date:20060430

Are you sure it should be 4ess switchtype?  Have you tried national?  Is 
it only on 1800/toll free numbers?  Pridialplan=unknown, have you tried 
anything different for this value?


I can dial into my T3 just fine but I cannot dial out to toll free or 
911.  Any regular toll call out works, local or longdistance.


There are a couple strange things in your debug:
Called Number (len=10) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0) '18005551212' ]

-- Channel 0/1, span 1 received AOC-E charging 0 units

Thanks,
Steve

Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on asterisk.  Inbound 
calls are working just fine but I am not able to make outbound calls.  
Does anyone know  what I need to change to make outbound calls work?  
Right now the PRI is instantly hanging up on the outbound calls.

I have included full debug info as well as config files.


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

relevant portions of zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-pstn
pridialplan=unknown
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
;signaling=em_w
switchtype=4ess
group=0
channel = 1-23


debug info:
-- Executing Dial(SIP/202-2d92, zap/g0/18005551212) in new stack
-- Making new call for cr 32771
-- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=48
 Call Ref: len= 2 (reference 3/0x3) (Originator)
 Message type: SETUP (5)
 [04 02 80 90]
 Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
  Ext: 0  User information layer 1: 
Unknown (24)

 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

   Ext: 1  Channel: 1 ]
 [28 0f b1 74 65 73 74 20 65 78 74 65 6e 73 69 6f 6e]
 Display (len=15) Charset: 31 [ test extension ]
 [6c 05 21 83 32 30 32]
 Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation allowed of 
network provided number (3) '202' ]

 [70 08 80 33 38 31 36 30 36 38]
 Called Number (len=10) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0) '18005551212' ]

-- Called g0/18005551212
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 2 (reference 3/0x3) (Terminator)
 Message type: RELEASE COMPLETE (90)
 [08 03 83 e0 20]
 Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Transit network (3)
  Ext: 1  Cause: Unknown (96), class = Protocol Error 
(6) ]

  Cause data 1: 20 (32)
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
-- Channel 0/1, span 1 received AOC-E charging 0 units
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)




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Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk


Kristian Kielhofner wrote:

Steve Totaro wrote:
I have searched google and came up with too many options and packages 
that may or may not work for my needs, most articles seem to be for 
setting up routers.  Maybe someone on the list can give me some 
better insight.


I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box 
for all calls.  We have over one hundred agents and tons of 
recordings in wav format.  I also have a cron job that runs a script 
to mux the in and out files and ftp them to a NAS device and it runs 
every five minutes.
The NAS device and the * box are both directly connected to a Cisco 
Gigabit switch.  I have had complaints of calls fading in and out and 
also cutting off.  After reviewing the recordings, some of these 
complaints seem valid and I suspect the sheer bandwidth of the FTP 
traffic is causing the issues.  I also run nagios checks on the box 
and get ping warnings on a regular basis.
My question is, how can I throttle the FTP (Standard with dist) 
transfers using out of the box CentOS4.3 (or any easy to use, low 
learning curve package)?  I thought about FTPing the files at less 
frequent intervals but that just makes the issue less frequent but 
last longer.
I would like to accomplish throttling FTP on the Linux box with a 
solution that is not too elegant since this is a production machine 
in a busy call center.  If I cannot do it on the * box I guess my 
next step is to see if the Cisco Gigabit switch has any QoS 
functionality.

Thanks,
Steve
___


Steve,

If you don't want to get too fancy, you should switch to using 
rsync (if possible) and use the --bwlimit option.  If you MUST use 
ftp, try using trickle:


http://monkey.org/~marius/pages/?page=trickle

I haven't used it, but you should be able to call your FTP upload 
binary (whatever it may be) with it and force a lower transfer speed.


Let us know how it goes!


Trickle does not  seem to work with the IA64 procs :(
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Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk

2006-04-30 Thread Peter Bowyer


On 30/04/06, Steve Totaro [EMAIL PROTECTED] wrote:

My question is, how can I throttle the FTP (Standard with dist)
transfers using out of the box CentOS4.3 (or any easy to use, low
learning curve package)?  I thought about FTPing the files at less
frequent intervals but that just makes the issue less frequent but last
longer.


It's a while since I've looked at it, but I seem to recall that
ProFTPD has options for bandwidth limiting per login - you could take
a look at that.

I just took a glance at the online docs - 'TransferRate' - 'The
TransferRate directive is used to set transfer rates limits on the
transfer of data. This directive allows for transfer rates to be set
in a wide variety of contexts, on a per-command basis, and for certain
subsets of users. Note that this limit only applies to a single
connection, and not to the overall transfer rate of the server.'

www.proftpd.org

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
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[Asterisk-Users] How to monitor DTMF tones in a call?

2006-04-30 Thread Obelix


Is there a way to monitor the DTMF tones on a channel?

I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.

Is there way to monitor a sequence of DTMF tones and cancel the call?

If I use a SIP gateway or proxy rather than dial asterisk directly will that be
possible?

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Re: [Asterisk-Users] FW: NuFone Update: DIDs

2006-04-30 Thread Wilson Pickett


On 4/28/06, Matt [EMAIL PROTECTED] wrote:

Well services broke.  It's down.. DIDs ring fast busy.Does anyone
know the details of why nufone did not have backup providers?  How can
someone lose a contract with a CLEC like that?!  Is there more to this
story then we know?


Ok, NOW you can yell the sky is falling :)

I was in contact with two people from NuFone yesterday and I don't
have the *impression* they're going belly up, but obviously I could be
wrong. I don't know why they don't have a backup.

Unlike many providers, they have definitely posted info in the public
accessible area of their web site about this issue. The latest
statement says they expect the number to come back during the week.
One of the people I spoke with yesterday said within a week.

I'm just relaying what I've heard and read, please don't argue
directly with me, I have no stake in this except as a customer of
Nufone (and at least 10 other providers). I myself do have backup
providers ;)
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Re: [Asterisk-Users] stupid trick of the day (fried polycom)

2006-04-30 Thread Wilson Pickett


On 4/29/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:

[EMAIL PROTECTED] wrote:
 I've been playing around with a new system I'm going to install in
 another office.  In setting up the Polycom's, I accidently used a new
 power supply from a new 601 (24VDC) with an 600.  The 600 only require
 12VDC.  Now, I get nothing on the screen of the 600 when I plug in 12
 VDC.  (At the time, I didn't even realize the power supplies were
 supplying different voltages.)

Yes, this is one of my peeves with the 601... they changed power supply
voltages without changing connector styles, leading to this being a very
easy mistake to make.


At the least a HUGE RED sticker covering the socket warning about this
would a good idea. Incredible that this happened.
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[Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame

2006-04-30 Thread Hatami Nugraha


Hi all,

I always get this error message after I hangup a call, what does it mean ?

WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame

cheers,
hn.
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Re: [Asterisk-Users] stupid trick of the day (fried polycom)

2006-04-30 Thread Bruce Reeves

On a side note, I used a 601 adapter on a 501 and the unit failed to
power up. Once I realized the diffence in amps of each power supply I
swaped them and the 501 was fine. It would be nice if the adapters had
some distinction between them.On 4/30/06, Wilson Pickett [EMAIL PROTECTED] wrote:
On 4/29/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote:  I've been playing around with a new system I'm going to install in
  another office.In setting up the Polycom's, I accidently used a new  power supply from a new 601 (24VDC) with an 600.The 600 only require  12VDC.Now, I get nothing on the screen of the 600 when I plug in 12
  VDC.(At the time, I didn't even realize the power supplies were  supplying different voltages.) Yes, this is one of my peeves with the 601... they changed power supply voltages without changing connector styles, leading to this being a very
 easy mistake to make.At the least a HUGE RED sticker covering the socket warning about thiswould a good idea. Incredible that this happened.___--Bandwidth and Colocation provided by 
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-- BruceNortex Networks
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Re: [Asterisk-Users] (Semi-OT) QoS Question FTP Living with Asterisk

2006-04-30 Thread Assaf Flatto


Steve

From what i read here and from what others have suggested i can only 
surmise that you tried almost everything besides the simplest thing .


out of the box CentOs installs proftpd (AFAIK ) , this ftp engine has a 
simple too called mod_shaper

which allows as you can assume shape traffic rates .

example of the  configuration is :
IfModule mod_shaper.c
   ShaperEngine on
   ShaperLog /var/log/ftpd/shaper.log
   ShaperTable /var/log/ftpd/shaper.tab
   # Enableing FXP
   AllowForeignAddress on

   # An overall rate (in KB/s) must be set.  This line explicitly
   # sets both the download and upload rates to be the same.
***ShaperAll downrate 100 uprate 100

This is the line to manipulate

   # Allow all system users to see shaper info
   #ShaperControlsACLs info allow user *

   # Allow FTP admins to alter settings both overall and per-session
   ShaperControlsACLs all,sess allow group ftpadm


 /IfModule

you'll need to see if the module was installed in the initial 
installation , and if not - download it and compile it .



Assaf

Steve Totaro wrote:
I have searched google and came up with too many options and packages 
that may or may not work for my needs, most articles seem to be for 
setting up routers.  Maybe someone on the list can give me some better 
insight.


I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box 
for all calls.  We have over one hundred agents and tons of recordings 
in wav format.  I also have a cron job that runs a script to mux the 
in and out files and ftp them to a NAS device and it runs every five 
minutes.
The NAS device and the * box are both directly connected to a Cisco 
Gigabit switch.  I have had complaints of calls fading in and out and 
also cutting off.  After reviewing the recordings, some of these 
complaints seem valid and I suspect the sheer bandwidth of the FTP 
traffic is causing the issues.  I also run nagios checks on the box 
and get ping warnings on a regular basis.
My question is, how can I throttle the FTP (Standard with dist) 
transfers using out of the box CentOS4.3 (or any easy to use, low 
learning curve package)?  I thought about FTPing the files at less 
frequent intervals but that just makes the issue less frequent but 
last longer.
I would like to accomplish throttling FTP on the Linux box with a 
solution that is not too elegant since this is a production machine in 
a busy call center.  If I cannot do it on the * box I guess my next 
step is to see if the Cisco Gigabit switch has any QoS functionality.

Thanks,
Steve
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--
Assaf Flatto
Atelis IT Manager
Cellular: +972-54-5679230
e-mail: [EMAIL PROTECTED]

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[Asterisk-Users] some sip clients unreachable on sip-reload

2006-04-30 Thread Florian Meister

hi,

my asterisk is managing around 500 sip peers, and everytime I do a sip
reload many sip-peers get LAGGED and some get even UNREACHABLE. Any
suggestions ?

cu, florian
-- 

florian meister

EMAIL:   [EMAIL PROTECTED]
TELEPHONE:   +43 5572 501 134
FAX: +43 5572 501 97134
ADDRESS: gutenbergstrasse 1
 6858 schwarzach
 vorarlberg
 austria
WWW: www.medienhaus.at

 o If practice makes perfect, and nobody's perfect, why practice?
 o The solution of this problem is trival and is left as an exercise for
   the reader.
 o Recursive,adj.; see recursive.
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[Asterisk-Users] Fwd: can modify CHAN_SIP.c to generate a new exten= ext, 2, dial(tech/peer) ?

2006-04-30 Thread 陈帆

-- Forwarded message --From: 陈帆 [EMAIL PROTECTED]Date: Apr 30, 2006 10:30 AMSubject: can modify CHAN_SIP.c to generate a new exten= ext,2,dial(tech/peer) ?
To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

Hi, ALL,

Whether can modify CHAN_SIP.cto add one new int ast_add_extension
(const char *context, int replace, const char *extension, int priority, const char *label, const
 char *callerid,const char *application, void *data, void (*datad)(void *), const char *registrar) to generate a new exten= ext,2,dial(tech/peer)

01610 void register_peer_exten(struct sip_peer *peer,intonoff)01613 {01614char multi[256];01615char *stringp, *ext;01616if (!ast_strlen_zero(regcontext)) {01617 ast_copy_string(multi, ast_strlen_zero(peer-regexten) ? peer-name : peer-regexten, sizeof(multi)); 
01618 stringp = multi;01619 while((ext = strsep(stringp, ))) {01620if (onoff)ra01621 ast_add_extension(regcontext, 1, ext, 1, NULL, NULL, Noop, strdup(peer-name), free, channeltype); 


ast_add_extension(regcontext, 1, ext, 2, NULL, NULL, Dial, strdup(peer-name), free, channeltype); /*how to set the *data,,oomadd the exten = ext, 2, dial(channeltype/peername) */ 


01622else01623 ast_context_remove_extension(regcontext, ext, 1, NULL);01624 }01625}01626 }-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 
-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150 
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Re: [Asterisk-Users] Error : ast_readaudio_callback: Failed to write frame

2006-04-30 Thread Imran Ahmed


On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote:

Hi all,

I always get this error message after I hangup a call, what does it mean ?

WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame



This means you hungup while asterisk was trying to play a file to you.
It should be of no concern as long as it does not happen during a call.
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RE: [Asterisk-Users] Compare to Skype

2006-04-30 Thread mgraves

  Original Message 
 
 Skype uses iLBC codec, which has great jitter compensation.  IIRC, the
 newer SIP channels of * are supposed to have the same capabilities, but
 I have not tested.  I really do not like Skype (prefer FWD), but I must
 say, over satellite, etc, they provide quality..  All about the codec in
 this case..


Errr...no...this is wrong. 

Skype uses ISAC from Global IP Sound. iLBC is something different see
http://www.globalipsound.com/solutions/solutions_Codecs.php

One of the reasons Skype sounds good is that its a closed system and so
can leverage a wideband codec. Instead of the normal 8khz sample rate
it uses 16khz. That makes for clearer sound. Since ISAC is a
proprietary relative of iLBC its jitter compensation is also very good.

My understanding is that Asterisk cannot presently use any wideband
codecs as it is hard coded to the 8khz sample rate at its core.
Adapting Asterisk to wideband capability has been discussed but will be
a huge amount of work. Further, only if you know that the calls will
stay wideband end-to-end will the benefits of wideband be apparent.
That means no PSTN segments.

Michael Graves
[EMAIL PROTECTED]


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Re: [Asterisk-Users] Compare to Skype

2006-04-30 Thread Ronald Wiplinger


[EMAIL PROTECTED] wrote:

 Original Message 

Skype uses iLBC codec, which has great jitter compensation.  IIRC, the
newer SIP channels of * are supposed to have the same capabilities, but
I have not tested.  I really do not like Skype (prefer FWD), but I must
say, over satellite, etc, they provide quality..  All about the codec in
this case..




Errr...no...this is wrong. 


Skype uses ISAC from Global IP Sound. iLBC is something different see
http://www.globalipsound.com/solutions/solutions_Codecs.php

One of the reasons Skype sounds good is that its a closed system and so
can leverage a wideband codec. Instead of the normal 8khz sample rate
it uses 16khz. That makes for clearer sound. Since ISAC is a
proprietary relative of iLBC its jitter compensation is also very good.

My understanding is that Asterisk cannot presently use any wideband
codecs as it is hard coded to the 8khz sample rate at its core.
Adapting Asterisk to wideband capability has been discussed but will be
a huge amount of work. Further, only if you know that the calls will
stay wideband end-to-end will the benefits of wideband be apparent.
That means no PSTN segments.

Michael Graves
[EMAIL PROTECTED]

  


Sadly to say, but users do not care about the why, they only care about 
the quality! and they simple ask to fix it!


I hope there is soon a solution, otherwise, we have to skip all our 
effort and just use skype!
And I would hate to see that. I just lost 20 US$ to Ebay - the newly 
parent company of skype, for a not received parcel, but the rules says, 
below 25 US$ there is no guarantee that you get anything



bye

Ronald Wiplinger

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RE: [Asterisk-Users] Compare to Skype

2006-04-30 Thread mgraves

What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required in an
implementation that processed the streams to bridge into the TDM/PSTN
world. It would be greatbut don't hold your breath.

For now there are Skype bridges like PSWG and Uplink that interface
Skype to SIP. These are simplistic but sometimes workable.

Does anyone here have experience with Uplink? I tried PSGW and gave up
eventually.

Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262



  Original Message 
 Subject: Re: [Asterisk-Users] Compare to Skype
 From: Ronald Wiplinger [EMAIL PROTECTED]
 Date: Sun, April 30, 2006 9:09 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 [EMAIL PROTECTED] wrote:
   Original Message 
 
  Skype uses iLBC codec, which has great jitter compensation.  IIRC, the
  newer SIP channels of * are supposed to have the same capabilities, but
  I have not tested.  I really do not like Skype (prefer FWD), but I must
  say, over satellite, etc, they provide quality..  All about the codec in
  this case..
  
 
 
  Errr...no...this is wrong. 
 
  Skype uses ISAC from Global IP Sound. iLBC is something different see
  http://www.globalipsound.com/solutions/solutions_Codecs.php
 
  One of the reasons Skype sounds good is that its a closed system and so
  can leverage a wideband codec. Instead of the normal 8khz sample rate
  it uses 16khz. That makes for clearer sound. Since ISAC is a
  proprietary relative of iLBC its jitter compensation is also very good.
 
  My understanding is that Asterisk cannot presently use any wideband
  codecs as it is hard coded to the 8khz sample rate at its core.
  Adapting Asterisk to wideband capability has been discussed but will be
  a huge amount of work. Further, only if you know that the calls will
  stay wideband end-to-end will the benefits of wideband be apparent.
  That means no PSTN segments.
 
  Michael Graves
  [EMAIL PROTECTED]
 

 
 Sadly to say, but users do not care about the why, they only care about 
 the quality! and they simple ask to fix it!
 
 I hope there is soon a solution, otherwise, we have to skip all our 
 effort and just use skype!
 And I would hate to see that. I just lost 20 US$ to Ebay - the newly 
 parent company of skype, for a not received parcel, but the rules says, 
 below 25 US$ there is no guarantee that you get anything
 
 
 bye
 
 Ronald Wiplinger
 
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Re: [Asterisk-Users] Compare to Skype

2006-04-30 Thread Ronald Wiplinger


[EMAIL PROTECTED] wrote:

What would be ideal is the introduction of an open source wideband codec
implementation. Then you could see it adopted into SIP end points and
used with SER realtively quickly. Sadly, an Asterisk implmentation
would lag a little behind due to the amount of work required in an
implementation that processed the streams to bridge into the TDM/PSTN
world. It would be greatbut don't hold your breath.

For now there are Skype bridges like PSWG and Uplink that interface
Skype to SIP. These are simplistic but sometimes workable.

Does anyone here have experience with Uplink? I tried PSGW and gave up
eventually.
  
I am also a supporter of PSGW although on my AMD it never worked. Now 
it is getting obsolete at all, since I switch next week finally to a 
Linux desktop 


I never heard about Uplink, where is it, does it work?
From the uplink web:


   System Requirements

   * Windows 98/2000/Me/XP/2003

sigh 


bye

Ronald Wiplinger
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[Asterisk-Users] Intermittent problem dialling out on a SIP channel

2006-04-30 Thread hugolivude


Hi,

Red Hat 9.0
Asterisk 1.2.7.1

I'm having a bit of an intermittent problem with my SIP account. 
Often (but not always) when I start * or RELOAD my dial plan from the

CLI I get this message:


Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822

add_realm_authentication: Format for authentication entry is
user[:[EMAIL PROTECTED] at line 31

Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable

to lookup ''

Line 31 of my sip.conf is auth=md5
.  Whenever I see that message, I am unable to dial out on the SIP channel:


-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]||t|) in new stack
Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such

host: 6477235412

Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)


If I repeatedly RELOAD enough times from the CLI though, eventually
one will work without the error messages and I can dial out.

I tried commenting out auth=md5 in my SIP.conf.  That seemed to
eliminate the add_realm_authentication error, but I still see the
ast_get_ip_or_srv from time to time, and when I do, I can't dial
out.

Also, while I am successful at dialling out from time-to-time,
depending upon how the RELOAD goes, I havn't yet been able to receive
a SIP call.

Finally, another thing that troubles me is that sometimes I can use
QUIT or EXIT to exit the CLI, but other times it just doesn't work as
shown below:


Use EXIT or QUIT to exit the asterisk console
Reloading MGCP
 == Parsing '/etc/asterisk/mgcp.conf': Found
 == MGCP Listening on 0.0.0.0:2727
 == Using TOS bits 0

Use EXIT or QUIT to exit the asterisk console
 == Parsing '/etc/asterisk/sip_notify.conf': Found
*CLIquit
No such command 'quit' (type 'help' for help)
*CLI QUIT
No such command 'QUIT' (type 'help' for help)
*CLI EXIT
No such command 'EXIT' (type 'help' for help)


Any ideas?  My sip.conf is provided below:

[general]
;
context=incoming-bogus-calls
port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0  ; Address to bind to (all addresses on machine)
maxexpirey=3600   ; Must be larger than the re-register timeout 
on the router
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw
;
; This section is because i'm behind nat
;

register=6477235412:mypassword@sip.unlimitel.ca/6477235412
externip=mystaticIPaddress ;Outside address
localnet=192.168.0.148/255.255.255.0 ;Inside Network
;
;
[6477235412]
type=peer
auth=md5
username=6477235412
fromuser=6477235412
fromdomain=unlimitel.ca
secret=mypassword
host=sip.unlimitel.ca
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=g729
dtmfmode=rfc2833
insecure=very
context=incoming
;
;-
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Re: [Asterisk-Users] Compare to Skype

2006-04-30 Thread Eric \ManxPower\ Wieling






One of my user is praising Skype!!!

I cannot figure out anymore what I can improve!

This users sip show peers  is jumping from 65 msec to 1800 all the 
time. Of course his voice quality is like a morse code with dashes or 
dots of connection time.
The next minute he calls me via Skype and it works fine  What 
indicates that there is no fault on his Internet connection!!!


He is using his notebook and Xlite, but also tried the snom 360.

Any hints?


There are 2 issues here.

1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to 
transport audio for SIP (and other protocols).  This means that ANY 
jitter on the SIP Phone - Asterisk link will cause audio problems.


2) Asterisk times it's outgoing audio based on the incoming audio. 
Therefore, if there is jitter on the SIP Phone - Asterisk link then 
Asterisk will replicate that jitter on the Asterisk - SIP Phone direction.


REMEMBER, a jitter buffer only applies on INCOMING audio (from the 
standpoint of the device).


These two issues are the main reason I have not deployed remote SIP 
phones for my clients.


I believe that BOTH of these issues will be fixed in Asterisk 1.4.x, 
which should be released sometime this summer.


--
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Huntsville, Chattanooga, and Montgomery.

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Re: [Asterisk-Users] problame with outbound calls on pri

2006-04-30 Thread Eric \ManxPower\ Wieling


Have you tried switchtype=national ?

Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on asterisk.  Inbound 
calls are working just fine but I am not able to make outbound calls.  
Does anyone know  what I need to change to make outbound calls work?  
Right now the PRI is instantly hanging up on the outbound calls.

I have included full debug info as well as config files.


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

relevant portions of zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-pstn
pridialplan=unknown
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
;signaling=em_w
switchtype=4ess
group=0
channel = 1-23


debug info:
-- Executing Dial(SIP/202-2d92, zap/g0/18005551212) in new stack
-- Making new call for cr 32771
-- Requested transfer capability: 0x00 - SPEECH
  Protocol Discriminator: Q.931 (8)  len=48
  Call Ref: len= 2 (reference 3/0x3) (Originator)
  Message type: SETUP (5)
  [04 02 80 90]
  Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
   Ext: 0  User information layer 1: 
Unknown (24)

  [18 03 a9 83 81]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

Ext: 1  Channel: 1 ]
  [28 0f b1 74 65 73 74 20 65 78 74 65 6e 73 69 6f 6e]
  Display (len=15) Charset: 31 [ test extension ]
  [6c 05 21 83 32 30 32]
  Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation allowed of 
network provided number (3) '202' ]

  [70 08 80 33 38 31 36 30 36 38]
  Called Number (len=10) [ Ext: 1  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0) '18005551212' ]

-- Called g0/18005551212
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 2 (reference 3/0x3) (Terminator)
 Message type: RELEASE COMPLETE (90)
 [08 03 83 e0 20]
 Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Transit network (3)
  Ext: 1  Cause: Unknown (96), class = Protocol Error 
(6) ]

  Cause data 1: 20 (32)
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
-- Channel 0/1, span 1 received AOC-E charging 0 units
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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[Asterisk-Users] newbie-too much latency

2006-04-30 Thread Ryder Brook

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS.  The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log :    Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command'  Apr 30 10:28:28 VERBOSE[3051] logger.c: -- Starting simple switch on 'Zap/4-1'  Apr 30 10:28:32 NOTICE[3051] chan_zap.c: Got event 18 (Ring Begin)...  Apr 30 10:28:34 NOTICE[3051] chan_zap.c: Got event 2 (Ring/Answered)...  Apr 30 10:28:34 VERBOSE[3051] logger.c: -- Executing Set("Zap/4-1", "FROM_DID=s") in new stack  Apr 30 10:28:34 VERBOSE[3051] logger.c: -- Executing Set("Zap/4-1", "FAX_RX=disabled") in new stack  Apr 30 10:28:34 VERBOSE[3051] logger.c: -- Executing Answer("Zap/4-1", "") in new stack  Apr 30 10:28:34 DEBUG[3051] chan_zap.c: Took
 Zap/4-1 off hook  Apr 30 10:28:34 DEBUG[3051] chan_zap.c: Enabled echo cancellation on channel 4  Apr 30 10:28:34 DEBUG[3051] chan_zap.c: Engaged echo training on channel 4  Apr 30 10:28:34 VERBOSE[3051] logger.c: -- Executing PlayTones("Zap/4-1", "ring") in new stack  Apr 30 10:28:34 DEBUG[3051] channel.c: Scheduling timer at 160 sample intervals  Apr 30 10:28:34 VERBOSE[3051] logger.c: -- Executing NVFaxDetect("Zap/4-1", "20") in new stack  Apr 30 10:28:34 DEBUG[3051] app_nv_faxdetect.c: Preparing detect of fax (waitdur=20ms, sildur=1000ms, mindur=100ms, maxdur=-1ms)  Apr 30 10:28:34 DEBUG[3051] channel.c: Generator got voice, switching to phase locked mode  Apr 30 10:28:34 DEBUG[3051] channel.c: Scheduling timer at 0 sample intervals  Apr 30 10:28:34 DEBUG[3051] app_nv_faxdetect.c: Start of voice token!  Apr 30 10:28:34 DEBUG[3051] app_nv_faxdetect.c: Found unqualified token of 0 ms  Apr 30 10:28:50 DEBUG[3050]
 manager.c: Manager received command 'Command'  Apr 30 10:28:50 DEBUG[3050] manager.c: Manager received command 'Command'  Apr 30 10:28:55 VERBOSE[3051] logger.c: -- Executing Goto("Zap/4-1", "ivr-4|s|1") in new stack  Apr 30 10:28:55 VERBOSE[3051] logger.c: -- Goto (ivr-4,s,1)  Apr 30 10:28:55 VERBOSE[3051] logger.c: -- Executing Set("Zap/4-1", "LOOPCOUNT=0") in new stack  Apr 30 10:28:55 VERBOSE[3051] logger.c: -- Executing Answer("Zap/4-1", "") in new stack  Apr 30 10:28:55 VERBOSE[3051] logger.c: -- Executing Wait("Zap/4-1", "1") in new stack  Apr 30 10:28:56 VERBOSE[3051] logger.c: -- Executing Set("Zap/4-1", "TIMEOUT(digit)=10") in new stack  Apr 30 10:28:56 VERBOSE[3051] logger.c: -- Digit timeout set to 10  Apr 30 10:28:56 VERBOSE[3051] logger.c: -- Executing Set("Zap/4-1", "TIMEOUT(response)=10") in new stack  Apr 30 10:28:56 VERBOSE[3051] logger.c: -- Response timeout set to 10  Apr 30
 10:28:56 VERBOSE[3051] logger.c: -- Executing BackGround("Zap/4-1", "custom/welcomeNineToFive") in new stack == Hope someone can help me. - balu  __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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[Asterisk-Users] Legacy PBX integration


Hi list!

I managed to come reasonably far (farther than I thought I would) but have 
two problems.


I still need to pass calls to the Legacy PBX for Fax (I need it as a 
channel bank).


I have calls coming in into asterisk, that works fine. Based on the DID I 
can route calls to the Legacy PBX but I'm puzzled how.


I guess I need a new dial command for that? All fax calls are now coming 
in a new context which I called topbx. If I issue a dial command there the 
legacy PBX treats it as a local extension call and not a call from the 
outside.


Which dial command do I need to use to make the old PBX believe the call 
came from outside?


(All the pages I found on this subject mention something about retaining 
caller ID which is nice but now I need to retain DID info on the call I 
guess?)


Thanks for any help!


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Re: [Asterisk-Users] problame with outbound calls on pri

2006-04-30 Thread Doug Langley

I started with national but then changed it once we looked and the 
other pbx was set for 4ess.  I'll put it back and look at the debug info again.


At 09:07 AM 4/30/2006, you wrote:

Have you tried switchtype=national ?

Doug Langley wrote:
Hi. recently I have been trying to setup a PRI on 
asterisk.  Inbound calls are working just fine but I am not able to 
make outbound calls.

Does anyone know  what I need to change to make outbound calls work?
Right now the PRI is instantly hanging up on the outbound calls.
I have included full debug info as well as config files.

/etc/zaptel.conf
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
relevant portions of zapata.conf:
[trunkgroups]
[channels]
language=en
context=from-pstn
pridialplan=unknown
signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
;signaling=em_w
switchtype=4ess
group=0
channel = 1-23

debug info:
-- Executing Dial(SIP/202-2d92, zap/g0/18005551212) in new stack
-- Making new call for cr 32771
-- Requested transfer capability: 0x00 - SPEECH
  Protocol Discriminator: Q.931 (8)  len=48
  Call Ref: len= 2 (reference 3/0x3) (Originator)
  Message type: SETUP (5)
  [04 02 80 90]
  Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info 
transfer capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)
   Ext: 0  User information layer 1: 
Unknown (24)

  [18 03 a9 83 81]
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, 
Exclusive Dchan: 0

 ChanSel: Reserved
Ext: 1  Coding: 0   Number 
Specified   Channel Type: 3

Ext: 1  Channel: 1 ]
  [28 0f b1 74 65 73 74 20 65 78 74 65 6e 73 69 6f 6e]
  Display (len=15) Charset: 31 [ test extension ]
  [6c 05 21 83 32 30 32]
  Calling Number (len= 7) [ Ext: 0  TON: National Number 
(2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation allowed of 
network provided number (3) '202' ]

  [70 08 80 33 38 31 36 30 36 38]
  Called Number (len=10) [ Ext: 1  TON: Unknown Number Type 
(0)  NPI: Unknown Number Plan (0) '18005551212' ]

-- Called g0/18005551212
 Protocol Discriminator: Q.931 (8)  len=10
 Call Ref: len= 2 (reference 3/0x3) (Terminator)
 Message type: RELEASE COMPLETE (90)
 [08 03 83 e0 20]
 Cause (len= 5) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: Transit network (3)
  Ext: 1  Cause: Unknown (96), class = Protocol Error (6) ]
  Cause data 1: 20 (32)
-- Processing IE 8 (cs0, Cause)
-- Channel 0/1, span 1 got hangup
-- Channel 0/1, span 1 received AOC-E charging 0 units
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time (1:0/0/1)

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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[Asterisk-Users] Change in audio file while listening to it

2006-04-30 Thread Marco Trucchi


Hello everybody,
does anybody know how to handle the following problem?

I update some gsm audio files every 10 minutes, by rewriting directly on them.

I've noticed that if the file is being played by asterisk exactly in 
the moment when I rewrite onto it, who is calling hears a small 
jump and then it is the updated file that starts being played, 
starting from about the same position of the old one.
(i.e. if the update arrives after 10 seconds of playing on the old 
file, the updated file starts after the jump about at its 10 second position).
I would prefer to end up with the old file without changing the 
current conversation.


Maybe a parameter handle this?
Otherwise the only way that I see is to give a different name to each 
file, then let asterisk read the most up-to-date. But it would be not 
as easy as it is now.


For information, I use Asterisk 1.2.6 on Linux CentOS. I pay a SIP 
Gateway to have a geographical number that points to my asterisk 
(sorry if I do not use the correct terms).



Thanks a lot!
Cheers,
Marco

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[Asterisk-Users] Asterisk is stripping my area code

I've installed [EMAIL PROTECTED] and gotten inbound calls going to an extension, extension to extension calling works but I'm still missing a few pieces. The most annoying one is that apparently asterisk is stripping the area code from the number I'm dialing but I can't figure out how to stop it. I have in my outbound route under Dialing rules:
1NXXNXXNXXNXXWe are required to dial all 10 numbers since there are 3 area codes in Atlanta now.Using freePBX admin. I think [EMAIL PROTECTED] is version 2.7. One less than the most recent, in any case.
Any suggestions would be helpful.Thanks,Jim.
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[Asterisk-Users] RE: Asterisk is stripping my area code

I don't know if this helps, from the log.Jim.Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Executing Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new stack 
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Deferring dialing... Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 1/97707190239
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, channel 1 Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) 

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Re: [Asterisk-Users] Codec G729 no longer works.

2006-04-30 Thread Patrick

On Sat, 2006-04-29 at 22:49 -0400, Jason A. Kates wrote:
 I upgraded my server from Fedora Core 4 to Fedora Core 5.
 
 I was wondering if anybody else has run into the problem and know's the
 fix?
 
 I recompiled asterisk and if I don't have
 the /usr/lib/asterisk/modules/codec_g729a.so
 file in place it works.
 
 I use or used to use the licensed G729 Codec from Digium.
 
 This is the error message from asterisk -vvg:
  [app_playback.so] = (Sound File Playback Application)
   == Registered application 'Playback'
  [app_dumpchan.so] = (Dump Info About The Calling Channel)
   == Registered application 'DumpChan'
  [app_zapateller.so] = (Block Telemarketers with Special Information
 Tone)
   == Registered application 'Zapateller'
  [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator)
   == Registered translator 'ilbctolin' from format ilbc to slin, cost 7
   == Registered translator 'lintoilbc' from format slin to ilbc, cost
 245
  [codec_g729a.so]Apr 29 22:25:25 WARNING[16253]: loader.c:325
 __load_resource: /usr/lib/asterisk/modules/codec_g729a.so: cannot
 restore segment prot after reloc: Permission denied

Looks like an SELinux issue. Try booting with selinux=0 or disable
SELinux in /etc/sysconfig/selinux, reboot and see if it works then.

Regards,
Patrick

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RE: [Asterisk-Users] RE: Asterisk is stripping my area code




Are you dialing 9 first? It is showing that the digits you 
dialed are:

9-770-719-0239
Using your dialplan you should be dialing 
1-770-719-0239



Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:10 AMTo: 
  Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE: 
  Asterisk is stripping my area code
  I don't know if this helps, from the log.Jim.Apr 30 
  12:58:55 VERBOSE[4225] logger.c: -- Executing 
  Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new 
  stack 
  Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
  12:58:55 DEBUG[4225] chan_zap.c: Deferring 
  dialing... 
  Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 
  1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, 
  channel 
  1 
  Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
  Complete(12) on channel 1 (index 
  0) 
  
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Re: [Asterisk-Users] Codec G729 no longer works.

2006-04-30 Thread Mathieu Chouquet-Stringer

[EMAIL PROTECTED] (Patrick) writes:
 Looks like an SELinux issue. Try booting with selinux=0 or disable
 SELinux in /etc/sysconfig/selinux, reboot and see if it works then.

If you to double check it is a SELinux issue, no need to reboot:
'setenforce permissive' will (temporarily) do the trick (man setenforce for
more information)
-- 
Mathieu Chouquet-Stringer
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Re: [Asterisk-Users] RE: Asterisk is stripping my area code

No, I'm just dialing 7707190239. When I tried it with a 1, it gave me the same result, a nice lady telling me when making a local call you must first dial the areacode or words to that effect. >From the log, after using the 1:
Apr 30 13:29:04 DEBUG[4242] pbx.c: Function result is ' 7707190069'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing NoOp(SIP/200-fa0b, CallerID set to  7707190069) in new stack
Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, GROUP()=OUT_1) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is '1'Apr 30 13:29:04 DEBUG[4242] 
pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?108) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] 
logger.c: -- Executing Set(SIP/200-fa0b, DIAL_NUMBER=17707190239) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, DIAL_TRUNK=1) in new stack
Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing AGI(SIP/200-fa0b, fixlocalprefix) in new stackApr 30 13:29:04 VERBOSE[4242] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
Apr 30 13:29:04 VERBOSE[4242] logger.c: fixlocalprefix: Removed prefix. New number: 7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 VERBOSE[4242] 
logger.c: -- Executing Set(SIP/200-fa0b, OUTNUM=97707190239) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Set(SIP/200-fa0b, custom=ZAP/1) in new stack
Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?16) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c
: Not taking any branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- Executing Dial(SIP/200-fa0b, ZAP/1/97707190239|120|W) in new stackApr 30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239'
Apr 30 13:29:04 DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] logger.c: -- Called 1/97707190239Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0)
Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 1So what should I have in my dialing plan to let me dial 7707190239 and have it use that exact number? Or do I have to dial 9 first?Thanks,
Jim.On 4/30/06, Kerry Garrison [EMAIL PROTECTED] wrote:





Are you dialing 9 first? It is showing that the digits you 
dialed are:

9-770-719-0239
Using your dialplan you should be dialing 
1-770-719-0239



Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- 
[EMAIL PROTECTED]

http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:10 AMTo: 
  Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] RE: 
  Asterisk is stripping my area code
  I don't know if this helps, from the log.Jim.Apr 30 
  12:58:55 VERBOSE[4225] logger.c: -- Executing 
  Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new 
  stack 
  Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
  12:58:55 DEBUG[4225] chan_zap.c: Deferring 
  dialing... 
  Apr 30 12:58:55 VERBOSE[4225] logger.c: -- Called 
  1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 19, 
  channel 
  1 
  Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
  Complete(12) on channel 1 (index 
  0) 
  

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RE: [Asterisk-Users] RE: Asterisk is stripping my area code




Do you have 9 as a prefix in the trunk? It is actually 
ADDING a 9 to the phone number before it dials.
-Kerry


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] RE: Asterisk is stripping my area code
  No, I'm just dialing 7707190239. When I tried it with a 1, it 
  gave me the same result, a nice lady telling me "when making a local call you 
  must first dial the areacode" or words to that effect. From 
  the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c: 
  Function result is '"" 7707190069'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing NoOp("SIP/200-fa0b", "CallerID 
  set to "" 7707190069") in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "GROUP()=OUT_1") in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function 
  result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is 
  '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing GotoIf("SIP/200-fa0b", "0?108") in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "DIAL_NUMBER=17707190239") in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "DIAL_TRUNK=1") in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing AGI("SIP/200-fa0b", 
  "fixlocalprefix") in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: fixlocalprefix: Removed prefix. New number: 
  7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: 
  -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 
  VERBOSE[4242] logger.c: -- Executing 
  Set("SIP/200-fa0b", "OUTNUM=97707190239") in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set("SIP/200-fa0b", 
  "custom=ZAP/1") in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ 
  result is '0'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing GotoIf("SIP/200-fa0b", "0?16") 
  in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any 
  branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing Dial("SIP/200-fa0b", "ZAP/1/97707190239|120|W") in new stackApr 
  30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04 
  DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Called 1/97707190239Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 
  (index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 
  1So what should I have in my dialing plan to let me dial 7707190239 
  and have it use that exact number? Or do I have to dial 9 
  first?Thanks, Jim.
  On 4/30/06, Kerry 
  Garrison [EMAIL PROTECTED] 
  wrote:
  

Are you 
dialing 9 first? It is showing that the digits you dialed 
are:

9-770-719-0239
Using your dialplan you 
should be dialing 1-770-719-0239



Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- 
[EMAIL PROTECTED] 
http://www.techdatapros.com 


  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Jim LynchSent: Sunday, April 30, 2006 10:10 
  AMTo: Asterisk-Users@lists.digium.comSubject: 
  [Asterisk-Users] RE: Asterisk is stripping my area 
  code

I don't know if this helps, from the log.Jim.Apr 
30 12:58:55 VERBOSE[4225] logger.c: -- Executing 
Dial("SIP/200-5677", "ZAP/1/97707190239|120|W") in new 
stack 
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
12:58:55 DEBUG[4225] chan_zap.c: Deferring 
dialing... 
Apr 30 12:58:55 VERBOSE[4225] logger.c: -- 
Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 
19, channel 
1 
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
Complete(12) on channel 1 (index 
0) 

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[Asterisk-Users] Asterisk 1.2.7.1 Fritz!PCI or AVM A1

2006-04-30 Thread Rainer Maier

Hi all,
I am running Debian Sarge testing with Kernel 2.6.16.9.
I installed Asterisk 1.2.7.1.
I want to run an AVM Fritz!PCI v2.0 ISDN (rev 02) and an AVM A1 ISDN [Fritz]
card on separate computers.  

How can I include them in the new Asterisk ?
Probably mISDN ?
How are they configured at Asterisk ?

Any suggestions ?
What is the best way to include it ?
Any links to pages where they describe it ?

Thank you for your help.
Best regards 
Rainer


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[Asterisk-Users] Asterisk 1.2.7.1 Fritz!PCI or AVM A1

2006-04-30 Thread Rainer Maier

Hi all,
I am running Debian Sarge testing with Kernel 2.6.16.9.
I installed Asterisk 1.2.7.1.
I want to run an AVM Fritz!PCI v2.0 ISDN (rev 02) and an AVM A1 ISDN [Fritz]
card on separate computers.  

How can I include them in the new Asterisk ?
Probably mISDN ?
How are they configured at Asterisk ?

Any suggestions ?
What is the best way to include it ?
Any links to pages where they describe it ?

Thank you for your help.
Best regards 
Rainer


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Re: [Asterisk-Users] Compare to Skype

2006-04-30 Thread Time Bandit


There are 2 issues here.

1) Asterisk does not have a RTP Jitter Buffer.RTP is what is used to
transport audio for SIP (and other protocols).  This means that ANY
jitter on the SIP Phone - Asterisk link will cause audio problems.

2) Asterisk times it's outgoing audio based on the incoming audio.
Therefore, if there is jitter on the SIP Phone - Asterisk link then
Asterisk will replicate that jitter on the Asterisk - SIP Phone direction.

REMEMBER, a jitter buffer only applies on INCOMING audio (from the
standpoint of the device).

These two issues are the main reason I have not deployed remote SIP
phones for my clients.


So, he should probably try an IAX softphone and see how that compare

hth
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Re: [Asterisk-Users] newbie-too much latency

2006-04-30 Thread Time Bandit


 The latency is very high, in that, it picks up after 8 rings. I don't know
what I can tune to reduce to 2 or 3 rings. If it's of any help , I am
posting a section of the log :


Do you get CallerID on that line ?

If, in zapata.conf, you have it set to get the CallerID
(usecallerid=yes) and the line is not providing it, asterisk will wait
some time trying to get it. Usually, the CallerID is sent between the
first and second ring.

Try to disable it (usecallerid=no) and see if that help

hth
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[Asterisk-Users] Bristuff 1.2.7.1?

2006-04-30 Thread Vidar




Has anyone managed to add the bristuff patch to 
1.2.7.1 successfully?
My attempts has ended up bad, so if anyone has a 
working patchfile for 1.2.7.1 I would be grateful to receive it.

Thanks,
Vidar

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[Asterisk-Users] integrated voip originator, to digitize audio once and only once?

2006-04-30 Thread Tom Engleward

Calling  from a local extension on my local
network, I get good voice quality from asterisk, and
asterisk reliably recognizes my dtmf input.
I set up a sipphone trunk (free) and called in to it
via a separate sipphone account on another computer,
and got slightly lower, but still good, audio quality.
I set up a FWD trunk (free) and called in from the
other computer, and got somewhat lower quality, barely
acceptable.
I set up an ipkall account (free) pointing to the FWD
number, and called from a pstn number, and got less
than acceptable, barely understandable audio, with
asterisk frequently missing dtmf digits and frequently
getting duplicate digits. Finally, I set up a toll
free number at kall8.com (for a few bucks) and pointed
it to the ipkall number, called from a pstn line, and
got audio that I could barely even recognize was my
own recorded greeting being played by asterisk, and
sending accurate dtmf was hopeless.
My calls are being routed through multiple networks,
and probably passing through multiple digital-analog
conversions.
I need a toll free pstn originator which will send
calls directly to my asterisk machine via iax2, with a
grand total of only one digital-analog conversion
(besides whatever the pstn company does on its own
networks). The only possibly suitable originators I've
found want substantial money up-front, before I even
get a chance to test their audio quality. I have very
low volume requirements and only need capability for
two or three simultaneous calls. I do realize that
using free providers is not giving me an accurate
impression of the potential quality of voip, but I'm
not willing to pay somebody like broadvoice $50 up
front just for the chance to find out whether they can
provide toll free origination for me with good enough
quality that I can actually understand what my
asterisk machine is saying over the phone.
Are there an quality direct toll-free-to-iax2
originators which charge only per-minute usage fees
(and optionally a very low monthly fee), rather than
charging several tens of dollars in setup fees?

Tom


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RE: [Asterisk-Users] Digium cards, so disappointing !



On Sun, 30 Apr 2006, Boris Bakchiev wrote:


I must say, spending just a little extra to get good hardware pays off
in the long run.



If you have any questions, email.




Wow, impressive results  must say. Thanks for the specs and test results.

I had hoped that with the Dell 2850 I would have bought a decent piece of 
hardware, it isn't.


I e-mailed Dell support and asked them if it is possibel to assign a 
unique IRQ to one of the three PCI slots.


Their reply was, not possible, you are ALWAYS sharing IRQ's, I guess this 
is the reason for the poor results I'm seeing.


I will try to find a solution.

Thanks again!
Remco

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Re: [Asterisk-Users] RE: Asterisk is stripping my area code

That was it! Thanks.Jim.On 4/30/06, Kerry Garrison 
[EMAIL PROTECTED] wrote:





Do you have 9 as a prefix in the trunk? It is actually 
ADDING a 9 to the phone number before it dials.
-Kerry


  
  
  From: 

[EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] On Behalf Of Jim 
  LynchSent: Sunday, April 30, 2006 10:33 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] RE: Asterisk is stripping my area code
  No, I'm just dialing 7707190239. When I tried it with a 1, it 
  gave me the same result, a nice lady telling me when making a local call you 
  must first dial the areacode or words to that effect. From 
  the log, after using the 1: Apr 30 13:29:04 DEBUG[4242] pbx.c: 
  Function result is ' 7707190069'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing NoOp(SIP/200-fa0b, CallerID 
  set to  7707190069) in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set(SIP/200-fa0b, 
  GROUP()=OUT_1) in new stackApr 30 13:29:04 DEBUG[4242] pbx.c: Function 
  result is '1'Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ result is 
  '0'Apr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing GotoIf(SIP/200-fa0b, 0?108) in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Not taking any branchApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set(SIP/200-fa0b, 
  DIAL_NUMBER=17707190239) in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set(SIP/200-fa0b, 
  DIAL_TRUNK=1) in new stack Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing AGI(SIP/200-fa0b, 
  fixlocalprefix) in new stackApr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/fixlocalprefix Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: fixlocalprefix: Removed prefix. New number: 
  7707190239Apr 30 13:29:04 VERBOSE[4242] logger.c: 
  -- AGI Script fixlocalprefix completed, returning 0Apr 30 13:29:04 
  VERBOSE[4242] logger.c: -- Executing 
  Set(SIP/200-fa0b, OUTNUM=97707190239) in new stackApr 30 13:29:04 
  DEBUG[4242] pbx.c: Function result is 'ZAP/1'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing Set(SIP/200-fa0b, 
  custom=ZAP/1) in new stack Apr 30 13:29:04 DEBUG[4242] pbx.c: _expression_ 
  result is '0'Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Executing GotoIf(SIP/200-fa0b, 0?16) 
  in new stackApr 30 13:29:04 DEBUG[4242] pbx.c : Not taking any 
  branchApr 30 13:29:04 VERBOSE[4242] logger.c: -- 
  Executing Dial(SIP/200-fa0b, ZAP/1/97707190239|120|W) in new stackApr 
  30 13:29:04 DEBUG[4242] chan_zap.c: Dialing '97707190239' Apr 30 13:29:04 
  DEBUG[4242] chan_zap.c: Deferring dialing...Apr 30 13:29:04 VERBOSE[4242] 
  logger.c: -- Called 1/97707190239Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Exception on 19, channel 1Apr 30 13:29:05 
  DEBUG[4242] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 
  (index 0) Apr 30 13:29:07 DEBUG[4242] chan_zap.c: Exception on 19, channel 
  1So what should I have in my dialing plan to let me dial 7707190239 
  and have it use that exact number? Or do I have to dial 9 
  first?Thanks, Jim.
  On 4/30/06, Kerry 
  Garrison [EMAIL PROTECTED] 
  wrote:
  

Are you 
dialing 9 first? It is showing that the digits you dialed 
are:

9-770-719-0239
Using your dialplan you 
should be dialing 1-770-719-0239



Kerry 
GarrisonPublisher - http://VOIPSpeak.net
(949)502-7819 x200- 


[EMAIL PROTECTED] 


http://www.techdatapros.com

 


  
  
  From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of 
  Jim LynchSent: Sunday, April 30, 2006 10:10 
  AMTo: Asterisk-Users@lists.digium.comSubject: 
  [Asterisk-Users] RE: Asterisk is stripping my area 
  code

I don't know if this helps, from the log.Jim.Apr 
30 12:58:55 VERBOSE[4225] logger.c: -- Executing 
Dial(SIP/200-5677, ZAP/1/97707190239|120|W) in new 
stack 
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Dialing '97707190239'Apr 30 
12:58:55 DEBUG[4225] chan_zap.c: Deferring 
dialing... 
Apr 30 12:58:55 VERBOSE[4225] logger.c: -- 
Called 1/97707190239Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Exception on 
19, channel 
1 
Apr 30 12:58:55 DEBUG[4225] chan_zap.c: Got event Hook Transition 
Complete(12) on channel 1 (index 
0) 

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[Asterisk-Users] FreePBX in production?


Has anyone attempted to use FreePBX for a business in production mode?

Initial take is there are lots of things scripted but a lot of 
limitations in terms of supporting basic business functions. Inability 
(or lack of flexibility) is handling multiple incoming pstn lines, 
dialplan limitations, poor/no documentation, etc, to mention a few.


Maybe its just me, but it appears its no where near usable even with the 
latest beta1 code.


Is it just me or what?

Rich

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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
Maybe its just me, but it appears its no where near usable even with the 
latest beta1 code.


Its just you. I have FreePBX running on 6 production boxes across the 
country. I do very little additional scripting. 5 of the servers have a 
Eicon Diva Server V-4BRI card. The other (head office) server has a 
Digium TDM400P (4x FXO) and a Sangoma a101u (ISDN20). FreePBX manages 
all of those lines just fine.


What problems are you having? Personally, I don't have any requirements 
over and above the standard FreePBX installation. And if I do, I just go 
bug the developers until they put it in. :)


cYa,
Avi

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RE: [Asterisk-Users] FreePBX in production?

Its just you.

There is much more flexibility on handling incoming pstn lines than there
was in the last version of AMP 

If you like manually creating config files with custom settings for each
user, then a GUI is not for you.  I have several clients using freePBX
because it is easier to maintain some of the features they wanted this way
than dealing with the config files. 

Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com



 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Rich Adamson
 Sent: Sunday, April 30, 2006 2:20 PM
 To: Asterisk Users-List
 Subject: [Asterisk-Users] FreePBX in production?
 
 Has anyone attempted to use FreePBX for a business in production mode?
 
 Initial take is there are lots of things scripted but a lot 
 of limitations in terms of supporting basic business 
 functions. Inability (or lack of flexibility) is handling 
 multiple incoming pstn lines, dialplan limitations, poor/no 
 documentation, etc, to mention a few.
 
 Maybe its just me, but it appears its no where near usable 
 even with the latest beta1 code.
 
 Is it just me or what?
 
 Rich
 
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Re: [Asterisk-Users] FreePBX in production?

2006-04-30 Thread Ariel Batista


Rich Adamson wrote:

Has anyone attempted to use FreePBX for a business in production mode?


Yes it works great in business applications.


Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions. Inability
(or lack of flexibility) is handling multiple incoming pstn lines,
dialplan limitations, poor/no documentation, etc, to mention a few.


Yes it does have limitations, which you can get by with some use of there 
custom.conf files.  Documentation for asterisk and freepbx is done via the 
users and there is not much on it. But it's getting better.  You can see lot 
of info for it on http://aussievoip.com.au/wiki/ . Also there is a new 
update that you can route on the Zap channel number now.



Maybe its just me, but it appears its no where near usable even with
the latest beta1 code.


If your able to work with asterisk without a GUI it's better due to you can 
do more. But remember Freepbx has asterisk as it's main part and it works 
just the same.  It's easyer for many to use it but again this comes with 
some short commings.  But all around for the price is the best GUI out 
there.



Is it just me or what?

Rich

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Re: [Asterisk-Users] How to monitor DTMF tones in a call?

2006-04-30 Thread Moises Silva


I have uploaded a patch for some manager events that allow to know
when DTMF has been received or sent. Please take a look at this:

http://bugs.digium.com/view.php?id=6082

and if you can, test it and report feedback. Im having problems to
call the attention of bug marshalls for comitting this change. I think
this week i will enter to IRC in asterisk-dev to try to make that
bugmarshalls pay attention to it.

Best Regards

On 4/30/06, Obelix [EMAIL PROTECTED] wrote:


Is there a way to monitor the DTMF tones on a channel?

I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.

Is there way to monitor a sequence of DTMF tones and cancel the call?

If I use a SIP gateway or proxy rather than dial asterisk directly will that be
possible?

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Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [Asterisk-Users] FreePBX in production?


Avi Miller wrote:

Rich Adamson wrote:
Maybe its just me, but it appears its no where near usable even with 
the latest beta1 code.


Its just you. I have FreePBX running on 6 production boxes across the 
country. I do very little additional scripting. 5 of the servers have a 
Eicon Diva Server V-4BRI card. The other (head office) server has a 
Digium TDM400P (4x FXO) and a Sangoma a101u (ISDN20). FreePBX manages 
all of those lines just fine.


What problems are you having? Personally, I don't have any requirements 
over and above the standard FreePBX installation. And if I do, I just go 
bug the developers until they put it in. :)


Up until this beta1, I could not find a way to support the TDM400 analog 
pstn card for incoming calls. For example, pstn line #1 receives normal 
business calls, pstn line #2 receives special calls that need to be 
routed differently then the context for #1, pstn lines #3 and #4 drop 
strictly into an IVR.


With the FreePBX v2 code, I could not find a way to handle any 
incoming TDM400 calls. With the beta1 code, they've added the ability to 
address zap interfaces, but implies all four lines have to drop into the 
same context. Not usable given the above.


I've got another system that has a PRI and a TDM400, and the PRI has to 
handle DID's (which the v2 code appears it might be able to do), but 
fell short on the TDM400 non-DID calls.


After implementing the beta1 code yesterday, it looks like they removed 
several items (such as being able to edit conf files directly, crm, etc) 
with no indication as to whether that is permanent or what.


So, how did you handle the TDM400 incoming pstn calls prior to beta1?

Rich

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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
address zap interfaces, but implies all four lines have to drop into the 
same context. Not usable given the above.


The new beta (2.1) allows you to route inbound based on Zap channel -- 
you could set each channel to route to a specific destination, and 
FreePBX will create the dialplan for you.


After implementing the beta1 code yesterday, it looks like they removed 
several items (such as being able to edit conf files directly, crm, etc) 
with no indication as to whether that is permanent or what.


No, those are [EMAIL PROTECTED] specific additions and have never been part of AMP or 
FreePBX (i.e. the maintenance tab and the SugarCRM integration). FreePBX 
is merely the GUI that creates/manages your dialplan.


Prior to 2.1 and even post 2.1, I have all my TDM400P inbound calls 
coming to the same destination: The office IVR. Prior to 2.1, I used the 
catch-all destination (i.e. no DID/CID defined) for these. Post-2.1, I 
you could do it by Zap channel.


Check out #freepbx on irc.freenode.net for more support, or the 
Documentation Wiki at http://aussievoip.com.au/wiki/freePBX


FreePBX is as flexible as you make it, essentially -- if it doesn't do 
what you want it to do, feel free to write your own module (or fund the 
development of one). =D


cYa,
Avi

P.S. I'm not a FreePBX developer -- I just hang out in IRC and bug the 
real developers periodically. FreePBX does what I need, but obviously 
Your Mileage May Vary.


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Re: [Asterisk-Users] FW: NuFone Update: DIDs

2006-04-30 Thread Hermann Wecke


Matt wrote:

Is there more to this story then we know?


No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.
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Re: [Asterisk-Users] FreePBX in production?


Avi Miller wrote:

Rich Adamson wrote:
address zap interfaces, but implies all four lines have to drop into 
the same context. Not usable given the above.


The new beta (2.1) allows you to route inbound based on Zap channel -- 
you could set each channel to route to a specific destination, and 
FreePBX will create the dialplan for you.


After implementing the beta1 code yesterday, it looks like they 
removed several items (such as being able to edit conf files directly, 
crm, etc) with no indication as to whether that is permanent or what.


No, those are [EMAIL PROTECTED] specific additions and have never been part of AMP or 
FreePBX (i.e. the maintenance tab and the SugarCRM integration). FreePBX 
is merely the GUI that creates/manages your dialplan.


Well... all those things were installed with FreePBX, they just didn't 
grow there. ;)


Prior to 2.1 and even post 2.1, I have all my TDM400P inbound calls 
coming to the same destination: The office IVR. Prior to 2.1, I used the 
catch-all destination (i.e. no DID/CID defined) for these. Post-2.1, I 
you could do it by Zap channel.


Check out #freepbx on irc.freenode.net for more support, or the 
Documentation Wiki at http://aussievoip.com.au/wiki/freePBX


I've been to the wiki several times, but its very short on any any form 
of documentation. And, obviously the Handbook was borrowed from the 
[EMAIL PROTECTED] disto and doesn't actually follow the FreePBX implementations.


FreePBX is as flexible as you make it, essentially -- if it doesn't do 
what you want it to do, feel free to write your own module (or fund the 
development of one). =D


Is there a user's mailing list for this, or just the irc channel?

R.

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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
Well... all those things were installed with FreePBX, they just didn't 
grow there. ;)


Honestly, those utilities never been part of FreePBX (nor are they 
installed by FreePBX). They are only ever installed as part of 
[EMAIL PROTECTED] However, one of the FreePBX developers is currently 
implementing a lot of the stuff from [EMAIL PROTECTED] into FreePBX (like the 
Maintenance tab to hand edit the conf files and the Java SSH client).


I've been to the wiki several times, but its very short on any any form 
of documentation. And, obviously the Handbook was borrowed from the 
[EMAIL PROTECTED] disto and doesn't actually follow the FreePBX implementations.


Obviously, the Wiki documentation is a work-in-progress. Its a lot 
better than it used to be. If there are specific sections that you'd 
like more information about, please let the guys in the #freepbx channel 
know.



Is there a user's mailing list for this, or just the irc channel?


You can subscribe to the amportal-users list via the SourceForge project 
for AMP (which is now FreePBX).


cYa,
Avi

--
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  2/340 Gore Street  T: +61 (0) 3 9486 0411
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Re: [Asterisk-Users] Legacy PBX integration

2006-04-30 Thread Jerry Jones


You do not say how you have the two connected/

Are you connecting the * to stations via fxo or to lines via fxs on  
the legacy?



On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:


Hi list!

I managed to come reasonably far (farther than I thought I would)  
but have two problems.


I still need to pass calls to the Legacy PBX for Fax (I need it as  
a channel bank).


I have calls coming in into asterisk, that works fine. Based on the  
DID I can route calls to the Legacy PBX but I'm puzzled how.


I guess I need a new dial command for that? All fax calls are now  
coming in a new context which I called topbx. If I issue a dial  
command there the legacy PBX treats it as a local extension call  
and not a call from the outside.


Which dial command do I need to use to make the old PBX believe the  
call came from outside?


(All the pages I found on this subject mention something about  
retaining caller ID which is nice but now I need to retain DID info  
on the call I guess?)


Thanks for any help!


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Re: [Asterisk-Users] Asterisk is stripping my area code

2006-04-30 Thread Jerry Jones


What does you dial command look like?


On Apr 30, 2006, at 12:05 PM, Jim Lynch wrote:

I've installed [EMAIL PROTECTED] and gotten inbound calls going to an  
extension, extension to extension calling works but I'm still  
missing a few pieces.  The most annoying one is that apparently  
asterisk is stripping the area code from the number I'm dialing but  
I can't figure out how to stop it.  I have in my outbound route  
under Dialing rules:


1NXXNXX
NXXNXX

We are required to dial all 10 numbers since there are 3 area codes  
in Atlanta now.


Using freePBX admin.  I think [EMAIL PROTECTED] is version 2.7.  One  
less than the most recent, in any case.


Any suggestions would be helpful.

Thanks,
Jim.
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Re: [Asterisk-Users] FreePBX in production?




Ariel Batista wrote:

Rich Adamson wrote:

Has anyone attempted to use FreePBX for a business in production mode?


Yes it works great in business applications.


Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions. Inability
(or lack of flexibility) is handling multiple incoming pstn lines,
dialplan limitations, poor/no documentation, etc, to mention a few.


Yes it does have limitations, which you can get by with some use of 
there custom.conf files.  Documentation for asterisk and freepbx is 
done via the users and there is not much on it. But it's getting 
better.  You can see lot of info for it on 
http://aussievoip.com.au/wiki/ . Also there is a new update that you 
can route on the Zap channel number now.



Maybe its just me, but it appears its no where near usable even with
the latest beta1 code.


If your able to work with asterisk without a GUI it's better due to 
you can do more. But remember Freepbx has asterisk as it's main part 
and it works just the same.  It's easyer for many to use it but again 
this comes with some short commings.  But all around for the price is 
the best GUI out there.


I think it is better than the Signate system which costs quite a bit of 
money.



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Re: [Asterisk-Users] FW: NuFone Update: DIDs


Hermann Wecke wrote:

Matt wrote:

Is there more to this story then we know?


No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.


So why were their services cut?  Seems like an obvious omission.
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Re: [Asterisk-Users] FreePBX in production?

Well... all those things were installed with FreePBX, they just didn't 
grow there. ;)


Honestly, those utilities never been part of FreePBX (nor are they 
installed by FreePBX). They are only ever installed as part of 
[EMAIL PROTECTED] 


Actually, they were installed by FreePBX and I still have the iso disk 
to prove it, and the logo at the top of the screens say FreePBX (for the 
most part). Doesn't make a lot of difference right now, but integrating 
the various apps does have some appeal.


However, one of the FreePBX developers is currently 
implementing a lot of the stuff from [EMAIL PROTECTED] into FreePBX (like the 
Maintenance tab to hand edit the conf files and the Java SSH client).


I've never implemented [EMAIL PROTECTED], but it does appear that must have been the 
starting point for FreePBX.


I've been to the wiki several times, but its very short on any any 
form of documentation. And, obviously the Handbook was borrowed from 
the [EMAIL PROTECTED] disto and doesn't actually follow the FreePBX implementations.


Obviously, the Wiki documentation is a work-in-progress. Its a lot 
better than it used to be. If there are specific sections that you'd 
like more information about, please let the guys in the #freepbx channel 
know.



Is there a user's mailing list for this, or just the irc channel?


You can subscribe to the amportal-users list via the SourceForge project 
for AMP (which is now FreePBX).


Thanks...

R.

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[Asterisk-Users] WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'

2006-04-30 Thread hugolivude


Hi,

Red Hat 9.0
Asterisk 1.2.7.1

Whenever I start Asterisk, I am unable to call out on my SIP channel:


-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]||t|) in new stack
Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such

host: 6477235412

Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)


However, if I RELOAD my dial plan from the CLI I get this message, it
starts to work.  I think I've tracked it down to the following warning
message:


Apr 30 11:01:21 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable

to lookup ''

This message doesn't reappear when I do a RELOAD.  Anyone know what
this is all about?  My SIP.conf is below.  Notice how I've commented
out auth=md5.  This seems to have eliminated the following WARNING
message that used appear just before the one above:


Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822

add_realm_authentication: Format for authentication entry is
user[:[EMAIL PROTECTED] at line 31

Line 31 of my sip.conf was auth=md5.  I was able to get the SIP
channel working with this warning as well, but it took a lot more
RELOADs.

Any ideas?

SIP.conf
==

[general]
;
context=incoming-bogus-calls
bindport=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0  ; Address to bind to (all addresses on machine)
maxexpirey=3600   ; Must be larger than the
re-register timeout on the router
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw
;
; This section is because i'm behind nat
;

register=6477235412:mypassword@sip.unlimitel.ca/6477235412
externip=mystaticIPaddress ;Outside address
localnet=192.168.0.148/255.255.255.0 ;Inside Network
;
;
[6477235412]
type=peer
;auth=md5
username=6477235412
fromuser=6477235412
fromdomain=unlimitel.ca
secret=mypassword
host=sip.unlimitel.ca
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=g729
dtmfmode=rfc2833
insecure=very
context=incoming
;
;-
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Re: [Asterisk-Users] FreePBX in production?


Time Bandit wrote:

Up until this beta1, I could not find a way to support the TDM400 analog
pstn card for incoming calls. For example, pstn line #1 receives normal
business calls, pstn line #2 receives special calls that need to be
routed differently then the context for #1, pstn lines #3 and #4 drop
strictly into an IVR.

I have one in production with a TDM2400 with 10 incoming lines that
the first 7 go to an IVR, the 2 next ones go to a queue and the last
one go to a special application that let users record messages that
are emailed to an operator to process. The only thing I've done is
edit zapata.conf and put in different context. All the rest is done in
extension_custom.conf. AMP let me add/remove/config extensions with a
GUI while I can code anything I want in the config files.


So, how do you know which conf files one can hand edit versus those that 
might be overwritten?


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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
Actually, they were installed by FreePBX and I still have the iso disk 
to prove it


The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an 
ISO. FreePBX is simply one of the many software applications that have 
been combined to form the [EMAIL PROTECTED] distribution. :)


I've never implemented [EMAIL PROTECTED], but it does appear that must have been the 
starting point for FreePBX.


Actually, the other way around: FreePBX was probably one of the starting 
points for [EMAIL PROTECTED] :)


Hope that helps,
Avi

--
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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
So, how do you know which conf files one can hand edit versus those that 
might be overwritten?


You may only change the *_custom.conf files. :)

--
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Re: [Asterisk-Users] Legacy PBX integration

2006-04-30 Thread Lacy Moore - Aspendora

Also, what is the legacy PBX? On the Merlin Legend, for instance, there are special Class of Services that can be setup to go straight to the auto attendant. I'm not sure if that's what you need or not. The other question is, why can't you transfer the call straight to the extension the fax is on? Again, on the Legend, it defaults (I guess) to receving the extension number. For example, if my fax machine is located on ext. 170, then I just dial 170 from Asterisk on the PRI that is connected to the Legend. I didn't have to do half of what the manual says you have to do, because Asterisk takes care of all the translations.

On 4/30/06, Jerry Jones [EMAIL PROTECTED] wrote:
You do not say how you have the two connected/Are you connecting the * to stations via fxo or to lines via fxs on
the legacy?On Apr 30, 2006, at 11:22 AM, Remco Barende wrote: Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems.
I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the DID I can route calls to the Legacy PBX but I'm puzzled how.
I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial command there the legacy PBX treats it as a local extension call
and not a call from the outside. Which dial command do I need to use to make the old PBX believe the call came from outside? (All the pages I found on this subject mention something about
retaining caller ID which is nice but now I need to retain DID info on the call I guess?) Thanks for any help! ___ --Bandwidth and Colocation provided by
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc.
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Re: [Asterisk-Users] FreePBX in production?


Avi Miller wrote:

Rich Adamson wrote:
Actually, they were installed by FreePBX and I still have the iso disk 
to prove it


The ISO is [EMAIL PROTECTED], not FreePBX. FreePBX has never shipped as an 
ISO. FreePBX is simply one of the many software applications that have 
been combined to form the [EMAIL PROTECTED] distribution. :)


I've never implemented [EMAIL PROTECTED], but it does appear that must have been the 
starting point for FreePBX.


Actually, the other way around: FreePBX was probably one of the starting 
points for [EMAIL PROTECTED] :)


Now its making sense... sorry for being such a newbie on this; just 
haven't paid any attention to the [EMAIL PROTECTED], FreePBX, etc, before this past week.


Not understanding that prior to now, I apparently did install [EMAIL PROTECTED] v2.8 
from iso, which does display the FreePBX logo, and then overlaying part 
of that by installing freepbx-2.1-beta1 yesterday. Beta1 addressed the 
zap interface, but apparently undid what existed to edit conf files, 
crm, etc. That made things look like a step backwards.



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Re: [Asterisk-Users] FreePBX in production?


Rich Adamson wrote:
zap interface, but apparently undid what existed to edit conf files, 
crm, etc. That made things look like a step backwards.


Yeah, a lot of people get confused about that. I was just trying to 
clear things up. :)


--
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[Asterisk-Users] queues

2006-04-30 Thread Patrick Siglin

I am not understanding how queues are supposed to work. I am using
[EMAIL PROTECTED] and configured a queue in AMP. I have also set my static
extensions in the queue. If I set up the system to put people in the queue
on incoming it just hangs up on them. If I try to log in as an agent it says
I am logged in and then disconnects. If I do a show agents it says I'm not
logged in. I looked at some samples but not quite getting it. My queue is
100 and my two extensions are 200 and 201.

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RE: [Asterisk-Users] queues

This is not the right place for help with AAH. Use the AAH forum at sf.net.

If it is just hanging up on users, it is not configured properly.
-Kerry
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Patrick Siglin
 Sent: Sunday, April 30, 2006 8:25 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] queues
 
 I am not understanding how queues are supposed to work. I am 
 using [EMAIL PROTECTED] and configured a queue in AMP. I have 
 also set my static extensions in the queue. If I set up the 
 system to put people in the queue on incoming it just hangs 
 up on them. If I try to log in as an agent it says I am 
 logged in and then disconnects. If I do a show agents it says 
 I'm not logged in. I looked at some samples but not quite 
 getting it. My queue is 100 and my two extensions are 200 and 201.
 
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Re: [Asterisk-Users] Legacy PBX integration


Duh.. sorry for this dumb mistake, basicaly the connection is:

PRI -TE210 Port 1- * -TE210 Port 2- Legeacy PBX

Basically I need * to send whatever the telco used to send to the pri

Thanks!


On Sun, 30 Apr 2006, Jerry Jones wrote:


You do not say how you have the two connected/

Are you connecting the * to stations via fxo or to lines via fxs on the 
legacy?



On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:


Hi list!

I managed to come reasonably far (farther than I thought I would) but have 
two problems.


I still need to pass calls to the Legacy PBX for Fax (I need it as a 
channel bank).


I have calls coming in into asterisk, that works fine. Based on the DID I 
can route calls to the Legacy PBX but I'm puzzled how.


I guess I need a new dial command for that? All fax calls are now coming in 
a new context which I called topbx. If I issue a dial command there the 
legacy PBX treats it as a local extension call and not a call from the 
outside.


Which dial command do I need to use to make the old PBX believe the call 
came from outside?


(All the pages I found on this subject mention something about retaining 
caller ID which is nice but now I need to retain DID info on the call I 
guess?)


Thanks for any help!


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Re: [Asterisk-Users] Legacy PBX integration



Thanks!  The PBX is a Alcatel Novo Supreme. All calls go straight into the 
auto attendant no matter whoch extension I dial on the Zap group the PBX 
is connected to.  I tried dialling in by hand using several combinations 
but I always get the auto attendant.


How do you transfer the call straight to the extension the fax is on? I 
guess using a Dial command fro *?


I suspect that the PBX is missing some signalling.

Thanks!

On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote:


Also, what is the legacy PBX?  On the Merlin Legend, for instance, there are
special Class of Services that can be setup to go straight to the auto
attendant.  I'm not sure if that's what you need or not.  The other question
is, why can't you transfer the call straight to the extension the fax is
on?  Again, on the Legend, it defaults (I guess) to receving the extension
number.  For example, if my fax machine is located on ext. 170, then I just
dial 170 from Asterisk on the PRI that is connected to the Legend.  I didn't
have to do half of what the manual says you have to do, because Asterisk
takes care of all the translations.

On 4/30/06, Jerry Jones [EMAIL PROTECTED] wrote:


You do not say how you have the two connected/

Are you connecting the * to stations via fxo or to lines via fxs on
the legacy?


On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:

 Hi list!

 I managed to come reasonably far (farther than I thought I would)
 but have two problems.

 I still need to pass calls to the Legacy PBX for Fax (I need it as
 a channel bank).

 I have calls coming in into asterisk, that works fine. Based on the
 DID I can route calls to the Legacy PBX but I'm puzzled how.

 I guess I need a new dial command for that? All fax calls are now
 coming in a new context which I called topbx. If I issue a dial
 command there the legacy PBX treats it as a local extension call
 and not a call from the outside.

 Which dial command do I need to use to make the old PBX believe the
 call came from outside?

 (All the pages I found on this subject mention something about
 retaining caller ID which is nice but now I need to retain DID info
 on the call I guess?)

 Thanks for any help!


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--
Lacy Moore
Aspendora, Inc.


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Re: [Asterisk-Users] Legacy PBX integration

2006-04-30 Thread Lacy Moore - Aspendora

Hmm... In my case, it could be just dumb luck. I found some instructions on setting up DID on my pbx, and started that. Part way through, I wasn't sure what the rest of the instructions were talking about and felt I was getting in too deep. So, I decided to see what would happen if I just tried it. It worked. Since this is only a temporary solution until we move completely off the pbx to Asterisk, I felt like I didn't need to find out why it worked, just be thankful that it worked.

It sounds like your system is just answering the line and not paying attention to any DID information. I was able to find a lot of information on the tek-tips.com forums for the Merlin Legend. You may try a search on there and see.

Was DID working in the past, or have you just added it with the addition of the Asterisk system?
On 4/30/06, Remco Barende [EMAIL PROTECTED] wrote:
Thanks!The PBX is a Alcatel Novo Supreme. All calls go straight into theauto attendant no matter whoch extension I dial on the Zap group the PBX
is connected to.I tried dialling in by hand using several combinationsbut I always get the auto attendant.How do you transfer the call straight to the extension the fax is on? Iguess using a Dial command fro *?
I suspect that the PBX is missing some signalling.Thanks!On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote: Also, what is the legacy PBX?On the Merlin Legend, for instance, there are
special Class of Services that can be setup to go straight to the auto attendant.I'm not sure if that's what you need or not.The other question is, why can't you transfer the call straight to the extension the fax is
on?Again, on the Legend, it defaults (I guess) to receving the extension number.For example, if my fax machine is located on ext. 170, then I just dial 170 from Asterisk on the PRI that is connected to the Legend.I didn't
have to do half of what the manual says you have to do, because Asterisk takes care of all the translations. On 4/30/06, Jerry Jones [EMAIL PROTECTED]
wrote: You do not say how you have the two connected/ Are you connecting the * to stations via fxo or to lines via fxs on the legacy?
On Apr 30, 2006, at 11:22 AM, Remco Barende wrote: Hi list! I managed to come reasonably far (farther than I thought I would) but have two problems.
I still need to pass calls to the Legacy PBX for Fax (I need it as a channel bank). I have calls coming in into asterisk, that works fine. Based on the
DID I can route calls to the Legacy PBX but I'm puzzled how. I guess I need a new dial command for that? All fax calls are now coming in a new context which I called topbx. If I issue a dial
command there the legacy PBX treats it as a local extension call and not a call from the outside. Which dial command do I need to use to make the old PBX believe the
call came from outside? (All the pages I found on this subject mention something about retaining caller ID which is nice but now I need to retain DID info
on the call I guess?) Thanks for any help!___ --Bandwidth and Colocation provided by
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-- Lacy MooreAspendora, Inc.
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Re: [Asterisk-Users] Legacy PBX integration