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[Elliot Otchet]

Get it via the AMI.  If you're already using PHPAGI, it is trivial to get this 
data.  You can even find an example of how to call sip show peers and output 
the resulting response.  You avoid using the (-rx) and you get the data you 
were looking for.

http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI_AsteriskManager.html

other thoughtsIf you're already using PHPAGI often on a busy system, you 
might want to get more ram, use fastagi to move the PHP load to another system, 
or take Steve Edward's standard advice and rewrite it in C.  /other thoughts

-Elliot

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Tuesday, June 22, 2010 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL 
function be used to retrieve info about OTHER channels?

Thanks Tiago and Tzafrir. I agree with the heavy load that Tzafrir mentioned. I 
already made a phpagi that does a system() for asterisk -rx and it's not very 
responsive at time.

So what is the solution guys?

You see, I only want to know if g729 is being used because I want to determine 
if a trunk is being used or not. Now, don't be hasty and suggest GROUP_COUNT to 
me as I can not use that because I can only see the calls by sip show peers 
or core show channels and group show channels doesn't show me any channels 
because I do not have control over the calls place as they are placed by 
A2Billing.

Any more Gurus want to weigh in more?

On Tue, Jun 22, 2010 at 6:42 AM, Tzafrir Cohen 
tzafrir.co...@xorcom.commailto:tzafrir.co...@xorcom.com wrote:
On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
 Hi!

 If it was me, I would create a bash script calling asterisk -vrx core show
 commands

 something like:

 for chan in $(asterisk -vrx core show channels concise);
 do
 asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
 native;
 done
The overhead of each 'asterisk -rx' command is noticable. If you have 10
calls or more, this can have an odd effect.

Not to mention that the fact that it is so slow exposes its raciness[1].


 On 21 June 2010 16:08, bruce bruce 
 bruceb...@gmail.commailto:bruceb...@gmail.com wrote:

  Hi Everyone,
 
  I want to know if a specific codec type is used at least one. For example,
  I want to know if out of the 100 calls on the system if there is a 1 channel
  that is running G.729 codec right now. If using dial-plan and I dial in, I
  can use this to obtain information about CURRENT channel. But it won't allow
  me to obtain information about OTHER channels and that is what I want to do.
  I want a search for all channels and an output spit out as g729 or TRUE or
  FALSE if there is a g729 channel.
 
  exten = s,1,Answer()
  exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
  exten = s,n,NoOp(${foo})
 
  Above  NoOp spits out g729 if I call in with a g729 codec. But I want 
  that to be about other channels and not the one I am calling into.
 
  Thanks,
 
  Bruce
[1] Which should naturally be fixed using locks :-)

--
  Tzafrir Cohen
icq#16849755  
jabber:tzafrir.co...@xorcom.commailto:jabber%3atzafrir.co...@xorcom.com
+972-50-7952406   
mailto:tzafrir.co...@xorcom.commailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  
iax:gu...@local.xorcom.com/tzafrirhttp://iax:gu...@local.xorcom.com/tzafrir

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



This message is intended only for the use of the individual (s) or entity to 
which it is addressed and may contain information that is privileged, 
confidential, and/or proprietary to Calling Circles LLC and its affiliates. If 
the reader of this message is not the intended recipient, you are hereby 
notified that any dissemination, distribution, forwarding or copying of this 
communication is prohibited without the express permission of the sender. If 
you have received this communication in error, please notify the sender 
immediately and delete the original message.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[bruce bruce]

Thanks for the input. If this is doable via Asterisk AMI why not through
dial-plan? I mean it only makes sense to be possible through dial-plan where
all access is given as well just like the AMI. Am I wrong with this?

On Tue, Jun 22, 2010 at 4:01 PM, Elliot Otchet 
elliot.otc...@callingcircles.com wrote:

  Get it via the AMI.  If you’re already using PHPAGI, it is trivial to get
 this data.  You can even find an example of how to call “sip show peers” and
 output the resulting response.  You avoid using the (-rx) and you get the
 data you were looking for.



 http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI_AsteriskManager.html



 other thoughtsIf you’re already using PHPAGI often on a busy system, you
 might want to get more ram, use fastagi to move the PHP load to another
 system, or take Steve Edward’s standard advice and rewrite it in C.  /other
 thoughts



 -Elliot



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
 *Sent:* Tuesday, June 22, 2010 1:32 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL
 function be used to retrieve info about OTHER channels?



 Thanks Tiago and Tzafrir. I agree with the heavy load that Tzafrir
 mentioned. I already made a phpagi that does a system() for asterisk -rx and
 it's not very responsive at time.



 So what is the solution guys?



 You see, I only want to know if g729 is being used because I want to
 determine if a trunk is being used or not. Now, don't be hasty and suggest
 GROUP_COUNT to me as I can not use that because I can only see the calls by
 sip show peers or core show channels and group show channels doesn't
 show me any channels because I do not have control over the calls place as
 they are placed by A2Billing.



 Any more Gurus want to weigh in more?



 On Tue, Jun 22, 2010 at 6:42 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
 wrote:

 On Tue, Jun 22, 2010 at 11:25:29AM +0100, Tiago Geada wrote:
  Hi!
 
  If it was me, I would create a bash script calling asterisk -vrx core
 show
  commands
 
  something like:
 
  for chan in $(asterisk -vrx core show channels concise);
  do
  asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
  native;
  done

 The overhead of each 'asterisk -rx' command is noticable. If you have 10
 calls or more, this can have an odd effect.

 Not to mention that the fact that it is so slow exposes its raciness[1].


 
  On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:
 
   Hi Everyone,
  
   I want to know if a specific codec type is used at least one. For
 example,
   I want to know if out of the 100 calls on the system if there is a 1
 channel
   that is running G.729 codec right now. If using dial-plan and I dial
 in, I
   can use this to obtain information about CURRENT channel. But it won't
 allow
   me to obtain information about OTHER channels and that is what I want
 to do.
   I want a search for all channels and an output spit out as g729 or TRUE
 or
   FALSE if there is a g729 channel.
  
   exten = s,1,Answer()
   exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
   exten = s,n,NoOp(${foo})
  
   Above  NoOp spits out g729 if I call in with a g729 codec. But I
 want that to be about other channels and not the one I am calling into.
  
   Thanks,
  
   Bruce

 [1] Which should naturally be fixed using locks :-)


 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

 --
 _

 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



 --
 This message is intended only for the use of the individual (s) or entity
 to which it is addressed and may contain information that is privileged,
 confidential, and/or proprietary to Calling Circles LLC and its affiliates.
 If the reader of this message is not the intended recipient, you are hereby
 notified that any dissemination, distribution, forwarding or copying of this
 communication is prohibited without the express permission of the sender. If
 you have received this communication in error, please notify the sender
 immediately and delete the original message.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 

[asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

[bruce bruce]

Hi Everyone,

I was on Xorocom site but there is no clear and consice place to download
drivers and firmware. I am reading their instructions to install Astribank 8
channel FXO on Trixbox 2.8 and I seem to be missing files at this step:

[pbx.archology.com dahdi]# /usr/share/doc/astribank_upgrade
/usr/share/dahdi/
-bash: /usr/share/doc/astribank_upgrade: No such file or directory


Where the heck are these files on their site?

It's really a bugger when a manufacturer can't organize a site
nicelysigh

Thanks guys
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Lenz Emilitri]

It really depends on how large your CC will be and how much money is at
stake. :-)
We have a lot of clients who are very satisfied with small call centers
based on FreePBX or Trixbox CE. Of course I would not implement a 500-seats
call center out of a standard CD.
My suggestion is: make sure you have an experienced local consultant handy
in case something goes wrong - in real life, it always does.
Just my two eurocents,
l.

2010/6/22 Alejandro Cabrera Obed aco1...@gmail.com

 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Loway - home of QueueMetrics - http://queuemetrics.com
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] joining 2 conferences together

[Paul Belanger]

On Tue, Jun 22, 2010 at 3:31 PM, Daniel Knoll dan...@danielknoll.de wrote:
 Is it possible to join 2 meetme conferences (each on different server) 
 together, that if i load balance the callers, they can see altogether 
 something like a inter system communikation ?

You can IAX2 asterisk boxes together, but each conference will be
hosted on there respective server.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

[Steve Edwards]

On Tue, 22 Jun 2010, bruce bruce wrote:

 I was on Xorocom site but there is no clear and consice place to 
 download drivers and firmware. I am reading their instructions to 
 install Astribank 8 channel FXO on Trixbox 2.8 and I seem to be missing 
 files at this step:

[snip]

 Where the heck are these files on their site?

 It's really a bugger when a manufacturer can't organize a site 
 nicelysigh

Based on the professionalism I've always seen from Tzafrir this struck me 
as odd so I thought I'd take a look...

What's hard about:

1) Hover over Support

2) Select Upgrades  Downloads

3) Click on Astribank Drivers

Seemed pretty obvious to me. Am I missing something?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Gerardo Barajas]

Or you can try Elastix and it's Call Center Module, but as Lenz says it is
suitable for small call centers.
And pay for support consultancy.

On Tue, Jun 22, 2010 at 3:40 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:

 It really depends on how large your CC will be and how much money is at
 stake. :-)
 We have a lot of clients who are very satisfied with small call centers
 based on FreePBX or Trixbox CE. Of course I would not implement a 500-seats
 call center out of a standard CD.
 My suggestion is: make sure you have an experienced local consultant handy
 in case something goes wrong - in real life, it always does.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Anybody using TE410P on BT ISDN with DAHDI?

[A J Stiles]

Right.  I think I might be getting somewhere.

First I commented out all the lines relating to spans 2, 3 and 4 in 
my /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf, and set up a 
very minimal dialplan in /etc/asterisk/extensions.conf  (just 2 extensions).

Then I connected up just span 1  (which I know works, because it's been 
working fine with the old setup)  and started Asterisk.

Each extension managed to call the other OK.  Good so far.  And no warnings 
about missing D-channels.  Looking promising.  I even managed to call out -- 
but not back in, because my dialplan was incomplete.  One quick edit later, 
and I had inbound calls ringing both extensions.


Next I tried uncommenting just span 2 in  /etc/dahdi/system.conf 
and /etc/asterisk/chan_dahdi.conf, though without the cable plugged into the 
card.  And I got:

[Jun 22 21:12:00] WARNING[4175]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Jun 22 21:12:00] WARNING[4176]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 47 as D-channel anyway!

Plugging in span 2 made it work.  Unplugging span 2 made it not work:

[Jun 22 21:17:23] NOTICE[4616]: chan_dahdi.c:12690 pri_dchannel: PRI got 
event: HDLC Abort (6) on Primary D-channel of span 2
[Jun 22 21:17:34] ERROR[4616]: chan_dahdi.c:12389 dahdi_pri_error: PTP MDL 
can't handle error of type I
[Jun 22 21:17:34] ERROR[4616]: chan_dahdi.c:12389 dahdi_pri_error: MDL-ERROR 
(I): T200 = N200 in timer recovery state 8
[Jun 22 21:17:36] ERROR[4615]: chan_dahdi.c:12389 dahdi_pri_error: PTP MDL 
can't handle error of type I
[Jun 22 21:17:36] ERROR[4615]: chan_dahdi.c:12389 dahdi_pri_error: MDL-ERROR 
(I): T200 = N200 in timer recovery state 8
  == Primary D-Channel on span 2 down
[Jun 22 21:17:38] WARNING[4616]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 47 as D-channel anyway!
  == Primary D-Channel on span 1 down
[Jun 22 21:17:40] WARNING[4615]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 16 as D-channel anyway!
[Jun 22 21:17:42] WARNING[4616]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 47 as D-channel anyway!


So, as far as I can tell, the important thing is:  it doesn't like having 
spans uncommented in the config files that aren't connected to anything:  
even the ones that are connected to something, don't work.  In fact, even 
after commenting-out the unwanted lines and restarting DAHDI, I get:

[Jun 22 21:34:18] WARNING[5651]: chan_dahdi.c:4160 pri_find_dchan: No 
D-channels available!  Using Primary channel 16 as D-channel anyway!

and then I get

  == Primary D-Channel on span 1 up

after which, it works!  Calls between extensions, and in and out via the ISDN.  

Now I seem to be getting somewhere, at least.  Next step will be to go away 
for awhile and write a proper dialplan!

-- 
AJS

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Xorocom Missing files...where to get it? astribank_upgrade

[bruce bruce]

What about after step 3? That is where the messy instructions begin. I am
not trying to bash but I just had to resort to google to find  this which is
not included in the Trixbox 2.8 installation instructions:

wget http://svn.digium.com/svn/dahdi/tools/trunk/xpp/astribank_upgrade
chmod +x astribank_upgrade

There is a disconnect in steps and there is flow. That is what bugs me.
Anyone can find the support link on that page. I was talking about the
general status of installation instructions. I don't know why it is so hard
to do a 1,2,3 install and DONE, specially when they can separate Trixbox
from Elastix and version to version.


On Tue, Jun 22, 2010 at 4:53 PM, Steve Edwards asterisk@sedwards.comwrote:

 On Tue, 22 Jun 2010, bruce bruce wrote:

  I was on Xorocom site but there is no clear and consice place to
  download drivers and firmware. I am reading their instructions to
  install Astribank 8 channel FXO on Trixbox 2.8 and I seem to be missing
  files at this step:

 [snip]

  Where the heck are these files on their site?
 
  It's really a bugger when a manufacturer can't organize a site
  nicelysigh

 Based on the professionalism I've always seen from Tzafrir this struck me
 as odd so I thought I'd take a look...

 What's hard about:

 1) Hover over Support

 2) Select Upgrades  Downloads

 3) Click on Astribank Drivers

 Seemed pretty obvious to me. Am I missing something?

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[adriano ghezzi]

I suggest
asterisk
+free pbx
+astercc
ideal till 15 seats.
you have queue agents and acd
hth

Adriano.


2010/6/22 Alejandro Cabrera Obed aco1...@gmail.com:
 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] joining 2 conferences together

[daniel]

Hi Paul, 
Yes, i can use iax2, but this is rather a redirect to another server as 
connecting 2 confernce channels from 2 different server.
Can i join 2 dahdi (meetme) channels from different servers?

Regards Daniel 
--Originalnachricht--
Von: Paul Belanger
Absender:asterisk-users-boun...@lists.digium.com
An:Asterisk Users Mailing List - Non-Commercial Discussion
Antwort an:Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] joining 2 conferences together
Gesendet: 22. Jun. 2010 22:29

On Tue, Jun 22, 2010 at 3:31 PM, Daniel Knoll dan...@danielknoll.de wrote:
 Is it possible to join 2 meetme conferences (each on different server) 
 together, that if i load balance the callers, they can see altogether 
 something like a inter system communikation ?

You can IAX2 asterisk boxes together, but each conference will be
hosted on there respective server.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SMS in landline

[Tiago Geada]

Hi all.

I am searching for a way to send SMS via our E1 PRI line.

We are in Portugal and I have seen some Internet/TV/Phone providers (ZON for
those who know it) who install normal phones with SMS support in landline.

So I just found a page from PT (Portugal Telecom) stating that the SMC
number is either 12999 or 129990 (
http://www.ptcom.pt/PTResidencial2/Tabs/MyPTPublico/Apoio_a_Clientes/Servi%C3%A7os/SMS/caracteristicas/sms_caracteristicas.htm
)

Now I was trying to send a SMS via a PRI from PT (same provider)

context of dialplan is services

[r...@asterisk ~]# tail /etc/asterisk/extensions_services.ael -n 12
_00019 = { // TEST SMS
Noop(Testing SMS to ${EXTEN:4}...);
Answer();
SMS(services,,00351932485457,bla);
SMS(services);
Hangup();
//  129990
}

/ FINISHED TESTING /

}
 [r...@asterisk ~]# cat test.call
Channel: DAHDI/g7/12999
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: services
Extension: 0001932485457
Priority: 1
SetVar: MSG=hello


cp test.call /var/spool/asterisk/outgoing/  chown asterisk.asterisk
/var/spool/asterisk/outgoing/test.call  chmod 777
/var/spool/asterisk/outgoing/test.call  asterisk -vvr

Asterisk 1.6.2.9-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-rc2 currently running on asterisk (pid = 12521)
Verbosity is at least 14
-- Attempting call on DAHDI/g7/12999 for 0001932485...@services:1 (Retry 1)
-- Making new call for cr 32792
-- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=28
 Call Ref: len= 2 (reference 24/0x18) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
 Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
 (16)
User information layer 1: A-Law (35)
 [18 03 a1 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  
 Dchan: 0
ChanSel: As indicated in following octets
   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
   Ext: 1  Channel: 1 ]
 [6c 02 21 80]
 Calling Number (len= 4) [ Ext: 0  TON: National Number (2)  NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user number 
 not screened (0)  '' ]
 [70 06 a1 31 32 39 39 39]
 Called Number (len= 8) [ Ext: 1  TON: National Number (2)  NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '12999' ]
 [a1]
 Sending Complete (len= 1)
q931.c:3134 q931_setup: call 32792 on channel 1 enters state 1 (Call Initiated)
 Protocol Discriminator: Q.931 (8)  len=32
 Call Ref: len= 2 (reference 24/0x18) (Terminator)
 Message type: CALL PROCEEDING (2)
 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0
ChanSel: As indicated in following octets
   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
   Ext: 1  Channel: 1 ]
 [28 14 43 48 41 4d 41 44 41 20 45 4d 20 50 52 4f 47 52 45 53 53 4f]
 Display (len=20) [ CHAMADA EM PROGRESSO ]
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 40 (cs0, Display)
q931.c:3683 q931_receive: call 32792 on channel 1 enters state 3
(Outgoing call  Proceeding)
 Protocol Discriminator: Q.931 (8)  len=52
 Call Ref: len= 2 (reference 24/0x18) (Terminator)
 Message type: DISCONNECT (69)
 [08 02 84 9c]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Public network serving the remote user (4)
  Ext: 1  Cause: Invalid number format (28), class =
Normal Event (1) ]
 [1c 17 91 a1 14 02 01 2e 02 01 24 30 0c 30 0a a1 05 30 03 02 01 00 82 01 00]
 Facility (len=25, codeset=0) [ 0x91, 0xA1, 0x14, 0x02, 0x01, '.',
0x02, 0x01, '$0', 0x0C, '0', 0x0A, 0xA1, 0x05, '0', 0x03, 0x02, 0x01,
0x00, 0x82, 0x01, 0x00 ]
PROTOCOL 11
A1 0014 (CONTEXT SPECIFIC [1])
  02 0001 2E (INTEGER: 46)
  02 0001 24 (INTEGER: 36)
  30 000C (SEQUENCE)
30 000A (SEQUENCE)
  A1 0005 (CONTEXT SPECIFIC [1])
30 0003 (SEQUENCE)
  02 0001 00 (INTEGER: 0)
  82 0001 00 (CONTEXT SPECIFIC [2])
 [1e 02 82 88]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT 

Re: [asterisk-users] joining 2 conferences together

[Paul Belanger]

On Tue, Jun 22, 2010 at 5:47 PM,  dan...@danielknoll.de wrote:
 Can i join 2 dahdi (meetme) channels from different servers?

No
-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] joining 2 conferences together

[Danny Nicholas]

What you CAN do (depending on your hardware) is to call meetme1 then
conference that call into meetme2.  I've done this to join 2 DAHDI calls on
1.4.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Tuesday, June 22, 2010 4:56 PM
To: dan...@danielknoll.de; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] joining 2 conferences together

On Tue, Jun 22, 2010 at 5:47 PM,  dan...@danielknoll.de wrote:
 Can i join 2 dahdi (meetme) channels from different servers?

No
-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Tarek Sawah]

i have been struggling with call center Customers for a couple of years now.. i 
have a call center with 40 agents using elastix.. and quality is related to the 
source of calls inbound or outbound... the problem with call centers they need 
Visual .. like Flash Operator panel and CDRs.. if you can go with simply raw 
asterisk .. without any additions.. will be the best for you .. write your own 
dial plans.Flash operator Panel is not a flawless work.. and adds more burden 
on the resources.. esp when it's open by 7-8 persons at once.. regarding the 
ACD ..it's all about PHP and Database .. you can build your own reports and so. 
or you can use a2billing to do the billing and ACD.. Elastix has a good billing 
(without a2billing) .. but i prefer a clean installation of asterisk and work 
around with database and PHP much better.. Good Luck!

-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +1 (386) 492-9993


 Date: Tue, 22 Jun 2010 15:21:18 -0300
 From: aco1...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Asterisk distribution for a Call Center
 
 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.
 
 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another  version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.
 
 I've heart about AsteriskNow and I know it's free.
 
 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???
 
 Thanks a lot
 
 Alejandro
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
  
_
Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox.
http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[James Lamanna]

On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James

 Hello,

 you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY
 to $OPTIONS and make this extension in your default context:
 exten = s,1,hangup

 and you also would get a 200 ok for the keep alive package.

 IMHO a stale nonce would only occur when a user tries to register faster
 than 3600s cause of the register timeout used in asterisk. Maybe you
 should also try to set a higher register timeout on your phones. but i
 dont have an 1.4 system running, only around 2k of linksys phones on a
 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

I'm not sure how this works.
The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS
message never gets processed.
The options message I receive from a Linksys942 6.1.3(a) looks like this:

--- SIP read from xxx.xxx.xxx.xxx:8037 ---
OPTIONS
-

-- James

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 1.4.33.1 Released

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 
1.4.33.1.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.33.1 resolves a regression involving the use 
of FXO signaling in chan_dahdi where a channel could continue ringing 
after it has been answered.

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33.1

Thank you for your continued support of Asterisk!

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk distribution for a Call Center

[Luciano Moreira]

We use Vicidial for all size CallCenter. It's very powerful for multi
server and/or multi site. We have vicidial from tiny callcenter one
site with 5 agents to over 1000 Agents distributed in 20 cities
working as just one callcenter.

Info http://astguiclient.sourceforge.net/vicidial.html

__
Luciano Moreira

Logic Telecom LTDa
Fortaleza, CE

+55 (85) 4062-9150
+55 (85) 9701-2444
+1 360-717-1506 (USA)



2010/6/22 Tarek Sawah tareksa...@hotmail.com:
 i have been struggling with call center Customers for a couple of years
 now.. i have a call center with 40 agents using elastix.. and quality is
 related to the source of calls inbound or outbound...
 the problem with call centers they need Visual .. like Flash Operator panel
 and CDRs..
 if you can go with simply raw asterisk .. without any additions.. will be
 the best for you .. write your own dial plans.
 Flash operator Panel is not a flawless work.. and adds more burden on the
 resources.. esp when it's open by 7-8 persons at once..
 regarding the ACD ..it's all about PHP and Database .. you can build your
 own reports and so. or you can use a2billing to do the billing and ACD..
 Elastix has a good billing (without a2billing) .. but i prefer a clean
 installation of asterisk and work around with database and PHP much
 better..
 Good Luck!

 -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1
 (386) 492-9993


 Date: Tue, 22 Jun 2010 15:21:18 -0300
 From: aco1...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Asterisk distribution for a Call Center

 Dear all, I need to build a PBX based on Asterisk for a call center. I
 have worked with raw Asterisk but it's hard to work for big
 implementations think.

 Also I have worked with Trixbox CE for a small bussines and it was
 prette good, but I have not have many features like ACD. I know there
 is another version called Trixbox PRO -specially Call Center edition-
 that's not free but has got more features like ACD and billing.

 I've heart about AsteriskNow and I know it's free.

 What distribution/version do you recommend to me in order to implement
 a call center and taking into account I'm not an expert in programming
 from Asterisk CLI ???

 Thanks a lot

 Alejandro

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
 http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

 
 Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox.
 Learn more.
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[Ryan Wagoner]

On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James

 Hello,

 you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY
 to $OPTIONS and make this extension in your default context:
 exten = s,1,hangup

 and you also would get a 200 ok for the keep alive package.

 IMHO a stale nonce would only occur when a user tries to register faster
 than 3600s cause of the register timeout used in asterisk. Maybe you
 should also try to set a higher register timeout on your phones. but i
 dont have an 1.4 system running, only around 2k of linksys phones on a
 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

 I'm not sure how this works.
 The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS
 message never gets processed.
 The options message I receive from a Linksys942 6.1.3(a) looks like this:

 --- SIP read from xxx.xxx.xxx.xxx:8037 ---
 OPTIONS
 -

 -- James

 --

I had the same result when using $OPTIONS on a SPA941 phone with
firmware 5.1.8. I am running Asterisk 1.6.2.9 that has the keep-alive
support, however I still see Asterisk sending a 489 Bad Event. I just
reopened the issue and provided the necessary debug log at
https://issues.asterisk.org/bug_view_page.php?bug_id=17379

Ryan

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sangoma - how to show channels in use?

[Jeff LaCoursiere]

On Tue, 22 Jun 2010, Danny Nicholas wrote:

 Since you are already grepping, just add a grep -e zombie (you should
 probably go ahead and do core show channels instead of show channels
 since this will bite you at some time in the future).

True.  Its an old script ;)  But I used the zombie term adjectively - 
there is no zombie text in the output.  I just know that a call is not 
still ringing hours after it was initially placed.  Not sure how it is 
getting into that state... here is an example excerpt:

Zap/5-1  18666902...@from-pst Ringing AppDial((Outgoing Line))
SIP/7157787-08331ec8 18666902...@resident RingDial(Zap/g0/18666902511)
Zap/3-1  18666902...@from-pst Ringing AppDial((Outgoing Line))
SIP/7157787-08335df0 18666902...@resident RingDial(Zap/g0/18666902511)
Zap/2-1  18666902...@from-pst Ringing AppDial((Outgoing Line))
SIP/7157787-b6d28360 18666902...@resident RingDial(Zap/g0/18666902511)

It kind of looks like this one SIP endpoint tried to make the same call 
three times in a row without success, and all of the calls show as still 
active, though I know they are not (in fact they show as still ringing).
So are channels 2, 3, and 5 actually still busy from the telco's 
perspective because asterisk is keeping them open?  That would suck.  A 
lot.

I did get a reply from Sangoma, who basically said that their driver 
doesn't know about the individual channels - that is totally handled by 
asterisk.

So it seems there is no way other than what I am already doing to judge 
the channels in use?

Thanks,

j




 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
 LaCoursiere
 Sent: Tuesday, June 22, 2010 11:41 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Sangoma - how to show channels in use?


 Hi,

 I have several 1.4.29 installations with Sangoma AFT101d cards.  Normally
 we have been collecting the raw data and then graphing channel use for
 these customers with:

 asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l

 Then I recently noticed that there were some zombie calls in this list
 that were not actually active anymore.  They go away if I restart
 asterisk, but in the meantime channel use appears artificially inflated.

 I am wondering if there is a better method, perhaps with Sangoma CLI
 tools, to show which channels are ACTUALLY in use?  I played around with
 wanpipemon but that doesn't really give channel specific info.

 Any clues?  I posted on the Sangoma forums also...

 Thanks!

 j



 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] realtime queues membername problem

[Jean Chassoul]

anyone know something about this?

On Fri, May 14, 2010 at 10:56 AM, Jean Chassoul chass...@gmail.com wrote:

 Hi,

 I'm using dynamic realtime with asterisk 1.6.0.24, I'm having a strange
 problem with queue_members...

 If I update only 'membername' field on queue_members table asterisk won't
 refresh the change, but if I update another field like interface everything
 works as expected, I've found this problem also deleting a existing agent on
 queue_members and then inserting a new one with the same interface, penalty
 and pause but with another membername :( Asterisk won't refresh the change
 and show the old membername on CLI  (queue show my-queue...).

 It is possible that asterisk refresh these info?

 Thanks.


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)

[James Lamanna]

On Tue, Jun 22, 2010 at 4:31 PM, Ryan Wagoner rswago...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 6:26 PM, James Lamanna jlama...@gmail.com wrote:
 On Tue, Jun 22, 2010 at 12:06 PM, Stefan Schmidt s...@sil.at wrote:
 James Lamanna schrieb:
 If you've used Linksys phones against recent Asterisk 1.4.x you may
 have noticed
 that they may drop registration for a quick bit and then go back to being 
 ok
 if your phone is behind NAT.
 If you turn Asterisk's sip debug information on, you'll probably find
 errors like these in your logs:

 NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce
 received from '11 sip:999...@208.90.186.10'

 I believe I have determined that this is caused by a bug in the
 Linksys firmware that is related to the NAT Keep-Alive packets.
 Because recent Asterisk 1.4.x's do not establish a SIP dialog for
 NOTIFY requests, the 489 Bad Event
 replies were going back to the wrong address if your phone was behind NAT.
 This lack of reply would cause the next REGISTER message to use the
 same nonce as the previous REGISTER,
 resulting in the stale nonce errors and temporarily dropping
 registration. I've also seen the lack of response to
 the NAT keep-alive cause the phone to stop being able to register
 completely as well.

 Below I've posted a patch that responds with a 200 OK to these
 keep-alive requests, and I believe
 also solves the temporary loss of registration problem, though more
 testing in different environments
 for those who experience this problem would be greatly appreciated.

 The patch is against 1.4.32.

 -- James

 Hello,

 you also just could set the NAT KEEP ALIVE MESSAGE on Ext 1 from $NOTIFY
 to $OPTIONS and make this extension in your default context:
 exten = s,1,hangup

 and you also would get a 200 ok for the keep alive package.

 IMHO a stale nonce would only occur when a user tries to register faster
 than 3600s cause of the register timeout used in asterisk. Maybe you
 should also try to set a higher register timeout on your phones. but i
 dont have an 1.4 system running, only around 2k of linksys phones on a
 1.2.40 and 300 on 1.6.1.18 and i dont see this problem there.

 I'm not sure how this works.
 The OPTIONS message fails chan_sip.c:parse_request() so the OPTIONS
 message never gets processed.
 The options message I receive from a Linksys942 6.1.3(a) looks like this:

 --- SIP read from xxx.xxx.xxx.xxx:8037 ---
 OPTIONS
 -

 -- James

 --

 I had the same result when using $OPTIONS on a SPA941 phone with
 firmware 5.1.8. I am running Asterisk 1.6.2.9 that has the keep-alive
 support, however I still see Asterisk sending a 489 Bad Event. I just
 reopened the issue and provided the necessary debug log at
 https://issues.asterisk.org/bug_view_page.php?bug_id=17379

Ryan,
This is most likely because the packet never makes it to handle_request_notify.
I haven't looked at the code for 1.6.2.9 yet, but in 1.4.32 without my
patch, the
NOTIFY request would never make it out of find_call() and return early with a
489 Bad Event response.

Were you getting any response at 1.6.2.9 with the OPTIONS message?

-- James

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users