[asterisk-users] compile Dahdi !

[lizhong zhu]

hello, all of users:
i want to test the dahdi with asterisk-1.6, but there is no much source for 
this new project. the only information i got is from voip-info. my problem is 
that,  i can not enable the chan_zap, therefore i do not have chan_zap.so in 
asterisk/modules, i can not remane it to chan_dahdi.so. please give a details 
for that.
regards!
zhu

   
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Re: [asterisk-users] Codec and CPU load

[Gordon Henderson]

On Tue, 26 Aug 2008, aymen warfalli wrote:


 Hi

 as maximum link capacity could be calculated using codecs and channel 
 types so , regarding the CPU and processors load , Is there any formula 
 or (any relations could help ) that can give the maximum CPU load 
 (mainly processor and RAM ) or scalability average using asterisk 
 channels , codecs , applications ?.

The easy answer to this is: No.

Far too many variables. Not just cpu speed, but cpu cache size, memory 
type  speed, external interfaces (over a PCI bus/TDM or Ethernet/SIP/IAX) 
other tasks the CPU might be doing, compile-time optimisations, kernel 
optimisations, etc.

As a generalisation, without transcoding you'll be able to manage 100's of 
calls on a modern Intel or AMD server platform, but all bets are off when 
you turn on transcoding, echo cancellation, etc.

Gordon

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Re: [asterisk-users] Need application, CID number match list to call cell phone

[Gordon Henderson]

On Tue, 26 Aug 2008, JR Richardson wrote:

 Hi All,

 I received a request for a special application and need some guidance.
 Cust has there own Asterisk PBX with SIP phones, pretty standard
 setup.

 They want an after hours application that checks inbound caller ID
 numbers and matches them to a list, say 5 to 10 numbers of special VIP
 customers, if there is a match on the list, then forward the call
 straight to a cell phone, instead of ringing local extension and then
 to voicemail.

 The customer also wants to be able to manage this VIP list and the
 call forward cell phone number themselves, so it needs to be
 configured, numbers added and deleted, through a web page on the PBX.

 So I'm thinking I need a dialplan app that has to interface with a
 MySQL database that holds the list of numbers, so I can build a
 webpage to add/delete the numbers.

 Any ideas would be much appreciated.

From the Keep It Simple Stupid Department:

Use the astDB. Add entries like:

   /vipMap/custNumber: Target number

eg.

   /vipMap/441364698123: 07712191046

In this case, 441364123123 is my VIP customers number, and 07712191046 is 
my mobile number..

You can do a little bit of webby stuff to maintain this via the manager 
interface. If you can't find any, drop me an email - I have some I 
modified after finding some ideas online. (I use this way to manage 100's 
of number to name mappings)

To use:

   exten = s,n,Noop(Testing for VIP - caller is ${CALLERID(number)})
   exten = s,n,Set(vipTarget=${DB(vipMap/${CALLERID(number)})})
   exten = s,n,GotoIf($[${vipTarget} = ]?notVip)
   exten = s,n,Noop(We got a VIP - lets call their contact on ${vipTarget})
   exten = s,n,Dial(SIP/${OUT}/{vipTarget}) ; or whatever

   exten = s,n(notVip),Noop(Not a VIP - dump to answering machine)
   ...


Gordon

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Re: [asterisk-users] asterisk realtime

[Szasz Szabolcs]

Hi list! Thank for the help. Now, I can call the 8500 to listen to the 
inbound messages, change pin, but I have another problem. When I call a 
SIP extension configured in the MySQL database it says: Call from '101' 
to extension '102' rejected because extension not found. My vmusers 
table: 
+--+-+-+-+--+--+--+---+-+
 
| uniqueid | customer_id | context | mailbox | password | fullname | 
email | pager | stamp | 
+--+-+-+-+--+--+--+---+-+
 
| 1 | 101 | default | 101 | 264241 | | [EMAIL PROTECTED] | NULL | 2008-08-12 
11:59:34 | | 2 | 102 | default | 102 | NULL | | [EMAIL PROTECTED] | NULL | 
2008-08-12 11:59:40 | 
+--+-+-+-+--+--+--+---+-+
 
sipusers table: 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
| name | username | type | secret | host | callerid | context | mailbox 
| nat | qualify | fromuser | authuser | fromdomain | insecure | 
canreinvite | disallow | allow | restrictcid | defaultip | ipaddr | port 
| regseconds | 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
| 101 | 101 | friend | NULL | home | NULL | default | 101 | yes | no | 
101 | NULL | home | NULL | no | NULL | NULL | NULL | home | home | 5060 
| NULL | | 102 | 102 | friend | NULL | home | NULL | default | 102 | yes 
| no | 102 | NULL | home | NULL | no | NULL | NULL | NULL | home | home 
| 5060 | NULL | 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
Can you see the problem? Please help. Szasz Szabolcs 
-- Message: 16 Date: Mon, 25 Aug 2008 
10:23:31 -0500 From: Tilghman Lesher [EMAIL PROTECTED] 
Subject: Re: [asterisk-users] asterisk realtime To: Asterisk Users 
Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com Message-ID: 
[EMAIL PROTECTED] Content-Type: 
text/plain; charset=iso-8859-1 On Monday 25 August 2008 07:08:30 Szasz 
Szabolcs wrote:

  Hi!
  I am running CentOS 5 with Asterisk 1.4.21.2  I am trying to setup storage
  of voicemail messages into MySQL. I installed unixODBC unixODBC-devel
  libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured
  and built Asterisk, using menuconfig to turn on ODBC voicemail storage. 
  Here is the output of some config files:
 
  [MySQL]
  Description = ODBC for MySQL
  Driver  = /usr/lib/libmyodbc3.so
  Setup   = /usr/lib/libodbcmyS.so
  UsageCount  = 3
 
  [MySQL ODBC 3.51 Driver]
  Description = ODBC 3.51 for MySQL
  DRIVER  = /usr/lib/libmyodbc3.so
  SETUP   = /usr/lib/libmyodbc3S.so
  UsageCount  = 3
 
  [EMAIL PROTECTED] ~]# cat /usr/local/etc/odbc.ini
  [astrealtime]
  Description = MySQL Asterisk database
  Trace   = Off
  TraceFile   = stderr
  Driver  = MySQL
  SERVER  = localhost
  USER= asterisk
  PASSWORD= 123qwe
  PORT= 3306
  DATABASE= asterisk
 
  [EMAIL PROTECTED] ~]# cat /etc/asterisk/res_odbc.conf
  ;;; odbc setup file
 
  ; ENV is a global set of environmental variables that will get set.
  ; Note that all environmental variables can be seen by all connections,
  ; so you can't have different values for different connections.
  [ENV]
  INFORMIXSERVER = my_special_database
  INFORMIXDIR = /opt/informix
 
  ; All other sections are arbitrary names for database connections.
 
  [asterisk]
  enabled = yes
  dsn = astrealtime
  username = asterisk
  password = 123qwe
  pre-connect = yes
 
 
  ;[mysql2]
  ;enabled = no
  ;dsn = MySQL-asterisk
  ;username = myuser
  ;password = mypass
  ;pre-connect = yes
  ;
  ; On some databases, the connection times out and a reconnection will be
  ; necessary.  This setting configures the amount of time a connection
  ; may sit idle (in seconds) before a reconnection will be attempted.
  ;idlecheck = 3600
 
  ; Certain servers, such as MS SQL Server and Sybase use the TDS protocol,
  which ; limits the number of active queries per connection to 1.  By
  setting up pools ; of connections, Asterisk can be made to work with these
  servers. ;[sqlserver]
  ;enabled = no
  ;dsn = mickeysoft
  ;pooling = yes
  ;limit = 5
  ;username = oscar
  ;password = thegrouch
  ;pre-connect = yes
  ; Many databases have a 

Re: [asterisk-users] Digium Coffee anyone? PCI Expresso? WTF?

[Anthony Messina]

On Tuesday 26 August 2008 11:44:42 pm Karl Fife wrote:
 I'll be that none of the other coffee makers can handle anywhere NEAR 60
 voice channels, and don't get me started about HPEC!

 http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759

Good find! Does it grind it's own beans?

-- 
Anthony -  http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] compile Dahdi !

[Tzafrir Cohen]

On Wed, Aug 27, 2008 at 02:09:27PM +0800, lizhong zhu wrote:
 hello, all of users:
 i want to test the dahdi with asterisk-1.6, but there is no much 
 source for this new project. the only information i got is from 
 voip-info. my problem is that,  i can not enable the chan_zap, 
 therefore i do not have chan_zap.so in asterisk/modules, i can not 
 remane it to chan_dahdi.so. please give a details for that.

chan_dahdi (of both 1.4 SVN and 1.6.0 and trunk) works with DAHDI.
chan_zap no longer exists there.

In 1.4 SVN (soon to be released as 1.4.22) works with either zaptel or
dahdi. Note that this is set at build time.

(Self plug: I normally use the script from http://bugs.digium.com/11680
for simple test installations)

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

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Re: [asterisk-users] is shared_lastcall available in 1.4

[Atis Lezdins]

On Tue, Aug 26, 2008 at 7:26 PM, Bob Pierce [EMAIL PROTECTED] wrote:

 On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
  Are there any plans to back port this feature into upcoming 1.4
  releases?
 

 No, new features are added only in trunk, and released in next major
 release (1.6).

 So what would be involved in back porting this feature for our system?

 Do I simply follow the diff from the link you provided and apply the
 highlighted changes to the app_queue.c file in my Asterisk source
 directory before recompiling?



Generally yes. There's a patch file you can download for automatic
patching, but in this case it doesn't work automatically. So you
manually have to look all pieces that doesn't merge. I already took a
look, and hardest part would be update_status function, because
Asterisk 1.6 uses astobj2 (ao2_lock, ao2_iterator_* and other
functions) to access queue list. You will have to rewrite this part
using old functions - you can see that in update_queues function:

AST_LIST_LOCK(queues);
AST_LIST_TRAVERSE(queues, q, list) {
ast_mutex_lock(q-lock);

If you doubt about some part, you're welcome to ask, i'll try to help
you, but i don't want to provide complete backport to you, as i won't
be able to test it :)


Regards,
Atis


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

[Lee, John (Sydney)]

 
 Doesn't Queuemetrics run on a license basis?
 Anything else that's probably open source and free?


Does anyone have any comments/experience about using asteriskguru queue
statistics?
http://www.asteriskguru.com/tutorials/installation_guide.html


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[asterisk-users] asterisk-1.6, Remote-Party-ID Header not sent

[Alexander Zielke]

Hi List,

I recently switched to asterisk-1.6-beta9 because of the RPID support, 
but ran into the Problem, that the RPID-Header is not sent.


sendrpid is set to yes in my sip.conf, and i'm even sure that the 
add_header() function is called in chan_sip.c, but when i capture the 
SIP-Packets, which are sent out to the Phone, there is no 
Remote-Party-ID header in them.


I haven't found any reference on what could be causing this behavior. 
Anyone else has an idea?


--
with regards,
Alexander Zielke
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Re: [asterisk-users] DUNDI Help

[ronald ramos]

Hi Again,

Is there a way i can detect whether a user has been added into the regcontext?
Currently i'm seeing this and just gives a fast busy.

[Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel 
'SIP/10..10.10.10-b63101d0' sent into invalid extension '141100' in context 
'lookupdundi', but no invalid handler

can i detect it somehow, so i can inform user that the extensions is not 
available?

i have tried ChanIsAvail, but since i am using  realtime ChanIsAvail thinks it 
registered, since it really is registered on the other server. So it's trying 
to call it,  tries  it for 30 secs (i set it to timeout at 30),  after 30 secs 
then it will go to DUNDI/priv.  Is there a way that i can detect it first so it 
does not try to dial it on the local before askng dundi? thank you

regards,
Ron


--- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:
From: Bruce Reeves [EMAIL PROTECTED]
Subject: Re: [asterisk-users] DUNDI Help
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 8:16 PM

It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.

On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
wrote:
 Hi Bruce,

 my apologies, but the error was because of the key.
 i just run keys init on the CLI and it works,

 question on regcontext though, i set it to sipregistrations, how often
does
 an extension be added to the context sipregistrations and for how long
will
 it stay there? i'm looking at dialplan show sipregistration, sometimes
i
 only see one extension there. even though i know i have 4 ip phones
 registered to the asterisk.

 TIA

 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 6:23 PM

 Ron,

 What does the peers section in dundi.conf look like?

 On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos
[EMAIL PROTECTED]
 wrote:
 Would like to try setting up dundi with 3-4 asterisk.
 But for poc, i would like to try setting up dundi on between 2
asterisk.

 I copied the config from DUNDI enterprise SIP with no password. Only
thing
 i
 changed is the part where i used regcontext.
 on both boxes dundi.conf i have
 [mapping]
 priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

 i can see both peers on each server:
 CLI dundi show  peers
 EID  HostModel  AvgTime  Status
 00:8e:8c:8e:cb:5310.10.10.XX  (S) Symmetric  Unavail  OK (1 ms)

 i can see my extension being added
  on sipregistrations context
 Added extension '136101' priority 1 to sipregistrations

 tried a dundi lookup but got no result
 dundi lookup [EMAIL PROTECTED]
 DUNDi lookup returned no results.
 DUNDi lookup completed in 0 ms

 here's what's on extensions.conf

 ; Private DUNDi network
 [dundi-priv-canonical]
 ; Direct numbers

 [dundi-priv-customers]
 ; If you are an ITSP or Reseller, list your customers here.

 [dundi-priv-via-pstn]
 ; If you are freely delivering calls to the PSTN, list them here

 [dundi-priv-local]
 include = dundi-priv-canonical
 include = dundi-priv-customers
 include = dundi-priv-via-pstn

 [dundi-priv-switch]
 ; Just a wrapper for the switch
 switch = DUNDi/priv

 [dundi-priv-lookup]
 include =
  dundi-priv-local
 include = dundi-priv-switch

 [macro-dundi-priv]
 exten = s,1,Goto(${ARG1}|1)
 include = dundi-priv-lookup

 [diallocal]
 exten = _1X,1,Macro(dundi-priv|${EXTEN})

 i also tried dialing from my xlite:
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Macro(SIP/138100-08269548, dundi-priv|136101)
in
 new stack
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Goto(SIP/138100-08269548, 136101|1) in new
stack
 [Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
 [Aug 26 15:58:07]   == Auto fallthrough, channel
 'SIP/138100-08269548'
 status is 'UNKNOWN'

 any guess what's wrong? Thanks

 ron


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 --
 *
 Bruce Reeves, dCAp
 EUS Networks
 Office: 212-624-5943
 Web: www.euscorp.com
 


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[asterisk-users] sip show peers from shell or from CLI

[Olivier]

Hello,

On a 1.2 Asterisk / Debian Sarge, I noticed that :

ipbx*CLI sip show peers
Name/username  HostDyn Nat ACL Port Status
4201/4201  192.168.100.111  D  5060 OK (8 ms)
4200/4200  192.168.100.110  D  5060 OK (8 ms)

but from shell, I've got

# asterisk -rx sip show peers
on
Name/username  HostDyn Nat ACL Port Status
4201/4201  192.168.100.111  D  5060 OK (6 ms)
4200/4200  192.168.100.110  D  5060 OK (9 ms)


I never noticed this on word before.
Can anyone explain ?

I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e

Regards
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Re: [asterisk-users] sip show peers from shell or from CLI

[Steven Howes]

Probably another left over word from another message. Is it repeatable?

On 27 Aug 2008, at 13:00, Olivier wrote:

 Hello,

 On a 1.2 Asterisk / Debian Sarge, I noticed that :

 ipbx*CLI sip show peers
 Name/username  HostDyn Nat ACL Port Status
 4201/4201  192.168.100.111  D  5060 OK  
 (8 ms)
 4200/4200  192.168.100.110  D  5060 OK  
 (8 ms)

 but from shell, I've got

 # asterisk -rx sip show peers
 on
 Name/username  HostDyn Nat ACL Port Status
 4201/4201  192.168.100.111  D  5060 OK  
 (6 ms)
 4200/4200  192.168.100.110  D  5060 OK  
 (9 ms)


 I never noticed this on word before.
 Can anyone explain ?

 I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e

 Regards
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Re: [asterisk-users] sip show peers from shell or from CLI

[Olivier]

A closer inspection shows ^@ between on and Name as if these letters came
from a word previously cut (from connexion ?)s o shell command would show
# asterisk -rx sip show peers
on
[EMAIL PROTECTED]/username  HostDyn Nat ACL Port 
Status
4201/4201  192.168.100.111  D  5060 OK (6 ms)
4200/4200  192.168.100.110  D  5060 OK (9 ms)

When passing this to grep, grep replies it got binary data.
Strange, isn't ?


2008/8/27 Olivier [EMAIL PROTECTED]

 Hello,

 On a 1.2 Asterisk / Debian Sarge, I noticed that :

 ipbx*CLI sip show peers
 Name/username  HostDyn Nat ACL Port Status
 4201/4201  192.168.100.111  D  5060 OK (8 ms)
 4200/4200  192.168.100.110  D  5060 OK (8 ms)

 but from shell, I've got

 # asterisk -rx sip show peers
 on
 Name/username  HostDyn Nat ACL Port Status
 4201/4201  192.168.100.111  D  5060 OK (6 ms)
 4200/4200  192.168.100.110  D  5060 OK (9 ms)


 I never noticed this on word before.
 Can anyone explain ?

 I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e

 Regards

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Re: [asterisk-users] sip show peers from shell or from CLI

[Olivier]

2008/8/27 Steven Howes [EMAIL PROTECTED]

 Probably another left over word from another message. Is it repeatable?

At the moment, yes.

Now, I'm looking for a way to flush input/output, to protect shell script
from this type of side effect.




 On 27 Aug 2008, at 13:00, Olivier wrote:

  Hello,
 
  On a 1.2 Asterisk / Debian Sarge, I noticed that :
 
  ipbx*CLI sip show peers
  Name/username  HostDyn Nat ACL Port Status
  4201/4201  192.168.100.111  D  5060 OK
  (8 ms)
  4200/4200  192.168.100.110  D  5060 OK
  (8 ms)
 
  but from shell, I've got
 
  # asterisk -rx sip show peers
  on
  Name/username  HostDyn Nat ACL Port Status
  4201/4201  192.168.100.111  D  5060 OK
  (6 ms)
  4200/4200  192.168.100.110  D  5060 OK
  (9 ms)
 
 
  I never noticed this on word before.
  Can anyone explain ?
 
  I'm using Asterisk 1.2.17-BRIstuffed-0.3.0-PRE-1y-e
 
  Regards
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Re: [asterisk-users] sip show peers from shell or from CLI

[Steven Howes]

On 27 Aug 2008, at 13:23, Olivier wrote:
 2008/8/27 Steven Howes [EMAIL PROTECTED]
 Probably another left over word from another message. Is it  
 repeatable?
 At the moment, yes.

 Now, I'm looking for a way to flush input/output, to protect shell  
 script from this type of side effect.

[EMAIL PROTECTED] asterisk]#  asterisk -rx sip show peers
 -- Remote UNIX connection
Name/username  HostDyn Nat ACL Port Status


I get that on mine, every time. Guess its your machine not catching up  
in time to print that bit.. Might be possible to suppress the output  
of that somehow?

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Re: [asterisk-users] Need application, CID number match list to call cell phone

[JR Richardson]

 Is this a one VIP to one cell number match? Or is it on VIP to multiple
 cells?
 
 On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED]
 wrote:
  Hi All,
 
  I received a request for a special application and need some guidance.
   Cust has there own Asterisk PBX with SIP phones, pretty standard
  setup.
 
  They want an after hours application that checks inbound caller ID
  numbers and matches them to a list, say 5 to 10 numbers of special VIP
  customers, if there is a match on the list, then forward the call
  straight to a cell phone, instead of ringing local extension and then
  to voicemail.
 
  The customer also wants to be able to manage this VIP list and the
  call forward cell phone number themselves, so it needs to be
  configured, numbers added and deleted, through a web page on the PBX.
 
  So I'm thinking I need a dialplan app that has to interface with a
  MySQL database that holds the list of numbers, so I can build a
  webpage to add/delete the numbers.
 
[JR Richardson] 
The info I have is one cell phone, like an on-call cell that gets passed
around to on-call individuals.  But being able to change this number to a
different cell from time to time is required.

Thanks.

JR


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Re: [asterisk-users] Codec and CPU load

[Miguel Otamendi]

Hi

There are some tools that you may hold serve, check these link:

http://www.bandcalc.com/
http://codec-calculator.softonic.com/mac

Miguel Otamendi


2008/8/27 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]


 On Tue, 26 Aug 2008, aymen warfalli wrote:

 
  Hi
 
  as maximum link capacity could be calculated using codecs and channel
  types so , regarding the CPU and processors load , Is there any formula
  or (any relations could help ) that can give the maximum CPU load
  (mainly processor and RAM ) or scalability average using asterisk
  channels , codecs , applications ?.

 The easy answer to this is: No.

 Far too many variables. Not just cpu speed, but cpu cache size, memory
 type  speed, external interfaces (over a PCI bus/TDM or Ethernet/SIP/IAX)
 other tasks the CPU might be doing, compile-time optimisations, kernel
 optimisations, etc.

 As a generalisation, without transcoding you'll be able to manage 100's of
 calls on a modern Intel or AMD server platform, but all bets are off when
 you turn on transcoding, echo cancellation, etc.

 Gordon

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Re: [asterisk-users] DUNDI Help

[Bruce Reeves]

Sure, let me show you how I setup dundi on systems.

extensions.conf

exten = _1X,1,Goto(lookupdundi,${EXTEN},1)

[lookupdundi]
exten = _X,1,Goto(${ARG1},1)
switch = DUNDi/priv

exten = i,1,Playback(invalid)

You can have the i do whatever you want, and you can use the same
option in the macro you are using.

That is it, I leave out all the other context in the examples, from
time to time I add a dundi-static context and put in specific numbers
or patterns I want to accept, maybe for pstn calling or phones that
don't register, but in those cases I have multiple mappings in
dundi.conf for each context. For example:

priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial
priv = dundi-static,0,SIP,[EMAIL PROTECTED],nopartial




On Wed, Aug 27, 2008 at 3:56 AM, ronald ramos [EMAIL PROTECTED] wrote:
 Hi Again,

 Is there a way i can detect whether a user has been added into the
 regcontext?
 Currently i'm seeing this and just gives a fast busy.

 [Aug 27 16:44:46] WARNING[17402]: pbx.c:2483 __ast_pbx_run: Channel
 'SIP/10.10.10.10-b63101d0' sent into invalid extension '141100' in context
 'lookupdundi', but no invalid handler

 can i detect it somehow, so i can inform user that the extensions is not
 available?

 i have tried ChanIsAvail, but since i am using  realtime ChanIsAvail thinks
 it registered, since it really is registered on the other server. So it's
 trying to call it,  tries  it for 30 secs (i set it to timeout at 30),
 after 30 secs then it will go to DUNDI/priv.  Is there a way that i can
 detect it first so it does not try to dial it on the local before askng
 dundi? thank you

 regards,
 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED] wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 8:16 PM

 It is added when a phone registers, or re-registers. Depending on the
 timing of the registrations and any restarts on the asterisk process
 it may take some time for phones to re-register.

 On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED]
 wrote:
 Hi Bruce,

 my apologies, but the error was because of the key.
 i just run keys init on the CLI and it works,

 question
  on regcontext though, i set it to sipregistrations, how often
 does
 an extension be added to the context sipregistrations and for how long
 will
 it stay there? i'm looking at dialplan show sipregistration, sometimes
 i
 only see one extension there. even though i know i have 4 ip phones
 registered to the asterisk.

 TIA

 Ron


 --- On Tue, 8/26/08, Bruce Reeves [EMAIL PROTECTED]
 wrote:

 From: Bruce Reeves [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] DUNDI Help
 To: [EMAIL PROTECTED], Asterisk Users Mailing List -
 Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 6:23 PM

 Ron,

 What does the peers section in dundi.conf look like?

 On Tue, Aug 26, 2008 at 3:00 AM, ronald
  ramos
 [EMAIL PROTECTED]
 wrote:
 Would like to try setting up dundi with 3-4 asterisk.
 But for poc, i would like to try setting up dundi on between 2
 asterisk.

 I copied the config from DUNDI enterprise SIP with no password. Only
 thing
 i
 changed is the part where i used regcontext.
 on both boxes dundi.conf i have
 [mapping]
 priv = sipregistrations,0,SIP,[EMAIL PROTECTED],nopartial

 i can see both peers on each server:
 CLI dundi show  peers
 EID  HostModel  AvgTime  Status
 00:8e:8c:8e:cb:5310.10.10.XX  (S) Symmetric  Unavail  OK (1 ms)

 i can see my extension being added
  on sipregistrations context
 Added extension '136101' priority 1 to
  sipregistrations

 tried a dundi lookup but got no result
 dundi lookup [EMAIL PROTECTED]
 DUNDi lookup returned no results.
 DUNDi lookup completed in 0 ms

 here's what's on extensions.conf

 ; Private DUNDi network
 [dundi-priv-canonical]
 ; Direct numbers

 [dundi-priv-customers]
 ; If you are an ITSP or Reseller, list your customers here.

 [dundi-priv-via-pstn]
 ; If you are freely delivering calls to the PSTN, list them here

 [dundi-priv-local]
 include = dundi-priv-canonical
 include = dundi-priv-customers
 include = dundi-priv-via-pstn

 [dundi-priv-switch]
 ; Just a wrapper for the switch
 switch = DUNDi/priv


  [dundi-priv-lookup]
 include =
  dundi-priv-local
 include = dundi-priv-switch

 [macro-dundi-priv]
 exten = s,1,Goto(${ARG1}|1)
 include = dundi-priv-lookup

 [diallocal]
 exten = _1X,1,Macro(dundi-priv|${EXTEN})

 i also tried dialing from my xlite:
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Macro(SIP/138100-08269548, dundi-priv|136101)
 in
 new stack
 [Aug 26 15:58:07] -- Executing [EMAIL PROTECTED]:1]
 Goto(SIP/138100-08269548, 136101|1) in new
 stack
 [Aug 26 15:58:07] -- Goto (macro-dundi-priv,136101,1)
 [Aug 26 15:58:07]   == Auto fallthrough, channel
 

Re: [asterisk-users] is shared_lastcall available in 1.4

[Bob Pierce]

On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote:
 If you doubt about some part, you're welcome to ask, i'll try to help
 you, but i don't want to provide complete backport to you, as i won't
 be able to test it :)

Thanks Atis,

I'll probably try this in a few weeks when I start rebuilding the
permanent system that will replace our current temporary system.
That should give us the opportunity to test it on the bench instead of
playing around with the production box.

I'll probably be back to ask for help.

Have a great day,
Bob

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Re: [asterisk-users] Call transfer over IAX trunk

[Andrea Spadaccini]

Ciao Noah,

 What flags do you have in your Dial() statement?  If you want both
 parties to be able to transfer with the features.conf transfer, you
 need to have 'Tt' in your dial statement, like this:
 Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt)

Bingo. That was the problem.

Thanks a lot,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

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Re: [asterisk-users] sip show peers from shell or from CLI

[Olivier]

I think we're getting closer now as obviously Asterisk's greeting (...UNIX
connection) is mixed with its output.
(I can't understand why this happens now  as I never noticed this before and
didn't change anything).

I tried to use asterisk -rx '!sleep 1  sip show peers' to works around but
:
1. !sleep is not valid when issued from shell CLI (it's ok from Asterisk
CLI)
2. constructions like 'foo  foo' are not accepted by asterisk


Beside using sed to remove 'on^@', I can't imagine any smarter workaround
...

If anyone is inspired, please do not hesitate ...
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[asterisk-users] PRI Splitter

[Jeremy Mann]

Does anyone know of a pri splitter device?  Something that would take an 
incoming PRI, and based on DID send that out one of other multiple PRI ports?

I'm needing to take a single PRI from the telco, and send it to two separate 
phone systems(one asterisk) based on DID.

I know I could probably achieve the same thing with a 3 port PRI card in a 
server, but I'd like something braindead easy to configure from both a hardware 
and software perspective.



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Re: [asterisk-users] sip show peers from shell or from CLI

[Steven Howes]

On 27 Aug 2008, at 14:21, Olivier wrote:
 I think we're getting closer now as obviously Asterisk's greeting  
 (...UNIX connection) is mixed with its output.
 (I can't understand why this happens now  as I never noticed this  
 before and didn't change anything).

 I tried to use asterisk -rx '!sleep 1  sip show peers' to works  
 around but :
 1. !sleep is not valid when issued from shell CLI (it's ok from  
 Asterisk CLI)
 2. constructions like 'foo  foo' are not accepted by asterisk


 Beside using sed to remove 'on^@', I can't imagine any smarter  
 workaround ...

 If anyone is inspired, please do not hesitate ...

asterisk -rx'sip show peers' | grep -a '('

Bit hacky but works...


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Re: [asterisk-users] PRI Splitter

[Kevin P. Fleming]

Jeremy Mann wrote:

 I know I could probably achieve the same thing with a 3 port PRI card in
 a server, but I’d like something braindead easy to configure from both a
 hardware and software perspective.

Anything you use is going to (essentially) be a 3-port ISDN PRI capable
switch, because that is the only way to accomplish what you need. There
really isn't any way to 'split' a PRI, unlike a T1 using CAS signaling
which can be 'split' using a drop-and-insert multiplexer.

If you don't want to use a small PC with a 3-port T1 card in it, you can
use something like an Adtran Atlas to do the job.

Alternatively, just use a 2-port T1 card in the Asterisk server, and run
the PRI *through* the Asterisk server on the way to the other PBX.
That's the most common way to do what you want to do.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)


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[asterisk-users] Call Files

[Andy Dixon]

Hello again..

I am working on using call files to have a form of ringback - eg if an  
extension is busy, the caller can dial a number and when the callee is  
free, the call gets made.

I am trying to use a call file, which kind of works okay, however, if  
users have voicemail, it connects to that as opposed to waiting for  
the extension to become free.

Is there any known way around this..?

My call file looks like this:

Channel: Local/[EMAIL PROTECTED]
MaxRetries: 100
RetryTime: 1
WaitTime: 5
Extension: 666
Archive: yes
Callerid: Callback  666


Thanks!

Andy Dixon

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher 
[EMAIL PROTECTED] wrote:

 Given that this is the case, we may want to do one of the following:
 a) document that qualify=yes is incompatible with realtime, unless
 rtcachefriends is turned on, b) automatically disallow qualify=yes if the
 peer is realtime and caching is not turned on, or c) automatically cache
 realtime peers whose qualify field is set to yes.


What about an option d) If qualify=yes then Asterisk checks every 2 seconds,
if qualify=no then Asterisk does not check, and if qualify=numeric value
then Asterisk checks every numeric value milliseconds.  That would seem to
conform to how it behaves in sip.conf (reference:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf).

jm
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Re: [asterisk-users] sip show peers from shell or from CLI [SOLVED]

[Olivier]

It does work, here !!
Thanks you very much !!

2008/8/27 Steven Howes [EMAIL PROTECTED]


 On 27 Aug 2008, at 14:21, Olivier wrote:
  I think we're getting closer now as obviously Asterisk's greeting
  (...UNIX connection) is mixed with its output.
  (I can't understand why this happens now  as I never noticed this
  before and didn't change anything).
 
  I tried to use asterisk -rx '!sleep 1  sip show peers' to works
  around but :
  1. !sleep is not valid when issued from shell CLI (it's ok from
  Asterisk CLI)
  2. constructions like 'foo  foo' are not accepted by asterisk
 
 
  Beside using sed to remove 'on^@', I can't imagine any smarter
  workaround ...
 
  If anyone is inspired, please do not hesitate ...

 asterisk -rx'sip show peers' | grep -a '('

 Bit hacky but works...


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[asterisk-users] problem making outgoing calls

[bikrish]

Hi everybody

Here is my zapata.conf file and extension.conf file


zapata.conf

[channels]
;switchtype=national
;pridialplan=national
;signalling=pri_cpe
context=test
group=1
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
callprogress=no
callerid=asreceived
pickupgroup=1
pri_dialplan=unknown
immediate=no
signalling=pri_cpe
switchtype=national
channel = 1-15,17-31



extension.conf

[test]
exten = s,1,Wait(1)??? ; Wait a second, just for fun
exten = s,n,Dial(SIP/2000)
exten = 2000,1,Dial(SIP/2000)
exten = 2001,1,Dial(SIP/2001)
;exten = _X.,1,Dial(ZAP/g1/${EXTEN})
exten = _X.,1,Dial(ZAP/g1/${EXTEN},60)
exten = _X.,3,Hangup

I am from india. I am using centos 5 , the latest version of asterisk. With the 
above configuration i am able to make a call to my asterisk server and call to 
local mobile nos , but not able to call land line no.s and std no.s. Can anyone 
suggest where and what? i am missing. When i call to land line no or std no it 
gives me following log message

asterisk*CLI 
??? -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2002-097f63d0, 
ZAP/g1/32469868|60) in new stack
??? -- Requested transfer capability: 0x00 - SPEECH
??? -- Called g1/32469868
??? -- Channel 0/1, span 1 got hangup request, cause 3
??? -- Hungup 'Zap/1-1'
? == Everyone is busy/congested at this time (1:0/0/1)
? == Auto fallthrough, channel 'SIP/2002-097f63d0' status is 'CHANUNAVAIL'


Thanks 

bikrish
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Re: [asterisk-users] FreeTDS Versions?

[Tony Mountifield]

In article [EMAIL PROTECTED],
Norman Franke [EMAIL PROTECTED] wrote:
 Does any have some good experience with the various freetds variants?  
 Is 0.64 better or worse than 0.82? I know that to use 0.82 you have to  
 use ODBC, since libtds.a is not long installed. Which is more stable?  
 I plan on using it for CDR, realtime and func_odbc. I'm connecting to  
 SQL Server. I've had a few crashes with 0.82, I think, and I haven't  
 used 0.64.

I have eight systems that have been using FreeTDS 0.62 for CDR logging
for nearly four years with uptimes of over a year and no crashes.
I haven't tried later versions.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Wednesday 27 August 2008 08:56:02 J.M. wrote:
 On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher 

 [EMAIL PROTECTED] wrote:
  Given that this is the case, we may want to do one of the following:
  a) document that qualify=yes is incompatible with realtime, unless
  rtcachefriends is turned on, b) automatically disallow qualify=yes if the
  peer is realtime and caching is not turned on, or c) automatically cache
  realtime peers whose qualify field is set to yes.

 What about an option d) If qualify=yes then Asterisk checks every 2
 seconds, if qualify=no then Asterisk does not check, and if
 qualify=numeric value then Asterisk checks every numeric value
 milliseconds.  That would seem to conform to how it behaves in sip.conf
 (reference:
 http://www.voip-info.org/wiki-Asterisk+config+sip.conf).

Uh, no, you're misreading the documentation.  Qualify never checks every
2 seconds; the qualification time is the time in which a host must respond,
not the frequency with which Asterisk checks the host.  The frequency is
usually only once every 60 seconds.

-- 
Tilghman

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[asterisk-users] Callback voice Quality

[michel freiha]

Hi All,

I'm using A2billing application in order to make callback calls through my
asterisk server...Everything looks  fine except the voice quality...There is
a lot of cuts in the call with different codecs(G711, and G729)...Please
note that when making a call from any extensions to the same destination
number everything looks fine...
I used several carriers with the same result...What do you suggest?

Regards
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Re: [asterisk-users] PRI Splitter

[Jeremy Mann]

We've done the asterisk passthrough route, but if the asterisk box is down for 
whatever reason both systems are down.

Splitter wasn't the right word, but yes I see your point, I'll look into the 
Adtran.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Wednesday, August 27, 2008 8:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Splitter

Jeremy Mann wrote:

 I know I could probably achieve the same thing with a 3 port PRI card in
 a server, but I'd like something braindead easy to configure from both a
 hardware and software perspective.

Anything you use is going to (essentially) be a 3-port ISDN PRI capable
switch, because that is the only way to accomplish what you need. There
really isn't any way to 'split' a PRI, unlike a T1 using CAS signaling
which can be 'split' using a drop-and-insert multiplexer.

If you don't want to use a small PC with a 3-port T1 card in it, you can
use something like an Adtran Atlas to do the job.

Alternatively, just use a 2-port T1 card in the Asterisk server, and run
the PRI *through* the Asterisk server on the way to the other PBX.
That's the most common way to do what you want to do.

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk Experience (TM)


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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Wednesday 27 August 2008 09:29:55 Tilghman Lesher wrote:
 On Wednesday 27 August 2008 08:56:02 J.M. wrote:
  On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher 
 
  [EMAIL PROTECTED] wrote:
   Given that this is the case, we may want to do one of the following:
   a) document that qualify=yes is incompatible with realtime, unless
   rtcachefriends is turned on, b) automatically disallow qualify=yes if
   the peer is realtime and caching is not turned on, or c) automatically
   cache realtime peers whose qualify field is set to yes.
 
  What about an option d) If qualify=yes then Asterisk checks every 2
  seconds, if qualify=no then Asterisk does not check, and if
  qualify=numeric value then Asterisk checks every numeric value
  milliseconds.  That would seem to conform to how it behaves in sip.conf
  (reference:
  http://www.voip-info.org/wiki-Asterisk+config+sip.conf).

 Uh, no, you're misreading the documentation.  Qualify never checks every
 2 seconds; the qualification time is the time in which a host must respond,
 not the frequency with which Asterisk checks the host.  The frequency is
 usually only once every 60 seconds.

I have implemented option B, with a clear warning in the logs not to do this.
Testing is requested.

http://bugs.digium.com/view.php?id=13383

-- 
Tilghman

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[asterisk-users] VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass

[randulo]

Friday August 29, 2008 at 12 Noon EDT - VoIP Users Conference

Hi,

I was surprised to learn last week from Digium that asterisk 1.6 has
fax supported included. No more copying source files and editing
Makefile. While those tasks are not a big deal, they sort of defeat
the idea of using make. I greet the news with that fax is an integral
part of the asterisk install.  I wonder where the fax code comes from?
SpanDSP?

Digital-OPSiS is offering a Free Exhibitors Pass, valued at $695 to
one lucky winner and the drawing, postponed from last week will be
this Friday on Voip Users Conference.

 http://voipusersconference.org  -or-  http://bit.ly/voip

You can still  register to be part of the Digital-OPSiS team and get
all the action of Astricon.

  Go to http://bit.ly/astricon

For the rest of the time, we'll be talking about anything related to
telephony (and sometimes not related to anything) and especially VoIP,
SIP, hardware, asterisk, software.

Ironically, by definition being involved in VoIP means you have a
phone and usually have SIP providers or hardware and software to make
SIP calls. For years, the only vocal contact anyone had was the free
FWD network. Since March 2007, you can join us any Friday at 12 Noon
Eastern, 9AM Pacific or 1600 GMT to chat about anything that's on your
mind.

PSTN (724) 444-7444 and enter 22622# 1#
SIP [EMAIL PROTECTED] DTMF 22622# 1#

IRC.freenode.net #voip-users-conference

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[asterisk-users] Fax issue over cisco gateway

[Enrico Pasqualotto]

Hi all, I'm trying to send fax from Hylafax to a remote fax machine
through Asterisk and cisco 2801 as E1 gateway.
This is my architecture:

sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card 

For incoming fax I don't have any problem, but I'm not able to send fax
out of 2801.

My router conf: 

dial-peer voice 1 pots
 destination-pattern .T
 fax rate disable
 port 0/2/0:15
!
dial-peer voice 3 pots
 incoming called-number 53T
 fax rate disable
 direct-inward-dial
 forward-digits all
!
##

In asterisk console I see a lot of RTP packets lost:

  RTP-stats-003*CLI 
* Our Receiver:
  SSRC:  642188040
  Received packets: 17463
  Lost packets:  19686
  Jitter:0.
  Transit:   0.
  RR-count:  0
* Our Sender:
  SSRC:  1469234407
  Sent packets:  27926
  Lost packets:  0
  Jitter:0
  SR-count:  112
  RTT:   0.00

Anyone have idea of this problem? The packet lost quantity is normal?

Thanks Enrico.


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Re: [asterisk-users] problem making outgoing calls

[Tzafrir Cohen]

On Wed, Aug 27, 2008 at 10:16:34AM -0400, [EMAIL PROTECTED] wrote:
 Hi everybody
 
 Here is my zapata.conf file and extension.conf file
 
 
 zapata.conf
 
 [channels]
 ;switchtype=national
 ;pridialplan=national
 ;signalling=pri_cpe
 context=test
 group=1
 usecallerid=yes
 hidecallerid=no
 callwaiting=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 cancallforward=yes
 echocancel=yes
 rxgain=0.0
 txgain=0.0
 callprogress=no
 callerid=asreceived
 pickupgroup=1
 pri_dialplan=unknown
 immediate=no
 signalling=pri_cpe
 switchtype=national

If you have E1, chances are you need:

  switchtype = euroisdn

 channel = 1-15,17-31
 
 
 
 extension.conf
 
 [test]
 exten = s,1,Wait(1)??? ; Wait a second, just for fun
 exten = s,n,Dial(SIP/2000)
 exten = 2000,1,Dial(SIP/2000)
 exten = 2001,1,Dial(SIP/2001)
 ;exten = _X.,1,Dial(ZAP/g1/${EXTEN})
 exten = _X.,1,Dial(ZAP/g1/${EXTEN},60)
 exten = _X.,3,Hangup
 
 I am from india. I am using centos 5 , the latest version of asterisk. With 
 the above configuration i am able to make a call to my asterisk server and 
 call to local mobile nos , but not able to call land line no.s and std no.s. 
 Can anyone suggest where and what? i am missing. When i call to land line no 
 or std no it gives me following log message
 
 asterisk*CLI 
 ??? -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2002-097f63d0, 
 ZAP/g1/32469868|60) in new stack
 ??? -- Requested transfer capability: 0x00 - SPEECH
 ??? -- Called g1/32469868
 ??? -- Channel 0/1, span 1 got hangup request, cause 3
 ??? -- Hungup 'Zap/1-1'
 ? == Everyone is busy/congested at this time (1:0/0/1)
 ? == Auto fallthrough, channel 'SIP/2002-097f63d0' status is 'CHANUNAVAIL'
 
 
 Thanks 
 
 bikrish

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

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Re: [asterisk-users] VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass

[Tzafrir Cohen]

On Wed, Aug 27, 2008 at 07:46:10AM -0700, randulo wrote:
 Friday August 29, 2008 at 12 Noon EDT - VoIP Users Conference
 
 Hi,
 
 I was surprised to learn last week from Digium that asterisk 1.6 has
 fax supported included. No more copying source files and editing
 Makefile. While those tasks are not a big deal, they sort of defeat
 the idea of using make. I greet the news with that fax is an integral
 part of the asterisk install.  I wonder where the fax code comes from?
 SpanDSP?

Yes. Spandsp (0.0.5pre) is required for that.

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

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Re: [asterisk-users] Fax issue over cisco gateway

[Atis Lezdins]

On Wed, Aug 27, 2008 at 6:20 PM, Enrico Pasqualotto
[EMAIL PROTECTED] wrote:
 Hi all, I'm trying to send fax from Hylafax to a remote fax machine
 through Asterisk and cisco 2801 as E1 gateway.
 This is my architecture:

 sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card

 For incoming fax I don't have any problem, but I'm not able to send fax
 out of 2801.

 My router conf:
 
 dial-peer voice 1 pots
  destination-pattern .T
  fax rate disable
  port 0/2/0:15
 !
 dial-peer voice 3 pots
  incoming called-number 53T
  fax rate disable
  direct-inward-dial
  forward-digits all
 !
 ##

 In asterisk console I see a lot of RTP packets lost:

  RTP-stats-003*CLI
 * Our Receiver:
  SSRC:  642188040
  Received packets: 17463
  Lost packets:  19686
  Jitter:0.
  Transit:   0.
  RR-count:  0
 * Our Sender:
  SSRC:  1469234407
  Sent packets:  27926
  Lost packets:  0
  Jitter:0
  SR-count:  112
  RTT:   0.00

 Anyone have idea of this problem? The packet lost quantity is normal?


Hi Enrico,

In general SIP part is not good at all for fax transmission. However
if it's directly connected with no other packets flying by (and having
impact on bandwith/latency), you may have high degree of success by
using non-compressed codecs (for example G.711).

Could you provide SIP debug? I've seen that some switches
automatically listen for fax tones, and send T.38 handshake whenever
they detect fax on line. Looking into specs, says me that 2801
supports T.38, so perhaps it could be better idea (altough you would
have to use Asterisk 1.6 and app_txfax for sending faxes)

Also Hylafax log could say something.

Regards,
Atis


-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Atlanta Asterisk User's Group Conference Tonight Tuesday, August 26th at 7PM EDT

[Jay R. Ashworth]

On Tue, Aug 26, 2008 at 08:36:35PM -0400, SIP wrote:
 Jay R. Ashworth wrote:
  On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote:

  The shared desktop is available using a Java enabled browser at
  ???http://callin.xelatec.com/vnc??? with a password of ???aretta???.
 
  Of course you must first have Zoiper installed and then add a new Zoiper
  IAX account with Account name ???AtlaugConf???, Server Hostname
  ???pbx.aretta.net???, Username ???guest???, and no password or other
  information. Select Show Advanced Options for that account and uncheck
  ???Register on startup???. Apply the new account and click OK. Then from 
  the
  main user interface, select the new account, go off hook and dial
  ???2284???. That should connect you to the conference.
 
  Finally, for those really brave souls, you can also connect using the
  ITAD number of ???2284*455???. ITAD details are available at
  ???http://www.freenum.org/cookbook/???.
  
 
  Please don't post to mailing lists in non-7bit-ASCII unless you really
  have no other choice?
 
  That's how Mutt rendered your message here.
 
  I *think* those ???'s represent smart-quotes, but I really can't tell...
  and I'm probably not alone.
 
 Now, Jay... it's the global telecom world! Not everyone speaks ASCII-only.

I retract the tone; the headers clearly say UTF-8; it's my copy of Mutt's
fault; apparently.

 That's a little bit like standing in the United Nations and complaining 
 that not everyone speaks 'murican. ;)
 
 That said, you're correct. They're smart-quotes. I'm guessing it was 
 copy-pasted from a Word doc or some such.

Yup.

It is, however, rarely safe to assume that Any Given Mailing-list will
tolerate anything other than non-HTML ASCII-7, without advance notice...

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)

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[asterisk-users] Problem with Call Forward

[Dpto. Datos Television Costa Blanca]

Hi all.

This is my first post here and I searched a lot for a solution without luck.
Heres the problem; When I make a call forwarding from a extension to an 
external number (cell phone) it never work. Only work if the forward 
goes to another local extension. I dont know exactly what kind of 
information i should give so please, dont blame me much :P

Thanks in advance.

Dani B.

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[asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura

[Joseph]

Does anybody have an idea how to pass Off Hook caller ID to Asterisk via 
Linksys ?
I'm getting caller ID type I OK but when another customer rings the phone (when 
I'm on line)  the CID off hook is not coming through.
I think Off-Hook CID is called CID type II, isn't it?

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura

[RoLaNd RoLaNd]

i kinda have a relevant prob! 
my sipura 3102 wont pass CID to asterisk even though ive enabled such a feature 
in its web gui!



 Date: Wed, 27 Aug 2008 12:07:51 -0600
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via  
 Linsys/Sipura
 
 Does anybody have an idea how to pass Off Hook caller ID to Asterisk via 
 Linksys ?
 I'm getting caller ID type I OK but when another customer rings the phone 
 (when I'm on line)  the CID off hook is not coming through.
 I think Off-Hook CID is called CID type II, isn't it?
 
 -- 
 #Joseph
 GPG KeyID: ED0E1FB7
 
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Re: [asterisk-users] Problem with Call Forward

[Philipp Kempgen]

Dpto. Datos Television Costa Blanca schrieb:

 This is my first post here and I searched a lot for a solution without luck.
 Heres the problem; When I make a call forwarding from a extension to an 
 external number (cell phone) it never work. Only work if the forward 
 goes to another local extension. I dont know exactly what kind of 
 information i should give so please, dont blame me much :P

Post the dialplan (extensions.conf or extensions.ael, depending
on which one you use).


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass

[randulo]

On Wed, Aug 27, 2008 at 8:42 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
 part of the asterisk install.  I wonder where the fax code comes from?
 SpanDSP?

 Yes. Spandsp (0.0.5pre) is required for that.

Hi Tzfrir,

So this will be an option in selectmenu? (or menuselect or whatever
it's called, it's been a long time since I've built asterisk)

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[asterisk-users] OT Polycom URI and IP address dialing. Not.

[randulo]

Hi,

I've had the following problem with all Polycom phones. They will dial
a real SIP URI such as [EMAIL PROTECTED] but they will not
dial [EMAIL PROTECTED] which is the Talkshoe SIP server. Yet, any software
client I use and my Linksys SPA 941 will call both. The same is true
for the [EMAIL PROTECTED] of Talkshoe.

There would appear to be some kind of setting in the Polycom phone or
some mmethodology in the way the URI is called that differs from SIP
clients and the Linksys phone. Since only the Polycom phones sees
this distinction, what could it be? SRV records? What is the Talkshoe
address or server doing that onsip.com is not? Or vice versa? Any
suggestions from you Polycom geniuses out there?

tia,

randy

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Re: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura

[Joseph]

On 08/27/08 22:29, RoLaNd RoLaNd wrote:

i kinda have a relevant prob! 
my sipura 3102 wont pass CID to asterisk even though ive enabled such a 
feature in its web gui!


I can help you out with this, it easy :-)
Tell me what you have enabled?

In addition to the obvious one you need to set delay on PSTN line under:
FXO Timer Values (sec)
   PSTN Answer Delay: 3 (you might need higher number 4)

for me 3sec delay works

#Joseph
GPG KeyID: ED0E1FB7




 Date: Wed, 27 Aug 2008 12:07:51 -0600
 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via 
 Linsys/Sipura
 
 Does anybody have an idea how to pass Off Hook caller ID to Asterisk via 
 Linksys ?
 I'm getting caller ID type I OK but when another customer rings the phone 
 (when I'm on line)  the CID off hook is not coming through.
 I think Off-Hook CID is called CID type II, isn't it?
 
 -- 
 #Joseph
 GPG KeyID: ED0E1FB7

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[asterisk-users] Asterisk and Linksys One (PHB1100)

[George Pajari]

Has anyone tried using a Linksys One phone (such as the PHM1100) with 
Asterisk?

-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
  www.netvoice.ca  www.ip-centrex.ca  www.ip-pbx.ca  www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)


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Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

[Paul Hales]

One of the Asterisk people down here in Melb set it up for the company 
they used to work for, and I played with it once and it seemed to be usable.

PaulH


Lee, John (Sydney) wrote:
 Doesn't Queuemetrics run on a license basis?
 Anything else that's probably open source and free?

 

 Does anyone have any comments/experience about using asteriskguru queue
 statistics?
 http://www.asteriskguru.com/tutorials/installation_guide.html


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Re: [asterisk-users] PRI Splitter

[Paul Hales]

Asterisk?

PaulH


Jeremy Mann wrote:

 Does anyone know of a pri splitter device? Something that would take 
 an incoming PRI, and based on DID send that out one of other multiple 
 PRI ports?

 I’m needing to take a single PRI from the telco, and send it to two 
 separate phone systems(one asterisk) based on DID.

 I know I could probably achieve the same thing with a 3 port PRI card 
 in a server, but I’d like something braindead easy to configure from 
 both a hardware and software perspective.


 
 This e-mail, facsimile, or letter and any files or attachments 
 transmitted with it contains information that is confidential and 
 privileged. This information is intended only for the use of the 
 individual(s) and entity(ies) to whom it is addressed. If you are the 
 intended recipient, further disclosures are prohibited without proper 
 authorization. If you are not the intended recipient, any disclosure, 
 copying, printing, or use of this information is strictly prohibited 
 and possibly a violation of federal or state law and regulations. If 
 you have received this information in error, please notify Texas 
 Health Management Group immediately at 1-817-310-4999. Texas Health 
 Management Group, its subsidiaries, and affiliates hereby claim all 
 applicable privileges related to this information.
 

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[asterisk-users] Pri to sip interfaces

[Tom Moore]

Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?

Tom


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Re: [asterisk-users] PRI Splitter

[George Pajari]

Why a three-port PRI card?

Just put a two-port card into your Asterisk server, pull off those DIDs 
you want to process locally, and send the rest over the second port to 
the PBX. In the reverse direction, intercept calls from the PBX to the 
Asterisk DIDs but pass everything else to the telco.

We just finished installing just a system for a car dealership in BC 
that is splitting the body shop off into a separate building running off 
Asterisk while the rest of the company remains on their existing legacy 
PBX for a while longer (they'll come over later).

g.

Jeremy Mann wrote:

 Does anyone know of a pri splitter device? Something that would take 
 an incoming PRI, and based on DID send that out one of other multiple 
 PRI ports?

 I’m needing to take a single PRI from the telco, and send it to two 
 separate phone systems(one asterisk) based on DID.

 I know I could probably achieve the same thing with a 3 port PRI card 
 in a server, but I’d like something braindead easy to configure from 
 both a hardware and software perspective.
   

-- 
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
  www.netvoice.ca  www.ip-centrex.ca  www.ip-pbx.ca  www.vpas.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)


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Re: [asterisk-users] PRI Splitter

[Anciso, Roy]

Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then 
splitting it up so that 6 channels went to a video system (h.320) and 17 
channels to our PBX.  You can also convert the signaling or send out on 
different type of connections like v.35. Pretty cool device and rock solid. We 
never had any problems with it.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Wednesday, August 27, 2008 8:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Splitter


Asterisk?

PaulH


Jeremy Mann wrote:

 Does anyone know of a pri splitter device? Something that would take
 an incoming PRI, and based on DID send that out one of other multiple
 PRI ports?

 I'm needing to take a single PRI from the telco, and send it to two
 separate phone systems(one asterisk) based on DID.

 I know I could probably achieve the same thing with a 3 port PRI card
 in a server, but I'd like something braindead easy to configure from
 both a hardware and software perspective.


 
 This e-mail, facsimile, or letter and any files or attachments
 transmitted with it contains information that is confidential and
 privileged. This information is intended only for the use of the
 individual(s) and entity(ies) to whom it is addressed. If you are the
 intended recipient, further disclosures are prohibited without proper
 authorization. If you are not the intended recipient, any disclosure,
 copying, printing, or use of this information is strictly prohibited
 and possibly a violation of federal or state law and regulations. If
 you have received this information in error, please notify Texas
 Health Management Group immediately at 1-817-310-4999. Texas Health
 Management Group, its subsidiaries, and affiliates hereby claim all
 applicable privileges related to this information.
 

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Re: [asterisk-users] Pri to sip interfaces

[Darren Sessions]

Are you using an Asterisk PBX?


_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:


Hi guys,
What are your suggestions to people who have pbx systems that  
interface with
the world over pri and want to convert them to sip interfaces so  
that they

can use sip trunking?

Tom


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Re: [asterisk-users] Pri to sip interfaces

[Paul Hales]

Asterisk.

PaulH


Tom Moore wrote:
 Hi guys,
 What are your suggestions to people who have pbx systems that interface with
 the world over pri and want to convert them to sip interfaces so that they
 can use sip trunking?

 Tom


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Re: [asterisk-users] Pri to sip interfaces

[Tom Moore]

No, these are mainly Samsung pbx systems.
I know I can use Asterisk to do this but what be a solid platform to go with
that can go in the phone closet?
 
tom
 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Sessions
Sent: Wednesday, August 27, 2008 9:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pri to sip interfaces


Are you using an Asterisk PBX?



_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:


Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?

Tom


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Re: [asterisk-users] Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura

[Trevor Peirce]

Joseph wrote:
 Does anybody have an idea how to pass Off Hook caller ID to Asterisk via 
 Linksys ?
 I'm getting caller ID type I OK but when another customer rings the phone 
 (when I'm on line)  the CID off hook is not coming through.
 I think Off-Hook CID is called CID type II, isn't it?

   
I'm pretty confident the Linksys device does not support this 
functionality. Asterisk can't really do much with it anyway as it can't 
answer the call waiting call as long as the original call is still engaged.



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Re: [asterisk-users] Pri to sip interfaces

[Darren Sessions]

You can use an extremely simple Asterisk config to do the SIP-PRI  
call conversion that'd be very solid.



_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 27, 2008, at 7:37 PM, Tom Moore wrote:


No, these are mainly Samsung pbx systems.
I know I can use Asterisk to do this but what be a solid platform to  
go with that can go in the phone closet?


tom


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
] On Behalf Of Darren Sessions

Sent: Wednesday, August 27, 2008 9:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pri to sip interfaces

Are you using an Asterisk PBX?


_

Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_





On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:


Hi guys,
What are your suggestions to people who have pbx systems that  
interface with
the world over pri and want to convert them to sip interfaces so  
that they

can use sip trunking?

Tom


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Re: [asterisk-users] Pri to sip interfaces

[Paul Hales]

Are you looking for a hardware suggestion or a software suggestion?

PaulH


Tom Moore wrote:
 No, these are mainly Samsung pbx systems.
 I know I can use Asterisk to do this but what be a solid platform to 
 go with that can go in the phone closet?
  
 tom
  

 
 *From:* [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] *On Behalf Of *Darren 
 Sessions
 *Sent:* Wednesday, August 27, 2008 9:22 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Pri to sip interfaces

 Are you using an Asterisk PBX?


 _

 Darren Sessions
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 http://www.darrensessions.com
 _





 On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:

 Hi guys,
 What are your suggestions to people who have pbx systems that 
 interface with
 the world over pri and want to convert them to sip interfaces so that 
 they
 can use sip trunking?

 Tom


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Re: [asterisk-users] Pri to sip interfaces

[Francisco del rosario]

- Talk to a service provider that provide VoIP services.
- Does your PBX support SIP ?
- Does your PBX also provides Topology hiding and NAT traversal , otherwise, 
you may need a session border controller .
- Does your Service provider's softswitch has proven interworking tests with 
the brand of PBX you have ?
- Allow PRI to SIP trunking failover and vise versa

Good luck...By the way , where is your location ?


--- On Thu, 8/28/08, Tom Moore [EMAIL PROTECTED] wrote:

 From: Tom Moore [EMAIL PROTECTED]
 Subject: [asterisk-users] Pri to sip interfaces
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 asterisk-users@lists.digium.com
 Date: Thursday, August 28, 2008, 9:06 AM
 Hi guys,
 What are your suggestions to people who have pbx systems
 that interface with
 the world over pri and want to convert them to sip
 interfaces so that they
 can use sip trunking?
 
 Tom
 
 
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[asterisk-users] remove queue call

[Rilawich Ango]

Hi all,

  I have the following queue and members.  I found that there is a
call stuck in the queue so other call can't enter the queue.  I want
to know whether we can remove the call (by CLI) to free the queue.

ango

2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
holdtime), W:0, C:134, A:48, SL:88.8% within 120s
   Members:
  Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
   Callers:
  1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0)

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Re: [asterisk-users] OT Polycom URI and IP address dialing. Not.

[Michael Graves]

On Wed, 27 Aug 2008 14:05:05 -0700, randulo wrote:

Hi,

I've had the following problem with all Polycom phones. They will dial
a real SIP URI such as [EMAIL PROTECTED] but they will not
dial [EMAIL PROTECTED] which is the Talkshoe SIP server. Yet, any software
client I use and my Linksys SPA 941 will call both. The same is true
for the [EMAIL PROTECTED] of Talkshoe.

Junction (OnSIP) will not handle calls placed to [EMAIL PROTECTED]
no matter what the end point.

There would appear to be some kind of setting in the Polycom phone or
some mmethodology in the way the URI is called that differs from SIP
clients and the Linksys phone. Since only the Polycom phones sees
this distinction, what could it be? SRV records? What is the Talkshoe
address or server doing that onsip.com is not? Or vice versa? Any
suggestions from you Polycom geniuses out there?

I wonder if its a matter of DNS? I know that I can reach the Talkshoe
bridge by mapping the SIP URI to an OnSIP extension. Then in the
IP600/650 I just dial the extension. That's been wokring for me ever
since I found out about the [EMAIL PROTECTED] address.

Michael
--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245




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Re: [asterisk-users] remove queue call

[Lee, John (Sydney)]

 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
 holdtime), W:0, C:134, A:48, SL:88.8% within 120s
Members:
   Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
   Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
Callers:
   1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0)

Can you try ...
CLI module reload app_queue.so
CLI reload
CLI restart


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Re: [asterisk-users] X100P Card in OFFHOOK state

[Jay Ray]

This an FXO card, but for some reason it is configured as an FXS card on 
asteriskas per asterisk install Guide...

--- On Tue, 8/26/08, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
From: Eric ManxPower Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] X100P Card in OFFHOOK state
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 11:22 PM

It would be clearer if it said Hookstate (FXS ports only): Offhook

i.e. the state information is not valid for FXO ports.

Jay Ray wrote:
 Any pointers on this one?
 
 --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
 From: Jay Ray [EMAIL PROTECTED]
 Subject: [asterisk-users] X100P Card in OFFHOOK state
 To: asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 12:24 PM
 
 After I make a call o n the Zaptel Card X100P FXO moduleit remains
offhook state as shown here...
 
 Signalling Type: FXS Kewlstart
 Radio: 0re2uk*CLI
 Owner: None*CLI
 Real: Nonek*CLI
 Callwait: NoneI
 Threeway: NoneI
 Confno: -12uk*CLI
 Propagated Conference: -1
 Real in conference: 0
 DSP: noore2uk*CLI
 Relax DTMF: noCLI
 Dialing/CallwaitCAS: 0/0
 Default law: ulaw
 Fax Handled: noLI
 Pulse phone: noLI
 Echo Cancellation: 128 taps, currently OFF
 Actual Confinfo: Num/0, Mode/0x
 Actual Confmute: No
 Hookstate (FXS only): Offhook
 
 
 
 --
 Sometimes it still takes a new call while in this state and sometimes
rejects it...
 How to correct it such that after I hangup a call it goes back to onhook
  state...
 
 reloading wcfxo module using modprobe clears the issue
 
 
 
 
 
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-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.



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Re: [asterisk-users] X100P Card in OFFHOOK state

[Jay Ray]

Thx I will try that.

--- On Tue, 8/26/08, Guillermo Salas M. [EMAIL PROTECTED] wrote:
From: Guillermo Salas M. [EMAIL PROTECTED]
Subject: Re: [asterisk-users] X100P Card in OFFHOOK state
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 11:15 PM

El mar, 26-08-2008 a las 19:46 -0700, Jay Ray escribió:
 Any pointers on this one?
 
 --- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
 From: Jay Ray [EMAIL PROTECTED]
 Subject: [asterisk-users] X100P Card in OFFHOOK state
 To: asterisk-users@lists.digium.com
 Date: Tuesday, August 26, 2008, 12:24 PM
 
 After I make a call o n the Zaptel Card X100P FXO moduleit
 remains offhook state as shown here...
 
 Signalling Type: FXS Kewlstart
 Radio: 0re2uk*CLI
 Owner: None*CLI
 Real: Nonek*CLI
 Callwait: NoneI
 Threeway: NoneI
 Confno: -12uk*CLI
 Propagated Conference: -1
 Real in conference: 0
 DSP: noore2uk*CLI
 Relax DTMF: noCLI
 Dialing/CallwaitCAS: 0/0
 Default law: ulaw
 Fax Handled: noLI
 Pulse phone: noLI
 Echo Cancellation: 128 taps, currently OFF
 Actual Confinfo: Num/0, Mode/0x
 Actual Confmute: No
 Hookstate (FXS only): Offhook
 
 
 
 --
 Sometimes it still takes a new call while in this state and
 sometimes rejects it...
 How to correct it such that after I hangup a call it goes back
 to onhook state...
 
 reloading wcfxo module using modprobe clears the issue
 


Sounds like your card is not detecting the busy tone, try adding the
following line at your zapata.conf file:

busydetect=yes
busycount=6



Best regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net
SIP  : [EMAIL PROTECTED]
FWD  : 558563
USA  : 1 360 968 1701

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting
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Re: [asterisk-users] remove queue call

[Andy Kuo]

Hi,

Try   CLI soft hangup Local.

Andy

On 8/27/08, Rilawich Ango [EMAIL PROTECTED] wrote:
 Hi all,

  I have the following queue and members.  I found that there is a
 call stuck in the queue so other call can't enter the queue.  I want
 to know whether we can remove the call (by CLI) to free the queue.

 ango

 2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s
 holdtime), W:0, C:134, A:48, SL:88.8% within 120s
   Members:
  Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
  Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet
   Callers:
  1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0)

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[asterisk-users] can not load chan_dahdi.so from asterisk!

[lizhong zhu]

hello, all of users:
i have a problem with loading chan_dahdi.so. when i start asterisk, it always 
reports the can not open channel 1 in ...
here is my setting: in etc/system/dahdi.conf:
# Global data
fxsks=1
fxsks=2
fxoks=3
fxoks=4
loadzone= us
defaultzone = us
- in my chan_dahdi.conf---
group=0
signalling=fxs_ks
context=from-internal
channel = 1
channel = 2
signalling=fxo_ks
context=demo
channel = 3
channel = 4
--dmesg
dahdi: Registered Span 1 ('WCTDM/4') with 4 channels
dahdi: Span ('WCTDM/4') is new master
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXO (FCC mode)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXS/DPO
Module 3: Installed -- AUTO FXS/DPO
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
-
one error is that there is no transcode under /dev/dahdi, i do not why.
[EMAIL PROTECTED] dahdi]# ls
1  2  3  4  channel  ctl  pseudo  timer
[EMAIL PROTECTED] dahdi]#
and load from asterisk:
=
Connected to Asterisk SVN-trunk-r140246 currently running on new-host-13 (pid = 
2664)
Verbosity is at least 50
new-host-13*CLI module load chan_dahdi.so
Unable to load module chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
  == Parsing '/etc/asterisk/users.conf':   == Found
[Aug 28 11:25:57] WARNING[2797]: chan_dahdi.c:1139 dahdi_open: Unable to 
specify channel 1: No such device or address
[Aug 28 11:25:57] ERROR[2797]: chan_dahdi.c:8346 mkintf: Unable to open channel 
1: No such device or address
here = 0, tmp-channel = 1, channel = 1
[Aug 28 11:25:57] ERROR[2797]: chan_dahdi.c:13646 build_channels: Unable to 
register channel '1'
new-host-13*CLI
==
any one has idea for that problem? 
the second question is about modprobe the modules.  is the ztcfg command still 
workable  for DAHDI? it always opens zaptel.conf file.  what is the steps to 
load modules:
--- like this--
modprobe dahdi
modprobe wctdm
ztcfg -vvv; but my one open zaptel.conf file, i installed zaptel 
before. it should open dahdi.conf. 
i am still investgating DAHDI for further testing. thanks!
zhu

   
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