date:20090423

On 23/04/2009 11:12 p.m., Geraint Lee wrote:
 Hi Guys,

 I have a strong feeling the loads on my servers will be shooting up
 soon... anyone got any idea how many calls i can expect to put through a
 DL360:
 Dual Quad Core 2.33ghz
 4gb RAM with 1gb allocated for a ramdisk (call recordings)

 This server is recording calls (mixmonitor), codec is gsm (no conversion).

 I know there's a lot of other things to consider like AGI scripts and
 such things but i'd like to know what the capacity should be simply for
 sip registrations (which are in conf files) and calls (usually between
 20 and 60 concurrent calls at present (around 12,000 calls a day - so
 relatively low volume). No voicemail or meetme.

 I expect to be pushing 300-400 concurrent calls within the next 2 months.

 Next question... do i need to be looking at openSIPS or something
 similar to handle registrations?

 Any hints, tips and things to watch out for with a larger volume would
 be great.

Your biggest problem is likely to be the concurrent recording of channels.

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] Compact, fanless appliance?

2009-04-23 Thread Chris Bagnall

 So at this point, it seems like it boils down to this:
 Soekris
 PCEngines
 Atcom (IP01: £160+VAT)
 Herologic (HL-463: $259)
 uCpbx (235.00 EUR)
 AstBoxes (168.00 EUR)
 Gumstix
 HP Thinclient t5720

Probably worth adding the Asus eeeBox to the list. It doesn't have space for a 
PCI card and isn't *strictly* fanless (but the fan tends to be off most of the 
time), but we've used them for a few recent asterisk installs and been 
decidedly impressed with them. Considerably more powerful than most of the 
above, which might make a difference if your end users are doing transcoding.

Regards,

Chris


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Re: [asterisk-users] Asterisk and HUD server

2009-04-23 Thread Marco Sambo

Well, I see the rpm but my Asterisk box has Debian Linux, and I'm a little
afraid to use alien package to transform rpm to deb. Has HUDlite Server
source?? Like in tar.gz??



2009/4/23 David Klaverstyn d...@klaverstyn.com.au

  Hi Marco,



 Try this:
 http://yum.trixbox.org/centos/4/RPMS/hudlite-server-1.4.32-1.i386.rpm



 Regards

 David.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
 *Sent:* Thursday, 23 April 2009 7:29 PM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Asterisk and HUD server



 Hi,
 someone has installed on an Asterisk box (not Trixbox) with Debian Linux,
 the HUDlite Server?
 Can someone help me in retrieve and install packages???

 Thanks all

 Marco

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Re: [asterisk-users] Asterisk Capacity

2009-04-23 Thread z gringo


You don't say how many SIP registrations you are doing, but I have several 
servers with betwen 1000 and 1200 simultaneous registered users 24/7.   When we 
had the registrations in realtime (cached) with the mysql connector, everything 
started failing around 600 users.  With the ODBC connector we have not had that 
problem.  Ditto for putting the users in .conf files.  My servers all have 
around 300 to 400 simultaneous calls during peak periods, and I have a 1GB 
ramdisk for recordings.We are only recording a tiny percentage of those 
calls.  MySQL is running on a separate server dedicated to Databases.  The 
asterisks connect to the realtime DB via a private network on a second nic.

My thoughts are these:
1.  Asterisk is not going to be able to handle much more registration traffic 
than around 1200 registered users. (this depends on a whole lot of things 
though).  Eventually, it will need to be offloaded to something like OpenSIPs
2.  Somewhere around 800 simultaneous calls is about the most asterisk is going 
to be able to push.
3.  Your problem is going to be the call recording.  If you are trying to 
record all the calls on your server or even a large percentage of them, that is 
going to be your first problem area.

Another important thing to consider is how many calls you are setting up and 
tearing down each second.   If you have a bunch of users dialing manually and 
making long calls, that will be a lot easier to handle than if you have 3 
predictive dialers running against your server trying to bring up 30 calls per 
second.  If you are doing something like that, you will probably need to 
distribute accross multiple servers.



Date: Thu, 23 Apr 2009 12:12:35 +0100
From: gera...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Capacity

Hi Guys,

I have a strong feeling the loads on my servers will be shooting up soon... 
anyone got any idea how many calls i can expect to put through a
DL360:
Dual Quad Core 2.33ghz
4gb RAM with 1gb allocated for a ramdisk (call recordings)


This server is recording calls (mixmonitor), codec is gsm (no conversion).

I know there's a lot of other things to consider like AGI scripts and such 
things but i'd like to know what the capacity should be simply for sip 
registrations (which are in conf files) and calls (usually between 20 and 60 
concurrent calls at present (around 12,000 calls a day - so relatively low 
volume). No voicemail or meetme.


I expect to be pushing 300-400 concurrent calls within the next 2 months.

Next question... do i need to be looking at openSIPS or something similar to 
handle registrations?

Any hints, tips and things to watch out for with a larger volume would be great.


Cheers

Geraint

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[asterisk-users] UserEvent doc : is Uniqueid mandatory in 1.6

Hello,

I'm using 1.6.1-rc4.
When sending a userevent, (with UserEvent(MyEvent); in extensions.ael),
I've got :
Event: UserEvent
Privilege: user,all
UserEvent: MyEvent

I can't see any Uniqueid field as mentioned
http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or
http://www.the-asterisk-book.com/unstable/applikationen-userevent.html

Is this Uniqueid mandatory ?

Regards
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Re: [asterisk-users] Parked calls for multiple customers

2009-04-23 Thread Mike

No, but as I understand it 1.6 would have that possibility.

Mike
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of carl Lougher
 Sent: Thursday, April 23, 2009 4:54
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Parked calls for multiple customers
 
 
 Hi,
 
 Is there any method of getting call park working on different numbers for
 different customers on the same asterisk server?
 Currently running asterisk 1.4.23.1
 
 Cheers!!
 
 
 
 
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[asterisk-users] Asterisk Double Invite

2009-04-23 Thread Khaled W. Chehab

Dears

 

My scenario is incoming call to asterisk which asterisk in its term  will
dial it through its trunk .

I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can  see in the debugging below

The first is the default and the second when asterisk receives a 200 OK 

Why Asterisk(B2BUA) is  acting like that,  and from where I can get the
asterisk sip dial call flow 

 

Why Asterisk is sending double invite 

GW CLIENT IP=192.168.5.100

Asterisk IP=192.168.5.150

Termination GW=192.168.5.200

 

Capturing on eth0

  4.865698 192.168.5.100- 192.168.5.150 SIP/SDP Request: INVITE
sip:3316234335...@192.168.5.150, with session description

  4.871457 192.168.5.150 - 192.168.5.100SIP Status: 100 Trying

  4.876797 192.168.5.150 - 192.168.5.100SIP/SDP Status: 183 Session
Progress, with session description

  6.947270 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE
sip:3316234335...@192.168.5.200, with session description

  6.949157 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying

 12.759311 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 183 Session
Progress, with session description

 16.236320 192.168.5.200 - 192.168.5.150 SIP Status: 180 Ringing

 20.250002 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with
session description

 20.250395 192.168.5.150 - 192.168.5.200 SIP Request: ACK
sip:3316234335...@192.168.5.200:5060

 20.251267 192.168.5.150 - 192.168.5.100SIP/SDP Status: 200 OK, with
session description

 20.251752 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE
sip:3316234335...@192.168.5.200:5060, with session description

 20.252986 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying

 20.274788 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with
session description

 20.275143 192.168.5.150 - 192.168.5.200 SIP Request: ACK
sip:3316234335...@192.168.5.200:5060

 20.569819 192.168.5.100- 192.168.5.150 SIP Request: ACK
sip:3316234335...@192.168.5.150

 20.570303 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE
sip:551130338...@192.168.5.100, with session description

 20.900485 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with
session description

 20.902604 192.168.5.150 - 192.168.5.100SIP Request: ACK
sip:551130338...@192.168.5.100

 32.468119 192.168.5.200 - 192.168.5.150 SIP Request: BYE
sip:551130338...@192.168.5.150

 32.468411 192.168.5.150 - 192.168.5.200 SIP Status: 200 OK

 32.468750 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE
sip:551130338...@192.168.5.100, with session description

 32.822154 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with
session description

 32.822478 192.168.5.150 - 192.168.5.100SIP Request: ACK
sip:551130338...@192.168.5.100

 32.822928 192.168.5.150 - 192.168.5.100SIP Request: BYE
sip:551130338...@192.168.5.100

 33.140288 192.168.5.100- 192.168.5.150 SIP Status: 200 OK

 



 

 



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Re: [asterisk-users] Compact, fanless appliance?

2009-04-23 Thread Scott L. Lykens

 So at this point, it seems like it boils down to this:
 Soekris
 PCEngines
 Atcom (IP01: £160+VAT)
 Herologic (HL-463: $259)
 uCpbx (235.00 EUR)
 AstBoxes (168.00 EUR)
 Gumstix
 HP Thinclient t5720

I recently custom built an Intel Atom 330 Mini-ITX based system for a client 
who then took it by plane out of the country. Including SSD (only moving part 
in the entire system is a fan on the northbridge) I believe we had $300 in it 
and that was probably a little high with the components we selected. It is not 
intended to handle a ton of calls but will suit his small remote office quite 
nicely. We probably could have built it for $200 if we wanted.

sl

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Re: [asterisk-users] Asterisk Capacity

2009-04-23 Thread Geraint Lee

Thanks for that, it's pretty much confirming what i first anticipated... my
intentions are as follows:

agents register with opensips, opensips clusters a set of call recording
servers which then connect to our border servers which will save cdr and
choose the sip/iax provider to send the call to.

and for my predictive dialer, each server will spool as many calls as they
can before i see performance issues when they have an answer they too will
connect to the opensips server to get a call recording server which in turn
will pass it on to the agent again via opensips.

simples :)

looks like i need to install and learn opensips since this whole scenario
seems to be heavily relying on it :)

Cheers

2009/4/23 z gringo z_gri...@hotmail.com

You don't say how many SIP registrations you are doing, but I have several
servers with betwen 1000 and 1200 simultaneous registered users 24/7. When
we had the registrations in realtime (cached) with the mysql connector,
everything started failing around 600 users. With the ODBC connector we
have not had that problem. Ditto for putting the users in .conf files. My
servers all have around 300 to 400 simultaneous calls during peak periods,
and I have a 1GB ramdisk for recordings.We are only recording a tiny
percentage of those calls. MySQL is running on a separate server dedicated
to Databases. The asterisks connect to the realtime DB via a private
network on a second nic.

My thoughts are these:
1. Asterisk is not going to be able to handle much more registration
traffic than around 1200 registered users. (this depends on a whole lot of
things though). Eventually, it will need to be offloaded to something like
OpenSIPs
2. Somewhere around 800 simultaneous calls is about the most asterisk is
going to be able to push.
3. Your problem is going to be the call recording. If you are trying to
record all the calls on your server or even a large percentage of them, that
is going to be your first problem area.

Another important thing to consider is how many calls you are setting up
and tearing down each second. If you have a bunch of users dialing
manually and making long calls, that will be a lot easier to handle than if
you have 3 predictive dialers running against your server trying to bring up
30 calls per second. If you are doing something like that, you will
probably need to distribute accross multiple servers.

--
Date: Thu, 23 Apr 2009 12:12:35 +0100
From: gera...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Capacity

Hi Guys,

I have a strong feeling the loads on my servers will be shooting up soon...
anyone got any idea how many calls i can expect to put through a
DL360:
Dual Quad Core 2.33ghz
4gb RAM with 1gb allocated for a ramdisk (call recordings)

This server is recording calls (mixmonitor), codec is gsm (no conversion).

I know there's a lot of other things to consider like AGI scripts and such
things but i'd like to know what the capacity should be simply for sip
registrations (which are in conf files) and calls (usually between 20 and 60
concurrent calls at present (around 12,000 calls a day - so relatively low
volume). No voicemail or meetme.

I expect to be pushing 300-400 concurrent calls within the next 2 months.

Next question... do i need to be looking at openSIPS or something similar
to handle registrations?

Any hints, tips and things to watch out for with a larger volume would be
great.

Cheers

Geraint

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[asterisk-users] Zaptel Not Releasing Channel (PRI)

I have an issue with one of my installations running Asterisk 1.4.20 that I
need some help with.

Not sure if this is new or not but Zaptel or Libpri is not releasing
channels properly.  I have had issues with calls that stay up for eight
hours, long distance on the telco side, so it is more than just a nuisance.

Does anyone know if there was a related bug in the 1.4 branch of Asterisk,
Zaptel, Libpri?  I searched and did not find anything except references to
an issue a few years ago.  Any easy pointers before I start to dig more?

The box is running with a Digium Quad T1 card.  One port is set to normal
PRI settings for US to the telco.  Another span is connected to an Adit
Channel Bank and breakout box to twenty or so phones.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
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Re: [asterisk-users] Zaptel Not Releasing Channel (PRI)

2009-04-23 Thread Eve-Ellen Cole

I had this problem with a box that I was using Festival tts on.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Thursday, April 23, 2009 9:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zaptel Not Releasing Channel (PRI)

I have an issue with one of my installations running Asterisk 1.4.20 that
I need some help with.

Not sure if this is new or not but Zaptel or Libpri is not releasing
channels properly.  I have had issues with calls that stay up for eight
hours, long distance on the telco side, so it is more than just a
nuisance.

Does anyone know if there was a related bug in the 1.4 branch of Asterisk,
Zaptel, Libpri?  I searched and did not find anything except references to
an issue a few years ago.  Any easy pointers before I start to dig more?

The box is running with a Digium Quad T1 card.  One port is set to
normal PRI settings for US to the telco.  Another span is connected to
an Adit Channel Bank and breakout box to twenty or so phones.

-- 
Thanks,
Steve Totaro 
+18887771888 (Toll Free)
+12409381212 (Cell)

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Re: [asterisk-users] Cause 34 still there

2009-04-23 Thread Steve Davies

2009/4/23 Steve Davies davies...@gmail.com:
 My comment, (forwarded from Bristuff list) - A few people are seeing a
 Cause 34 (congestion) from ISDN installs, where there clearly is an
 available channel. This was originally related to Bristuff as it
 happens to ISDN2 users, but there is at least one report of an
 unpatched 1.6.x user seeing the same issue.

[snip]

Something I just found, which others might want to consider trying if
they get cause 34 issues is uncommenting the following line in
libpri's Makefile:

#LAYER2ALWAYSUP=-DLAYER2ALWAYSUP

This (in theory anyway) simply causes libpri to attempt to
re-negotiate L2 if the far end closes it down with a 'DISC'.

Regards,
Steve

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[asterisk-users] Do I need G729 codec for wholesale ?

2009-04-23 Thread Dusan Djordjevic



Hi all,

I am new to Asterisk, so I have few questions. I intent to run it wholesale 
traffic termination. Apart Asterisk, I want to use Asterist2Billing for 
billing. Here are my questions:

-Since my main area of work will be wholesale, is Asterisk2Billing good 
solution or you have better suggestion ? I prefer open source solutions, but 
commercial are OK also.
-I will need to use H323 and G729 mainly. Is H323 supported properly ?
-Since I will doing wholesale, traffic will just pass through server, no 
encoding/decoding on it. Would I need g729 licenses in this case ? If not, 
please point me to some resource how to set it up properly.
-At the end, do you have any other advice on how to do what I want to do. Maybe 
pointer to some good tutorial or something like that.

Thank you in advance.

Best regards,
Dusan



  

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[asterisk-users] Dial-out via AMI

2009-04-23 Thread Nhadie

Hi,

i'm currently using Originate command on AMI, i can call a certain 
channel like a SIP user SIP/1000 then once 1000 is answered it dials out 
to amobile or landline.

Would just like to know if i can use AMI to dialout to a mobile or 
landline first (instead of SIP user) and once answered, dial another 
mobile or landline again.

If not is it possible to call a macro from the AMI? i think i can 
probably use AGI for this, but i don't know if i can call a macro from 
the AMI command.

Thanks in advance.

Regards,
nhadie

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[asterisk-users] Fritz USB 2.1 on Asterisk 1.4.22 / trixbox

2009-04-23 Thread Akos Gabriel

Dear Sirs,

I've a Fritz USB 2.1 card like this:

Bus 002 Device 003: ID 057c:1900 AVM GmbH ISDN-Controller FRITZ!Card v2.1

I'd like to use it with trixbox/asterisk.
I've already some experience with other cards and mISDN, but I can't make
this card work with it.
I've downloaded the driver from ftp.avm.de but don't know what to do with
it.
I think I should load the firmware, but how?
I've googled a lot, I found a lot of outdated / not relevant info... :(
Please point me to a good direction (url, some keywords...)

Thx,
Akos
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[asterisk-users] AMD Not Working

2009-04-23 Thread Sam Hawkin

Hi All,

I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.

can any one suggest us, what might be the problem
and possible solution to it.

below is the log

 -- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256]
-- AMD: HANGUP

any help is highly appreciated.

Thanks.
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Re: [asterisk-users] AMD Not Working

2009-04-23 Thread Matt Florell

On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:


 Hi All,

 I am trying to use the AMD (Answering Machine Detect).
 But it is not sending the AMD_Status as either
 the Human or Machine, it hangs up in middle.

 can any one suggest us, what might be the problem
 and possible solution to it.

 below is the log

  -- Executing AMD(SIP/sip-ffe0, ) in new stack
 -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
 Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using
 the default parameters.
  -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence
 [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence
 [50] maximumNumberOfWords [5] silenceThreshold [256]
 -- AMD: HANGUP

What version of Asterisk are you running this on?

What is the dialplan path that this is running through?

MATT---

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Re: [asterisk-users] Zaptel Not Releasing Channel (PRI)

2009-04-23 Thread Jared Smith

On Thu, 2009-04-23 at 09:18 -0400, Steve Totaro wrote:
 Not sure if this is new or not but Zaptel or Libpri is not releasing
 channels properly.  I have had issues with calls that stay up for
 eight hours, long distance on the telco side, so it is more than just
 a nuisance.

Have you examined the output of core show channels to see what
application the hung channels are in?  I'd start there.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] AMD Not Working

2009-04-23 Thread Ruddy Gbaguidi

Maybe  the customer hangs up during the AMD analysis or you don't have any
audio coming to asterisk through your sip channel.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Hawkin
Sent: April-23-09 11:00 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AMD Not Working

 

Hi All,

I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.

can any one suggest us, what might be the problem
and possible solution to it.

below is the log

 -- Executing AMD(SIP/sip-ffe0, ) in new stack
-- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4)
Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using
the default parameters.
-- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300]
totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50]
maximumNumberOfWords [5] silenceThreshold [256] 
-- AMD: HANGUP

any help is highly appreciated.

Thanks.

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[asterisk-users] AGI PHP script

2009-04-23 Thread James A. Shigley

I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing

 

 

This is the CLI output

-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1)
in

new stack

-- Goto (newhire,s,1)

-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:3] Monitor(DAHDI/50-1,
wav,/var/lib/asterisk/soun

ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack

-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new
stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

-- DAHDI/50-1AGI Script newhire.php completed, returning 0

-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

-- Hungup 'DAHDI/50-1'

 

Here is the script

 

 

#!/usr/bin/php5 

?php

 

// Get AGI vars from *

 

 $agivars = array();

 while (!feof(STDIN)) {

 $agivar = trim(fgets(STDIN));

 if ($agivar === '') {

 break;

 }

 $agivar = explode(':', $agivar);

 $agivars[$agivar[0]] = trim($agivar[1]);

 }

 extract($agivars);

 

// Variable Declarations

 

$agi_uniqueid;

$agi_callerid;

$agi_calleridname;

$agi_extension;

$agi_uniqueid;

$UNIQUEID = $agi_uniqueid;

$CALLERID = $agi_callerid;

$EXTEN = $agi_extension;

$attachment =
/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav
;

$from = @xxx.com; 

$to =j...@answeringserv.com ;

$subject=New Applicant;

$headers = From: $from;

$message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;

mail($to,$subject,$message,$headers);

?

 

 

So is it anything obviously wrong with the script I'm missing?

 

Besides something not being configured in php.ini correctly any other
ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments. 

 

Common sense is the collection of prejudices acquired by age eighteen.
-- Albert Einstein 

Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy. -- Albert Einstein

I know a little of everything, but a lot of nothing

 

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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread Geraint Lee

Check you can run the script from th ecommand line and successfully send
email... have you considered using phpagi for your scripts?

2009/4/23 James A. Shigley j...@answeringserv.com

  I have the below script that doesn’t seem to be working. I don’t know if
 I have something in the script wrong that I am just missing. Or if I don’t
 have the php.ini set correctly for emailing





 This is the CLI output

 -- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1)
 in

 new stack

 -- Goto (newhire,s,1)

 -- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack

 -- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack

 -- Executing [...@newhire:3] Monitor(DAHDI/50-1,
 wav,/var/lib/asterisk/soun

 ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack

 -- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new
 stack

 -- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

 -- DAHDI/50-1AGI Script newhire.php completed, returning 0

 -- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

 -- Hungup 'DAHDI/50-1'



 Here is the script





 #!/usr/bin/php5

 ?php



 // Get AGI vars from *



  $agivars = array();

  while (!feof(STDIN)) {

  $agivar = trim(fgets(STDIN));

  if ($agivar === '') {

  break;

  }

  $agivar = explode(':', $agivar);

  $agivars[$agivar[0]] = trim($agivar[1]);

  }

  extract($agivars);



 // Variable Declarations



 $agi_uniqueid;

 $agi_callerid;

 $agi_calleridname;

 $agi_extension;

 $agi_uniqueid;

 $UNIQUEID = $agi_uniqueid;

 $CALLERID = $agi_callerid;

 $EXTEN = $agi_extension;

 $attachment =
 /var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;

 $from = �...@xxx.com;

 $to =j...@answeringserv.com ;

 $subject=New Applicant;

 $headers = From: $from;

 $message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;

 mail($to,$subject,$message,$headers);

 ?





 So is it anything obviously wrong with the script I’m missing?



 Besides something not being configured in php.ini correctly any other
 ideas?



 James Shigley

 *Monroe Telephone Answering Service*

 409-981-9213**

 Infinity 5.5,UC 4.02.3803, Blink 3.0.104

 Ecreator:2.21, eResponse 1.1.7

 Webportal,WebApps,



 CONFIDENTIALITY NOTICE: This email, including any attachments, contains
 information which may be confidential or privileged. The information is
 intended to be for the use of the individual or entity named above. If you
 are not the intended recipient, be aware that any disclosure, copying,
 distribution or use of the contents of this information is prohibited. If
 you have received this email in error, please notify the sender immediately
 by reply to sender only message and destroy all electronic and hard copies
 of the communication, including attachments.



 Common sense is the collection of prejudices acquired by age eighteen. --
 Albert Einstein

 Once you can accept the universe as matter expanding into nothing that is
 something,wearing stripes with plaid comes easy. -- Albert Einstein

 I know a little of everything, but a lot of nothing



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Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help

2009-04-23 Thread Dan Austin

Jimmy wrote:

Second Call out the asterisk console looks like 
this-:
-- Executing [92952...@internal:1] Dial(SIP/222-09ab3588, 
SIP/Cisco1760/2952210) in new stack
-- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call 
from '222' to extension '2952210' rejected because extension not found.
-- Got SIP response 486 Busy here back from 172.17.2.1
-- SIP/Cisco1760-09ab7cf8 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing [92952...@internal:2] Congestion(SIP/222-09ab3588, ) in 
new stack
  == Spawn extension (internal, 92952210, 2) exited non-zero on 
'SIP/222-09ab3588'
localhost*CLI


--sip.conf -
[general]
bindaddr=0.0.0.0

[Cisco1760]
context=incoming_calls
type=friend
host=172.17.2.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very


--extensions.conf
[globals]
OUTBOUNDTRUNK=SIP/Cisco1760


[outbound-local]
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,n,Congestion()
exten = _9NXX,n,Hangup()

---Cisco 1760 config --
dial-peer voice 100 pots  (This line that is set to preference 2 does not work)
 huntstop
 preference 2
 destination-pattern .T
 port 0/0
 forward-digits all
!
dial-peer voice 2212 pots(This line that is set to Preference 1 is the one 
that works)
 huntstop
 preference 1
 destination-pattern .T
 port 0/1
 forward-digits all



You do not want to use huntstop on the dialpeers in this situation.
The huntstop option tells the call routing function in the router to
stop search for a call route if it encounters a failure.

Call number 2 hits dialpeer 1, finds it busy and the huntstop causes
the processing to stop.

Dan

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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread Nhadie


you can try enabling agi debug on your console, you might be able to see 
if there's an error on your agi script.

nhadie


James A. Shigley wrote:
 I have the below script that doesn’t seem to be working. I don’t know if 
 I have something in the script wrong that I am just missing. Or if I 
 don’t have the php.ini set correctly for emailing
 
  
 
  
 
 This is the CLI output
 
 -- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1) in
 
 new stack
 
 -- Goto (newhire,s,1)
 
 -- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack
 
 -- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack
 
 -- Executing [...@newhire:3] Monitor(DAHDI/50-1, 
 wav,/var/lib/asterisk/soun
 
 ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack
 
 -- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new stack
 
 -- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php
 
 -- DAHDI/50-1AGI Script newhire.php completed, returning 0
 
 -- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'
 
 -- Hungup 'DAHDI/50-1'
 
  
 
 Here is the script
 
  
 
  
 
 #!/usr/bin/php5
 
 ?php
 
  
 
 // Get AGI vars from *
 
  
 
  $agivars = array();
 
  while (!feof(STDIN)) {
 
  $agivar = trim(fgets(STDIN));
 
  if ($agivar === '') {
 
  break;
 
  }
 
  $agivar = explode(':', $agivar);
 
  $agivars[$agivar[0]] = trim($agivar[1]);
 
  }
 
  extract($agivars);
 
  
 
 // Variable Declarations
 
  
 
 $agi_uniqueid;
 
 $agi_callerid;
 
 $agi_calleridname;
 
 $agi_extension;
 
 $agi_uniqueid;
 
 $UNIQUEID = $agi_uniqueid;
 
 $CALLERID = $agi_callerid;
 
 $EXTEN = $agi_extension;
 
 $attachment = 
 /var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;
 
 $from = �...@xxx.com;
 
 $to =j...@answeringserv.com ;
 
 $subject=New Applicant;
 
 $headers = From: $from;
 
 $message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;
 
 mail($to,$subject,$message,$headers);
 
 ?
 
  
 
  
 
 So is it anything obviously wrong with the script I’m missing?
 
  
 
 Besides something not being configured in php.ini correctly any other ideas?
 
  
 
 James Shigley
 
 *Monroe Telephone Answering Service*
 
 409-981-9213**
 
 Infinity 5.5,UC 4.02.3803, Blink 3.0.104
 
 Ecreator:2.21, eResponse 1.1.7
 
 Webportal,WebApps,
 
  
 
 CONFIDENTIALITY NOTICE: This email, including any attachments, contains 
 information which may be confidential or privileged. The information is 
 intended to be for the use of the individual or entity named above. If 
 you are not the intended recipient, be aware that any disclosure, 
 copying, distribution or use of the contents of this information is 
 prohibited. If you have received this email in error, please notify the 
 sender immediately by reply to sender only message and destroy all 
 electronic and hard copies of the communication, including attachments.
 
  
 
 Common sense is the collection of prejudices acquired by age eighteen. 
 -- Albert Einstein
 
 Once you can accept the universe as matter expanding into nothing that 
 is something,wearing stripes with plaid comes easy. -- Albert Einstein
 
 I know a little of everything, but a lot of nothing
 
  
 
 
 
 
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Re: [asterisk-users] 2B Channel Transfer on XO-based T1

Martin wrote:
 pri debug span 1
 
 should show you the ISDN messages for 2BCT if there are any
 
 Also someone should have told you that the 2BCT code is by default not 
 compiling
 and you could enable it by editing chan_dahdi.c and adding
 
 #define PRI_2BCT
 
 Also since this flag is not present anywhere else in the code
 
 grep PRI_2BCT * -r
 channels/chan_dahdi.c:#ifdef PRI_2BCT
 channels/chan_dahdi.c:#ifdef PRI_2BCT
 
 it might actually only work in the version of Asterisk it was introduced for 
 ...

This flag is defined inside of libpri.h, which is included by 
chan_dahdi.c, which is why you do not see it inside chan_dahdi.c.

2BCT will automatically compile by default if the version of libpri you 
have support 2BCT.  If you have any version of libpri newer than a year 
or two ago, it supports all the currently supported switchtypes.  In 
fact, the earliest version of 2BCT supported was done probably 3 or 4 
years ago (RLT for DMS switches), so even very old versions of libpri 
will support compilation of that code.

Matthew Fredricikson
Digium, Inc.

 
 Martin
 
 On Wed, Apr 15, 2009 at 8:24 AM, Ron Joffe rjo...@sienatech.com wrote:
 On Tuesday 14 April 2009 18:41, Jared Smith wrote:
 Some time after the second leg of
 the call has answered, Asterisk will send a facility message to the CO
 switch saying Hey, mind bridging these two calls on your end, so I can
 free up the channels on my end?  If the switch says OK, you'll see
 the calls disappear from Asterisk (and the people on the calls won't
 know the difference).  Otherwise, the calls will continue to be bridged
 by Asterisk.
 Jared,

 Is there a debug mode where I can find these specific messages?

 Thanks,

 Ron


 --
 Ron Joffe
 Siena Tech, Inc.
 3319 Willow Glen Drive
 Oak Hill, VA 20171
 (919) 928-0404

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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread Ruddy Gbaguidi

First run 

/var/lib/asterisk/agi-bin/newhire.php

 

From linux command line to see if you don't have any error and that your AGI
is executable.

 

Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI PHP script

 

I have the below script that doesn't seem to be working. I don't know if I
have something in the script wrong that I am just missing. Or if I don't
have the php.ini set correctly for emailing

 

 

This is the CLI output

-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1) in

new stack

-- Goto (newhire,s,1)

-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:3] Monitor(DAHDI/50-1,
wav,/var/lib/asterisk/soun

ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack

-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

-- DAHDI/50-1AGI Script newhire.php completed, returning 0

-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

-- Hungup 'DAHDI/50-1'

 

Here is the script

 

 

#!/usr/bin/php5 

?php

 

// Get AGI vars from *

 

 $agivars = array();

 while (!feof(STDIN)) {

 $agivar = trim(fgets(STDIN));

 if ($agivar === '') {

 break;

 }

 $agivar = explode(':', $agivar);

 $agivars[$agivar[0]] = trim($agivar[1]);

 }

 extract($agivars);

 

// Variable Declarations

 

$agi_uniqueid;

$agi_callerid;

$agi_calleridname;

$agi_extension;

$agi_uniqueid;

$UNIQUEID = $agi_uniqueid;

$CALLERID = $agi_callerid;

$EXTEN = $agi_extension;

$attachment =
/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;

$from = @xxx.com; 

$to =j...@answeringserv.com ;

$subject=New Applicant;

$headers = From: $from;

$message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;

mail($to,$subject,$message,$headers);

?

 

 

So is it anything obviously wrong with the script I'm missing?

 

Besides something not being configured in php.ini correctly any other ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments. 

 

Common sense is the collection of prejudices acquired by age eighteen. --
Albert Einstein 

Once you can accept the universe as matter expanding into nothing that is
something,wearing stripes with plaid comes easy. -- Albert Einstein

I know a little of everything, but a lot of nothing

 

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Re: [asterisk-users] 2B Channel Transfer on XO-based T1

Jared Smith wrote:
 On Wed, 2009-04-15 at 09:58 -0500, Kevin P. Fleming wrote:
 It's not enabled by default because when it is used the Asterisk server
 loses control of the call and the CDR becomes incomplete. Not everyone
 wants that behavior.
 
 But since many people *would* like that behavior, wouldn't it make more
 sense to enable this via an option in chan_dahdi.conf?  Maybe
 enable2bct=yes?  (It's not like you don't already have to set
 facilityenable=yes and transfer=yes to get it anyway, and I doubt there
 are many people who want facilityenable=yes and transfer=yes but not
 2bct... But for those few, I guess we can add yet another option.)
 
 It seems silly to have to recompile just to get this functionality.
 

It *is* compiled in by default and it actually *is* configurable.  I've 
said this a few times in the archives, but just so that everyone knows, 
in order for it to work, 'transfer=yes' must be set in chan_dahdi.conf 
on each of the channels you would like to enable it on.

To disable it for a channel or group of channels, set 'transfer=no' 
above that group.

Matthew Fredrickson
Digium, Inc.

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Re: [asterisk-users] 2B Channel Transfer on XO-based T1

Don Kelly wrote:
 Someone referred to a facility message when the TBCT call is torn down.
 There are actually two messages--when the PSTN switch takes back the calls
 and completes the transfer, it sends a facility message including a unique
 ID. Then, when one of the parties disconnects, the switch sends another
 facility message with the same unique ID. This would provide information to
 complete the CDR record. Now that there seems to be some interest in TBCT,
 is someone interested in handling these two facility messages to update the
 CDR?

Unfortunately, when I implemented this code I did not add support for 
this feature since it would probably have required some core changes to 
do so.  So right now, we simply ignore that message and go about our 
merry way.

Matthew Fredrickson
Digium, Inc.

 
 --Don
 
 Don Kelly
 
 PCF Corp
 People Come First
 651 842-1000
 888 Don Kell(y)
 651 842-1001 fax
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
 Fleming
 Sent: Wednesday, April 15, 2009 9:58 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 2B Channel Transfer on XO-based T1
 
 Philipp Kempgen wrote:
 
 Could somebody shed some light on why PRI_2BCT is not enabled by
 default? Is it an experimental feature?

 I'd like to compile stuff without patching defines. :-)
 
 It's not enabled by default because when it is used the Asterisk server
 loses control of the call and the CDR becomes incomplete. Not everyone
 wants that behavior.
 


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Re: [asterisk-users] 2BCT last mile... Hopefully

Max Metral wrote:
 Ok, so I’ve made progress on 2BCT (2 B-Channel Transfer).  I’m assuming 
 that the debug info below shows that XO doesn’t have 2BCT enabled on my 
 line, but if anybody can confirm that’ll let me be way more indignant. J

It would appear that the switch doesn't like what you're sending it. 
That means either your switch at the other end is not configured with 
this feature enabled or that your switchtype is set incorrectly for the 
actual switch type that the other end is expecting.

 From the message you are sending to the other side, it would appear 
that you are configured for either 5ESS or national switch type.

Another possiblity (although low in probability) is that it doesn't like 
being sent the transfer message so soon, since it would appear that we 
have not yet received the CONNECT-ACK from the other switch by the time 
we send the transfer request.

You could try inserting a Wait(5) after you Answer() the call in your 
dialplan before Dial()'ing back out to verify that the call is 
completely setup.  Make sure you try explicitly Answer()'ing the call 
first in your dialplan before Wait()'ing or Dial()'ing back out, at 
least until you figure out what the problem is.

Matthew Fredrickson
Digium, Inc.


 
  
 
 -- Native bridging DAHDI/1-1 and DAHDI/3-1
 
   Protocol Discriminator: Q.931 (8)  len=28
 
   Call Ref: len= 2 (reference 801/0x321) (Terminator)
 
   Message type: FACILITY (98)
 
   [1c 15 91 a1 12 02 01 23 06 07 2a 86 48 ce 15 00 08 30 04 02 02 01 93]
 
   Facility (len=23, codeset=0) [ 0x91, 0xA1, 0x12, 0x02, 0x01, '#', 
 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x04, 0x02, 
 0x02, 0x01, 0x93 ]
 
 PROTOCOL 11I
 
 A1 0012 (CONTEXT SPECIFIC [1])
 
   02 0001 23 (INTEGER: 35)
 
   06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08)
 
   30 0004 (SEQUENCE)
 
 02 0002 01 93 (INTEGER: 403)
 
  Protocol Discriminator: Q.931 (8)  len=5
 
  Call Ref: len= 2 (reference 801/0x321) (Originator)
 
  Message type: CONNECT ACKNOWLEDGE (15)
 
 q931.c:3705 q931_receive: call 801 on channel 1 enters state 10 (Active)
 
  Protocol Discriminator: Q.931 (8)  len=16
 
  Call Ref: len= 2 (reference 801/0x321) (Originator)
 
  Message type: FACILITY (98)
 
  [1c 09 91 a3 06 02 01 23 02 01 00]
 
  Facility (len=11, codeset=0) [ 0x91, 0xA3, 0x06, 0x02, 0x01, '#', 
 0x02, 0x01, 0x00 ]
 
 PROTOCOL 11I
 
 A3 0006 (CONTEXT SPECIFIC [3])
 
   02 0001 23 (INTEGER: 35)
 
   02 0001 00 (INTEGER: 0)
 
 -- Processing IE 28 (cs0, Facility)
 
 Handle Q.932 ROSE return error component
 
 Unable to handle return result on switchtype 1!
 
 
 
 
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[asterisk-users] CDR issue

2009-04-23 Thread Gustavo A Gonzalez

Hello! Ive an issue whit CDR using asterisk 1.4.23.1. Ive configured mysql
to store cdr information, but, while I put into cdr_mysql.conf the field
userfield=1 and doing a query I found that this field is empty in the cdr
table. On the other hand I cant find records in the cdr table that show me
calls generated through  AMI using Originate Action, thats calls are not
stored in the CDR, but I dont know why if I calls from pstn are stored
whitout problems. Parameters that I send are:

 

Channe:SIP/$CHANNEL

CallerID:listener

Application:ChanSpy'

Data:SIP/AgentPhone.

Variable:CDR(userfield)=listened

Async=yes

 

Thanks for any idea about that

Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
 mailto:ggonza...@despegar.com ggonza...@despegar.com 

 

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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread Mik Cheez

This is the right suggestion:

Run something like the following:

[h...@mouth tmp]# echo this is a test | newhire.php

If the script runs, check your maillog (/var/log/maillog) to see if 
there's any evidence of what may have happened.

Geraint Lee wrote:
 Check you can run the script from th ecommand line and successfully send 
 email... have you considered using phpagi for your scripts?
 
 2009/4/23 James A. Shigley j...@answeringserv.com 
 mailto:j...@answeringserv.com
 
 I have the below script that doesn’t seem to be working. I don’t
 know if I have something in the script wrong that I am just missing.
 Or if I don’t have the php.ini set correctly for emailing
 
  
 
  
 
 This is the CLI output
 
 -- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1,
 newhire,s,1) in
 
 new stack
 
 -- Goto (newhire,s,1)
 
 -- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack
 
 -- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack
 
 -- Executing [...@newhire:3] Monitor(DAHDI/50-1,
 wav,/var/lib/asterisk/soun
 
 ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack
 
 -- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in
 new stack
 
 -- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php
 
 -- DAHDI/50-1AGI Script newhire.php completed, returning 0
 
 -- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'
 
 -- Hungup 'DAHDI/50-1'
 
  
 
 Here is the script
 
  
 
  
 
 #!/usr/bin/php5
 
 ?php
 
  
 
 // Get AGI vars from *
 
  
 
  $agivars = array();
 
  while (!feof(STDIN)) {
 
  $agivar = trim(fgets(STDIN));
 
  if ($agivar === '') {
 
  break;
 
  }
 
  $agivar = explode(':', $agivar);
 
  $agivars[$agivar[0]] = trim($agivar[1]);
 
  }
 
  extract($agivars);
 
  
 
 // Variable Declarations
 
  
 
 $agi_uniqueid;
 
 $agi_callerid;
 
 $agi_calleridname;
 
 $agi_extension;
 
 $agi_uniqueid;
 
 $UNIQUEID = $agi_uniqueid;
 
 $CALLERID = $agi_callerid;
 
 $EXTEN = $agi_extension;
 
 $attachment =
 /var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav;
 
 $from = �...@xxx.com http://xxx.com;
 
 $to =j...@answeringserv.com mailto:j...@answeringserv.com ;
 
 $subject=New Applicant;
 
 $headers = From: $from;
 
 $message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;
 
 mail($to,$subject,$message,$headers);
 
 ?
 
  
 
  
 
 So is it anything obviously wrong with the script I’m missing?
 
  
 
 Besides something not being configured in php.ini correctly any
 other ideas?
 
  
 
 James Shigley
 
 *Monroe Telephone Answering Service*
 
 409-981-9213**
 
 Infinity 5.5,UC 4.02.3803, Blink 3.0.104
 
 Ecreator:2.21, eResponse 1.1.7
 
 Webportal,WebApps,
 
  
 
 CONFIDENTIALITY NOTICE: This email, including any attachments,
 contains information which may be confidential or privileged. The
 information is intended to be for the use of the individual or
 entity named above. If you are not the intended recipient, be aware
 that any disclosure, copying, distribution or use of the contents of
 this information is prohibited. If you have received this email in
 error, please notify the sender immediately by reply to sender
 only message and destroy all electronic and hard copies of the
 communication, including attachments.
 
  
 
 Common sense is the collection of prejudices acquired by age
 eighteen. -- Albert Einstein
 
 Once you can accept the universe as matter expanding into nothing
 that is something,wearing stripes with plaid comes easy. -- Albert
 Einstein
 
 I know a little of everything, but a lot of nothing
 
  
 
 
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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread Darrick Hartman

Rob Hillis wrote:
 Kurian Thayil wrote:
 On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
   
 Daily Asterisk restart
 
 Do you think its mandatory in production env?
 
 Daily?  No.  However, after implementing a weekly restart of Asterisk,
 I've found the instance of lockups and CPU utilisation spikes have
 decreased significantly.

Unless you're using some unstable modules, there really should be no 
need to restart Asterisk.  Is there a certain activity that is causing 
these lockups?  I have low power systems which haven't had Asterisk 
restarted in months many times.  Granted, these are mostly low call 
volume systems, but unless there is a memory leak, you should not needed 
to restart the Asterisk process.  (my guess is one of the modules you 
are using has some sort of problem).

Darrick


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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread Jeff LaCoursiere


On Thu, 23 Apr 2009, Darrick Hartman wrote:

 Rob Hillis wrote:
 Kurian Thayil wrote:
 On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:

 Daily Asterisk restart

 Do you think its mandatory in production env?

 Daily?  No.  However, after implementing a weekly restart of Asterisk,
 I've found the instance of lockups and CPU utilisation spikes have
 decreased significantly.

 Unless you're using some unstable modules, there really should be no
 need to restart Asterisk.  Is there a certain activity that is causing
 these lockups?  I have low power systems which haven't had Asterisk
 restarted in months many times.  Granted, these are mostly low call
 volume systems, but unless there is a memory leak, you should not needed
 to restart the Asterisk process.  (my guess is one of the modules you
 are using has some sort of problem).


I had a 1.2 instance running over a year and a half with a ton of live 
calls to a custom AGI.  If you are having to restart asterisk something 
must be wrong.

j


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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread James A. Shigley

Actually I feel like an idiot. I had forgotten to put asterisk as an
allowed sender in the server that those emails are going out of.
(different from what * normally uses to email us)

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments. 

 

Common sense is the collection of prejudices acquired by age eighteen.
-- Albert Einstein 

Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy. -- Albert Einstein

I know a little of everything, but a lot of nothing

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy
Gbaguidi
Sent: Thursday, April 23, 2009 11:57 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] AGI PHP script

 

First run 

/var/lib/asterisk/agi-bin/newhire.php

 

From linux command line to see if you don't have any error and that your
AGI is executable.

 

Then run 'agi debug' from the asterisk cli, place a call and see what
was send and receive from your agi

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: April-23-09 12:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI PHP script

 

I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing

 

 

This is the CLI output

-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1)
in

new stack

-- Goto (newhire,s,1)

-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack

-- Executing [...@newhire:3] Monitor(DAHDI/50-1,
wav,/var/lib/asterisk/soun

ds/NewHire/Newhire-1240503071.15148-4099819213-s,o) in new stack

-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new
stack

-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php

-- DAHDI/50-1AGI Script newhire.php completed, returning 0

-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'

-- Hungup 'DAHDI/50-1'

 

Here is the script

 

 

#!/usr/bin/php5 

?php

 

// Get AGI vars from *

 

 $agivars = array();

 while (!feof(STDIN)) {

 $agivar = trim(fgets(STDIN));

 if ($agivar === '') {

 break;

 }

 $agivar = explode(':', $agivar);

 $agivars[$agivar[0]] = trim($agivar[1]);

 }

 extract($agivars);

 

// Variable Declarations

 

$agi_uniqueid;

$agi_callerid;

$agi_calleridname;

$agi_extension;

$agi_uniqueid;

$UNIQUEID = $agi_uniqueid;

$CALLERID = $agi_callerid;

$EXTEN = $agi_extension;

$attachment =
/var/lib/asterisk/sounds/NewHire/Newhire-$UNIQUEID-$CALLERID-$EXTEN.wav
;

$from = @xxx.com; 

$to =j...@answeringserv.com ;

$subject=New Applicant;

$headers = From: $from;

$message =$UNIQUEID , $CALLERID , $EXTEN , $attachment;

mail($to,$subject,$message,$headers);

?

 

 

So is it anything obviously wrong with the script I'm missing?

 

Besides something not being configured in php.ini correctly any other
ideas?

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments. 

 

Common sense is the collection of prejudices acquired by age eighteen.
-- Albert Einstein 

Once you can accept the universe as matter expanding into nothing that
is something,wearing stripes with plaid comes easy. -- Albert Einstein

I know a little of everything, but a lot of nothing

 

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[asterisk-users] Libpri-1.4.10 Released

2009-04-23 Thread Asterisk Development Team

The Asterisk.org development team has announced the release of Libpri 
version 1.4.10. This release contains bug fixes related to handling of 
BRI PTMP links and how messages are handled during link state 
transients, as well as other fixes.  Please see the Changelog for more 
details.

The release is available as a tarball as well as a patch against the 
previous release. It is available for download from downloads.digium.com.

Thank you for your support!

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[asterisk-users] Asterisk on Mac OS X

Hello list.
I posted this over on the Biz section but some of the members thought  
I might find more people running Asterisk on the Mac over here.

Here's my question:


I have looked at PHLink and PhoneValet and neither seem to be able to
do what I need, so I am looking at Asterisk.

What I want to do is allow callers to call a our phone line and
unsubscribe their phone number from our call center list.  So,
basically, when they call in, they would be greeted with a message
something like: please enter your 10 digit phone number followed by
the pound sign.  They would then have the number read back to them to
confirm it or reenter it.  Once confirmed, it would write the phone
number to a text file for importing into MySQL or FileMaker.

Is what I am trying to accomplish within the realm of what Asterisk
can do on the Mac platform... or any platform... and if so, how
difficult of an install is it?  I have read varying accounts from it
being a breeze to being frustrating.

I have already been told I can do this via both caller id and via  
number entry by touch tone, my question is, are there currently any  
users who are doing the above on a Mac or should I only consider Linux?

Also is there a GUI frontend to Asterisk for the Mac version?

TIA
--Rick





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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread hh174





Not true for me, a restart, if not mandatory, is a good idea, let's say
weekly or more depending on the usage.
My side is asterisk servers as gateways with thousands of calls a day
(at least 10.000 minutes a day).

I have asterisk(s) running for a long time( 4 years and more), with ss7
links and without, just as a b2bua.
I just see that Asterisk sometime doesn't close the connections
correctly (or the other party).
As an example, we have some servers redirecting to a specific provider,
after a few days even with no actives calls, some connections stay
open, probably due to the provider (no control on that)...
Then restarting asterisk is, for me, the only solution.

Another case is the ss7 links.
After a few days, some channels dies, the only solution is again to
restart asterisk and Dahdi(zaptel).

I don't say it's due to Asterisk, I just say it's not a bad idea to
have a cron with a 'restart when convenient' to have servers always up.

Regards,

Olivier


Jeff LaCoursiere a crit:

  On Thu, 23 Apr 2009, Darrick Hartman wrote:

  
  
Rob Hillis wrote:


  Kurian Thayil wrote:
  
  
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:



  Daily Asterisk restart

  

Do you think its mandatory in production env?

  
  Daily?  No.  However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
  

Unless you're using some unstable modules, there really should be no
need to restart Asterisk.  Is there a certain activity that is causing
these lockups?  I have low power systems which haven't had Asterisk
restarted in months many times.  Granted, these are mostly low call
volume systems, but unless there is a memory leak, you should not needed
to restart the Asterisk process.  (my guess is one of the modules you
are using has some sort of problem).


  
  
I had a 1.2 instance running over a year and a half with a ton of live 
calls to a custom AGI.  If you are having to restart asterisk something 
must be wrong.

j


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Re: [asterisk-users] AGI PHP script

2009-04-23 Thread Steve Edwards


Un-top-posting and de-crufting, with some comments cast inline...

2009/4/23 James A. Shigley j...@answeringserv.com 
mailto:j...@answeringserv.com


I have the below script that doesn?t seem to be working. I don?t
know if I have something in the script wrong that I am just missing.
Or if I don?t have the php.ini set correctly for emailing

This is the CLI output
-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1,
newhire,s,1) in new stack
-- Goto (newhire,s,1)
-- Executing [...@newhire:1] Ringing(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:2] Answer(DAHDI/50-1, ) in new stack
-- Executing [...@newhire:3] 
Monitor(DAHDI/50-1,wav,/var/lib/asterisk/sounds/NewHire/Newhire-1240503071.15148-4099819213-s,o)
 in new stack
-- Executing [...@newhire:4] AGI(DAHDI/50-1, newhire.php) in new 
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/newhire.php
-- DAHDI/50-1AGI Script newhire.php completed, returning 0
-- Auto fallthrough, channel 'DAHDI/50-1' status is 'UNKNOWN'
-- Hungup 'DAHDI/50-1'


Auto fallthrough means you fell off the end of your dialplan. I prefer 
to explicitly code a hangup so the next guy knows what was supposed to 
happen rather that what happened to happen :)


When you post CLI output, the relevant snippet of extensions.conf can be 
helpful.



Here is the script

#!/usr/bin/php5
?php

// Get AGI vars from *
 $agivars = array();
 while (!feof(STDIN)) {
 $agivar = trim(fgets(STDIN));
 if ($agivar === '') {
 break;
 }
 $agivar = explode(':', $agivar);
 $agivars[$agivar[0]] = trim($agivar[1]);
 }
 extract($agivars);


While your script is rather trivial, most coders use an established AGI 
library.


Besides something not being configured in php.ini correctly any other 
ideas?


Yes. Trim your signature splooge.

On Thu, 23 Apr 2009, Mik Cheez wrote:


Run something like the following:

[h...@mouth tmp]# echo this is a test | newhire.php

If the script runs, check your maillog (/var/log/maillog) to see if 
there's any evidence of what may have happened.


You can feed the entire AGI environment through STDIN. Here's a sample 
script:


# the standard AGI environment variables
echo agi_accountcode: 
echo agi_callerid: 1234567890
echo agi_calleridname: agi-environment.sh
echo agi_callingani2: 0
echo agi_callingpres: 0
echo agi_callingtns: 0
echo agi_callington: 0
echo agi_channel: SIP/1234567890
echo agi_context: newhire
echo agi_dnid: *
echo agi_enhanced: 0.0
echo agi_extension: 4099819213
echo agi_language: en
echo agi_priority: 1
echo agi_rdnis: unknown
echo agi_request: newhire.php
echo agi_type: SIP
echo agi_uniqueid: 1240503071.15148
echo 

Save it as agi-environment.sh and use it like:

sh agi-environment.sh | newhire.php

If you suspect an environment variable problem you could use:

sh agi-environment.sh | env --ignore-environment newhire.php

Your script doesn't do anything else with the AGI environment, but you can 
extend my script to include the proper response to most AGI requests. It's 
a very powerful debugging technique.


I write my AGIs in C. It totally rocks to edit your source code in emacs 
with gdb running in another buffer reading the AGI environment and request 
responses from a text file while you step through your code line by line.


I'm not a serious PHP hacker, but if you have a PHP IDE maybe the same 
technique would work.


Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help

2009-04-23 Thread Jimmy Ezell

Dan thank you, yes that seems to help.  It looks like the bridging is happening 
now and I see the light come on in the second FXO port, but then I get a busy 
signal after that and the call still does not complete.  If I set the second 
line as priority 1 it completes the first call on that line and second call 
gets the busy on the first line.  I even tried moving the lines to a different 
FXO card and the result is the same.

Here is my current config for the cisco dial-peers:


dial-peer voice 2212 pots
 preference 2
 destination-pattern .T
 port 2/0
 forward-digits all
!
dial-peer voice 2211 pots
 preference 1
 destination-pattern .T
 port 0/0
 forward-digits all


Thanks again Dan,  I think I am much closer now.
Jimmy




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Dan Austin
Sent: Thursday, April 23, 2009 09:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help


Jimmy wrote:

Second Call out the asterisk console looks like 
this-:
-- Executing [92952...@internal:1] Dial(SIP/222-09ab3588, 
SIP/Cisco1760/2952210) in new stack
-- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call 
from '222' to extension '2952210' rejected because extension not found.
-- Got SIP response 486 Busy here back from 172.17.2.1
-- SIP/Cisco1760-09ab7cf8 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
-- Executing [92952...@internal:2] Congestion(SIP/222-09ab3588, ) in 
new stack
  == Spawn extension (internal, 92952210, 2) exited non-zero on 
'SIP/222-09ab3588'
localhost*CLI


--sip.conf -
[general]
bindaddr=0.0.0.0

[Cisco1760]
context=incoming_calls
type=friend
host=172.17.2.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very


--extensions.conf
[globals]
OUTBOUNDTRUNK=SIP/Cisco1760


[outbound-local]
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,n,Congestion()
exten = _9NXX,n,Hangup()

---Cisco 1760 config --
dial-peer voice 100 pots  (This line that is set to preference 2 does not work)
 huntstop
 preference 2
 destination-pattern .T
 port 0/0
 forward-digits all
!
dial-peer voice 2212 pots(This line that is set to Preference 1 is the one 
that works)
 huntstop
 preference 1
 destination-pattern .T
 port 0/1
 forward-digits all



You do not want to use huntstop on the dialpeers in this situation.
The huntstop option tells the call routing function in the router to
stop search for a call route if it encounters a failure.

Call number 2 hits dialpeer 1, finds it busy and the huntstop causes
the processing to stop.

Dan

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Re: [asterisk-users] CDR issue

2009-04-23 Thread Miguel Molina


Gustavo A Gonzalez escribió:


Hello! I've an issue whit CDR using asterisk 1.4.23.1. I've configured 
mysql to store cdr information, but, while I put into cdr_mysql.conf 
the field 'userfield=1' and doing a query I found that this field is 
empty in the cdr table. On the other hand I can't find records in the 
cdr table that show me calls generated through  AMI using Originate 
Action, that's calls are not stored in the CDR, but I don't know why 
if I calls from pstn are stored whitout problems. Parameters that I 
send are:


 


Channe:SIP/$CHANNEL

CallerID:listener

Application:ChanSpy'

Data:SIP/AgentPhone.

Variable:CDR(userfield)=listened

Async=yes

 

Well, there's two things to look around here. First, the variable en the 
AMI Originate action lets you set channel variables on to the channel of 
the originated call, not onto the spied (and already established) call. 
Second, if you can't see the originated calls on the CDR, try setting 
the option unanswered=yes on cdr.conf and issue a cdr reload command 
on the CLI. You can check with cdr status that this option is set, 
that tells asterisk to maintain a CDR for *every* call attempt that goes 
through it. Then you can check if the originated calls show up on your 
CDR database.


Cheers,


Thanks for any idea about that

*Gustavo A. González*
Dto. de Infraestructura
Despegar.com, Inc.
ggonza...@despegar.com mailto:ggonza...@despegar.com

 




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--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
PBX: (+57 1)6500800 ext. 1201
Fax: (+57 1)6500816
Móvil: (+57)3138873587 

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Re: [asterisk-users] Asterisk on Mac OS X

2009-04-23 Thread Steve Edwards

On Thu, 23 Apr 2009, Rick Dwyer wrote:

 What I want to do is allow callers to call a our phone line and 
 unsubscribe their phone number from our call center list.  So, 
 basically, when they call in, they would be greeted with a message 
 something like: please enter your 10 digit phone number followed by the 
 pound sign.  They would then have the number read back to them to 
 confirm it or reenter it.  Once confirmed, it would write the phone 
 number to a text file for importing into MySQL or FileMaker.

Piece of cake -- use Read() and SayDigits().

Any reason you don't want to write the number directly to the database? 
Any interest in looking up the number in the database so they know if they 
are entering a subscribed number? How will you keep a disgruntled customer 
or employee from un-subscribing other customers?

 Is what I am trying to accomplish within the realm of what Asterisk can 
 do on the Mac platform... or any platform... and if so, how difficult of 
 an install is it?  I have read varying accounts from it being a breeze 
 to being frustrating.

It depends on the skills of the installer. I prefer to install from source 
but a lot of people depend on RPM or DEB packages or just use a boot  
nuke approach with PIAF or something similar.

 I have already been told I can do this via both caller id and via number 
 entry by touch tone, my question is, are there currently any users who 
 are doing the above on a Mac or should I only consider Linux?

Asterisk is developed on Linux. Most users run on Linux. You will have 
fewer problems on Linux. Can you run Linux on your Mac? (I mean booting 
Linux, not VMWare or Parallels).

Thanks in advance,

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

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Re: [asterisk-users] Asterisk on Mac OS X

2009-04-23 Thread Niles Ingalls


On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote:

 Hello list.
 I posted this over on the Biz section but some of the members thought
 I might find more people running Asterisk on the Mac over here.

 Here's my question:


 I have looked at PHLink and PhoneValet and neither seem to be able to
 do what I need, so I am looking at Asterisk.

 What I want to do is allow callers to call a our phone line and
 unsubscribe their phone number from our call center list.  So,
 basically, when they call in, they would be greeted with a message
 something like: please enter your 10 digit phone number followed by
 the pound sign.  They would then have the number read back to them to
 confirm it or reenter it.  Once confirmed, it would write the phone
 number to a text file for importing into MySQL or FileMaker.

 Is what I am trying to accomplish within the realm of what Asterisk
 can do on the Mac platform... or any platform... and if so, how
 difficult of an install is it?  I have read varying accounts from it
 being a breeze to being frustrating.

The main distinction between running Asterisk on Linux as opposed to  
OSX, is that you'll
have access to hardware device drivers.  If you're going to be using a  
SIP/IAX Trunk, then
you'll be just fine on OSX.  What your attempting to do falls closer  
to the category of breeze.
You can install the asterisk-addon package to handle your SQL queries  
from within the dialplan,
or you can use AGI to have a perl or php script do that work.
Niles

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Re: [asterisk-users] Asterisk on Mac OS X


On Apr 23, 2009, at 3:34 PM, Steve Edwards wrote:

 On Thu, 23 Apr 2009, Rick Dwyer wrote:

 What I want to do is allow callers to call a our phone line and
 unsubscribe their phone number from our call center list.  So,
 basically, when they call in, they would be greeted with a message
 something like: please enter your 10 digit phone number followed  
 by the
 pound sign.  They would then have the number read back to them to
 confirm it or reenter it.  Once confirmed, it would write the phone
 number to a text file for importing into MySQL or FileMaker.

 Piece of cake -- use Read() and SayDigits().

Excellent.



 Any reason you don't want to write the number directly to the  
 database?

No, this is even better... didn't know it could talk directly to  
MySQL.  Thanks.


 Any interest in looking up the number in the database so they know  
 if they
 are entering a subscribed number? How will you keep a disgruntled  
 customer
 or employee from un-subscribing other customers?

Good point... will have to give this more thought.



 Is what I am trying to accomplish within the realm of what Asterisk  
 can
 do on the Mac platform... or any platform... and if so, how  
 difficult of
 an install is it?  I have read varying accounts from it being a  
 breeze
 to being frustrating.

 It depends on the skills of the installer. I prefer to install from  
 source
 but a lot of people depend on RPM or DEB packages or just use a  
 boot 
 nuke approach with PIAF or something similar.

 I have already been told I can do this via both caller id and via  
 number
 entry by touch tone, my question is, are there currently any users  
 who
 are doing the above on a Mac or should I only consider Linux?

 Asterisk is developed on Linux. Most users run on Linux. You will have
 fewer problems on Linux. Can you run Linux on your Mac? (I mean  
 booting
 Linux, not VMWare or Parallels).

I don't know a thing about Linux and even on the Mac, my command line  
skills are basic.  So I would really be looking for a GUI to configure  
it, regardless of platform.  I assume this is available on Linux?

I am considering laying out our needs and having someone configure it  
for us... can anyone recommend a reliable source to:
A. install Linux on a hard drive I supply
B. configure asterisk per our requirements
C. do it as inexpensively as possible : )

Thanks,
--Rick














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

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Re: [asterisk-users] Asterisk on Mac OS X


On Apr 23, 2009, at 3:40 PM, Niles Ingalls wrote:


 On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote:

 Hello list.
 I posted this over on the Biz section but some of the members thought
 I might find more people running Asterisk on the Mac over here.

 Here's my question:


 I have looked at PHLink and PhoneValet and neither seem to be able to
 do what I need, so I am looking at Asterisk.

 What I want to do is allow callers to call a our phone line and
 unsubscribe their phone number from our call center list.  So,
 basically, when they call in, they would be greeted with a message
 something like: please enter your 10 digit phone number followed by
 the pound sign.  They would then have the number read back to them  
 to
 confirm it or reenter it.  Once confirmed, it would write the phone
 number to a text file for importing into MySQL or FileMaker.

 Is what I am trying to accomplish within the realm of what Asterisk
 can do on the Mac platform... or any platform... and if so, how
 difficult of an install is it?  I have read varying accounts from it
 being a breeze to being frustrating.

 The main distinction between running Asterisk on Linux as opposed to
 OSX, is that you'll
 have access to hardware device drivers.  If you're going to be using a
 SIP/IAX Trunk, then
 you'll be just fine on OSX.

I have a bunch of Handytone-488 boxes from Grandstream... are these  
the hardware devices that will allow me to run multiple lines on a Mac?




  What your attempting to do falls closer
 to the category of breeze.

Good to hear.


 You can install the asterisk-addon package to handle your SQL queries

Good.

Finally, the last thing that will probably determine Mac or Linux does  
the Mac version have a software GUI to adminster and configure  
Asterisk... or is this a Linux only item?

Thanks,
--Rick






 Niles

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Re: [asterisk-users] Asterisk on Mac OS X

2009-04-23 Thread Danny Nicholas

My .02 - you should verify the number and ask for a (numeric) password.
This will save a good deal of grief.



 Any interest in looking up the number in the database so they know  
 if they
 are entering a subscribed number? How will you keep a disgruntled  
 customer
 or employee from un-subscribing other customers?

Good point... will have to give this more thought.



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Re: [asterisk-users] Asterisk on Mac OS X

2009-04-23 Thread Eric Fort

How about posting a list of requirements on the biz list and soliciting
bids?

Eric Fort
FortConsulting

On Thu, Apr 23, 2009 at 12:47 PM, Rick Dwyer rdw...@quick-link.com wrote:



 I don't know a thing about Linux and even on the Mac, my command line
 skills are basic.  So I would really be looking for a GUI to configure
 it, regardless of platform.  I assume this is available on Linux?

 I am considering laying out our needs and having someone configure it
 for us... can anyone recommend a reliable source to:
 A. install Linux on a hard drive I supply
 B. configure asterisk per our requirements
 C. do it as inexpensively as possible : )

 Thanks,
 --Rick












 
 
  Thanks in advance,
  
  Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867
  PST
  Newline Fax:
  +1-760-731-3000
 
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[asterisk-users] Convert file in GSM codec to G729 codec

2009-04-23 Thread Shaun Wingrin

Hi,

I've tried the link
http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error 
at the moment.

Any other ideas most welcome.

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Re: [asterisk-users] Asterisk on Mac OS X


On Apr 23, 2009, at 3:58 PM, Eric Fort wrote:

 How about posting a list of requirements on the biz list and  
 soliciting bids?

Good advice... I will probably do so tomorrow after I talk to  
Ovolab... they have a product for OS X called Phlink they say can do  
what I need.

Thanks,
--Rick





 Eric Fort
 FortConsulting

 On Thu, Apr 23, 2009 at 12:47 PM, Rick Dwyer rdw...@quick-link.com  
 wrote:


 I don't know a thing about Linux and even on the Mac, my command line
 skills are basic.  So I would really be looking for a GUI to configure
 it, regardless of platform.  I assume this is available on Linux?

 I am considering laying out our needs and having someone configure it
 for us... can anyone recommend a reliable source to:
 A. install Linux on a hard drive I supply
 B. configure asterisk per our requirements
 C. do it as inexpensively as possible : )

 Thanks,
 --Rick












 
 
  Thanks in advance,
   
 
  Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867
  PST
  Newline Fax:
  +1-760-731-3000
 
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[asterisk-users] Record in mp3

2009-04-23 Thread Jose Enes Mateus

Somebody knows if I can save files in mp3 with the Record command on Asterisk?
I try to recompile sox to suport mp3 but Asterisk return the folowing message 
when I use the Record command:

- Executing [...@liberado15:15] Record(SIP/1201-083453c8, 
/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack
    -- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR')
[Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write MP3 
only read them.
[Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite 
format mp3
[Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to rewrite 
/var/spool/asterisk/alarme/alarme-1201-200905121212.mp3
[Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not create 
file /var/spool/asterisk/alarme/alarme-1201-200905121212

I'am doing something wrong?

Thanks



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[asterisk-users] dial and transfer while ringing

Hi,

My extensions.ael file includes :

context mylocal {
7530 = {
Dial(SIP/7530,,${OPTION});
NoOp(Here1);
};
7531 = {
Dial(SIP/7531);
NoOp(Here2);
};
};

If extension 7530 receives a call and transfer it while ringing to extension
7531 (a 302 Moved temporarily message is sent by callee),
then which value shall I put in OPTION to have NoOp(Here1) executed ?

In various tries, I could see NoOp(Here1) execution postponed right after
NoOp(Here2) or not happening at all.


Regards
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Re: [asterisk-users] Record in mp3

2009-04-23 Thread Danny Nicholas

The way I read to do this is to use sox to create a wav file, then use lame
to convert the wav to mp3.  I did this for some MOH files.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes
Mateus
Sent: Thursday, April 23, 2009 3:28 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Record in mp3

 


Somebody knows if I can save files in mp3 with the Record command on
Asterisk?
I try to recompile sox to suport mp3 but Asterisk return the folowing
message when I use the Record command:

- Executing [...@liberado15:15] Record(SIP/1201-083453c8,
/var/spool/asterisk/alarme/alarme-1201-200905121212:mp3) in new stack
-- SIP/1201-083453c8 Playing 'beep' (language 'pt_BR')
[Apr 23 17:22:36] ERROR[4494]: format_mp3.c:283 mp3_rewrite: I Can't write
MP3 only read them.
[Apr 23 17:22:36] WARNING[4494]: file.c:378 fn_wrapper: Unable to rewrite
format mp3
[Apr 23 17:22:36] WARNING[4494]: file.c:1092 ast_writefile: Unable to
rewrite /var/spool/asterisk/alarme/alarme-1201-200905121212.mp3
[Apr 23 17:22:36] WARNING[4494]: app_record.c:272 record_exec: Could not
create file /var/spool/asterisk/alarme/alarme-1201-200905121212

I'am doing something wrong?

Thanks

 

  _  

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Re: [asterisk-users] Asterisk on Mac OS X

2009-04-23 Thread Steve Edwards

On Thu, 23 Apr 2009, Rick Dwyer wrote:

 I don't know a thing about Linux and even on the Mac, my command line
 skills are basic.  So I would really be looking for a GUI to configure
 it, regardless of platform.  I assume this is available on Linux?

There are web based interfaces available that will work with either OS.

The pros of the GUI approach is that the learning curve can be flatter 
and that they impose some structure so you just fill in the blanks.

The cons are that they insulate you from learning the full capabilities 
of Asterisk and if your application doesn't fit in one in the blanks 
then either you have to jump through a bunch of hoops or it can't be done.

What you've described so far sounds like 1/2 page of dialplan with maybe 
an AGI thrown in to keep things simple and maintainable.

Asterisk takes very little in the way of hardware resources. I'd vote for 
a sub $300 x86 box running Linux. I'm a big fan of do 1 thing, do it 
well, and move on. If this application was running on a Mac, I'm sure 
somebody would want to use that cute box for all kinds of fun.

Thanks in advance,

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

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Re: [asterisk-users] Convert file in GSM codec to G729 codec

2009-04-23 Thread Tzafrir Cohen

On Thu, Apr 23, 2009 at 10:08:39PM +0200, Shaun Wingrin wrote:
 Hi,
 
 I've tried the link
 http://www.asteriskguru.com/tools/audio_conversion.php but it returns an 
 error at the moment.

sweetmorn*CLI help file convert
Usage: file convert file_in file_out
   Convert from file_in to file_out. If an absolute path
   is not given, the default Asterisk sounds directory
   will be used.

   Example:
   file convert tt-weasels.gsm tt-weasels.ulaw

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

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[asterisk-users] want to set up text based adventure for asterisk

2009-04-23 Thread Eric Fort

Anyone know where I could find a good beginning for using asterisk and the
text based game adventure together such that I could play from the nearest
phone?

Thanks,

Eric
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Re: [asterisk-users] CDR issue

2009-04-23 Thread Gustavo A Gonzalez

Thanks!Ive solve the issue setting: unanswered=yes on cdr.conf .

 

Cheers!

 

Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
ggonza...@despegar.com 

 

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Re: [asterisk-users] want to set up text based adventure for asterisk

2009-04-23 Thread Danny Nicholas

You would just use festival/Cepstral combined with Read and an AGI to take
options and speak the result back.  As long as you had a reasonably finite
number of possible outcomes, you could even do this just from a dialplan
without AGI.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort
Sent: Thursday, April 23, 2009 3:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] want to set up text based adventure for asterisk

 

Anyone know where I could find a good beginning for using asterisk and the
text based game adventure together such that I could play from the nearest
phone?

Thanks,

Eric

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[asterisk-users] BLINDTRANSFER and SIP hardphones

Hi,

When a SIP hardphone is transfering a call while ringing (caller and callee
don't speak to each other) using phone's Transfer key, it seems
BLINDTRANSFER remains empty.
Though I can see a 302 MOVED TEMPORARILY message coming in.

Is there a work around or something obvious I'm missing (it's the first time
I'm playing with Dialplan transfert features.

context mylocal {
7530 = {
NoOp(Here1 ${BLINDTRANSFER});
Dial(SIP/7530,);
NoOp(Here2 ${BLINDTRANSFER});
};
7531 = {
NoOp(Here3 ${BLINDTRANSFER});
Dial(SIP/7531);
NoOp(Here4 ${BLINDTRANSFER});
};
};


Regards
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Re: [asterisk-users] want to set up text based adventure for asterisk

2009-04-23 Thread Tim Nelson

A good place to start is here: 

http://www.venturevoip.com/news.php?rssid=1513 

FreePBX includes a module called 'Zoip' which allows you to play Zork via a 
Text-to-speech engine. 

Why on Earth someone would want to do so is beyond me but hey... why not. :-) 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 

- Eric Fort eric.f...@gmail.com wrote: 
 Anyone know where I could find a good beginning for using asterisk and the 
 text based game adventure together such that I could play from the nearest 
 phone? 
 
 Thanks, 
 
 Eric 
 
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Re: [asterisk-users] want to set up text based adventure for asterisk

2009-04-23 Thread Jon Pounder

Eric Fort wrote:
 Anyone know where I could find a good beginning for using asterisk and 
 the text based game adventure together such that I could play from 
 the nearest phone?

google on collossal cave.
honestly its the absolute worst unreadable mess of code ever conceived 
by man or beast.

that said I made a web version of it that actually works
here is where you start - hack the only semi unreadable portion for 
save and load games so it does not cost you points to save and load.
then make a script that does this :
start game, load game, inject next command to game, trap output of that 
move, save game again

every time the user does a command, run the script and then show/speak 
them the output.

yes its a kludge of the Nth order but at the end of the day it works, 
and you didn't even have to understand the garbage code that drives the 
thing.


let me know how it goes.




 Thanks,

 Eric
 

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Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help

2009-04-23 Thread Jonathan Thurman

We are in a similar situation to you as far as moving from Cisco to
Asterisk.  I have not got to the point of integrating Asterisk
directly to our PSTN gateways yet, but this might help.

On our H.323 gateways we use trunk groups for outbound call hunting.
You can create a single trunk group for outbound calls and add any
voice port to it that you want.  On the voice port you tell it what
trunk group it belongs to and the priority (1-64 I believe).  The
available port with the lowest priority wins.  This configuration is
from a 2801 using H.323 to CallManager 6.1:

Example:
-

trunk group Outbound
 description - Outbound calling hunt group
 hunt-scheme sequential
!

voice-port 0/0
 trunk-group Outbound 1
!
voice-port 2/0
 trunk-group Outbound 2
!

dial-peer voice 2000 pots
 trunkgroup Outbound
 description Outbound call hunting
 destination-pattern .T
!

--

-Jonathan


On Thu, Apr 23, 2009 at 12:18 PM, Jimmy Ezell jez...@hmhca.com wrote:
 Dan thank you, yes that seems to help.  It looks like the bridging is 
 happening now and I see the light come on in the second FXO port, but then I 
 get a busy signal after that and the call still does not complete.  If I set 
 the second line as priority 1 it completes the first call on that line and 
 second call gets the busy on the first line.  I even tried moving the lines 
 to a different FXO card and the result is the same.

 Here is my current config for the cisco dial-peers:


 dial-peer voice 2212 pots
  preference 2
  destination-pattern .T
  port 2/0
  forward-digits all
 !
 dial-peer voice 2211 pots
  preference 1
  destination-pattern .T
  port 0/0
  forward-digits all


 Thanks again Dan,  I think I am much closer now.
 Jimmy




 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of Dan Austin
 Sent: Thursday, April 23, 2009 09:39 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help


 Jimmy wrote:

 Second Call out the asterisk console looks like 
 this-:
    -- Executing [92952...@internal:1] Dial(SIP/222-09ab3588, 
 SIP/Cisco1760/2952210) in new stack
    -- Called Cisco1760/2952210
 [Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call 
 from '222' to extension '2952210' rejected because extension not found.
    -- Got SIP response 486 Busy here back from 172.17.2.1
    -- SIP/Cisco1760-09ab7cf8 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [92952...@internal:2] Congestion(SIP/222-09ab3588, ) in 
 new stack
  == Spawn extension (internal, 92952210, 2) exited non-zero on 
 'SIP/222-09ab3588'
 localhost*CLI


 --sip.conf -
 [general]
 bindaddr=0.0.0.0

 [Cisco1760]
 context=incoming_calls
 type=friend
 host=172.17.2.1
 dtmfmode=rfc2833
 disallow=all
 allow=ulaw
 insecure=very


 --extensions.conf
 [globals]
 OUTBOUNDTRUNK=SIP/Cisco1760


 [outbound-local]
 exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
 exten = _9NXX,n,Congestion()
 exten = _9NXX,n,Hangup()

 ---Cisco 1760 config --
 dial-peer voice 100 pots  (This line that is set to preference 2 does not 
 work)
  huntstop
  preference 2
  destination-pattern .T
  port 0/0
  forward-digits all
 !
 dial-peer voice 2212 pots    (This line that is set to Preference 1 is the 
 one that works)
  huntstop
  preference 1
  destination-pattern .T
  port 0/1
  forward-digits all


 
 You do not want to use huntstop on the dialpeers in this situation.
 The huntstop option tells the call routing function in the router to
 stop search for a call route if it encounters a failure.

 Call number 2 hits dialpeer 1, finds it busy and the huntstop causes
 the processing to stop.

 Dan

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Re: [asterisk-users] want to set up text based adventure for asterisk

2009-04-23 Thread Gordon Henderson

On Thu, 23 Apr 2009, Eric Fort wrote:

 Anyone know where I could find a good beginning for using asterisk and the
 text based game adventure together such that I could play from the nearest
 phone?

Heh... I wrote a MUD once (still online, but..)

Key 1 to get the long sword.
Key 2 to go north.
Key 4 to go west.

You have been eaten by a grue...

Hm. Maybe a bit slow!

Gordon

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[asterisk-users] about Asterisk and AudioCodes FXO/H323

2009-04-23 Thread Huseyin Sahbal

hi all;

I  need to find out about  how to configure Asterisk (h323.conf)  and 
Audiocodes FXO/H323 voip-gateway.Audiocodes side too complex for that.

thank you so much

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Re: [asterisk-users] Asterisk Capacity

On 24/04/2009 1:11 a.m., Geraint Lee wrote:
 Thanks for that, it's pretty much confirming what i first anticipated...
 my intentions are as follows:

 agents register with opensips, opensips clusters a set of call recording
 servers which then connect to our border servers which will save cdr and
 choose the sip/iax provider to send the call to.

You may find it better to have call recording on a separate machine 
which is connected to a mirrored port on the switch and sniffs the 
traffic via something like Orecx:

http://www.orecx.com/

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] AMD Not Working

On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
 Hi All,

 I am trying to use the AMD (Answering Machine Detect).
 But it is not sending the AMD_Status as either
 the Human or Machine, it hangs up in middle.

I'd say that the remote end of the call is hanging up - do a SIP debug 
so you can see what happens - the best way to test things like this is 
by calling your own number - that way you can guarantee it doesn't hang 
up :)

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] Dial-out via AMI

On 24/04/2009 2:24 a.m., Nhadie wrote:
 Hi,

 i'm currently using Originate command on AMI, i can call a certain
 channel like a SIP user SIP/1000 then once 1000 is answered it dials out
 to amobile or landline.

 Would just like to know if i can use AMI to dialout to a mobile or
 landline first (instead of SIP user) and once answered, dial another
 mobile or landline again.

 If not is it possible to call a macro from the AMI? i think i can
 probably use AGI for this, but i don't know if i can call a macro from
 the AMI command.

Asterisk will always dial the channel first then the 
context/extension/application etc.

Couple of things to bear in mind - you won't be able to tell if the call 
is answered if you are using analogue lines.

Action: Originate
Channel: Zap/g1/12345
Context: extensions
Extension: 1000
Priority: 1

The above will call 12345 and when connected (either when the call 
starts with analogue or when it is connected with digital) it will go to 
extension 1000 in the context extensions, where you would have something 
like:

[extensions]
exten = _1XXX,1,Dial(SIP/${EXTEN})

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] Dial-out via AMI

On Thu, Apr 23, 2009 at 6:11 PM, Matt Riddell li...@venturevoip.com wrote:

 On 24/04/2009 2:24 a.m., Nhadie wrote:
  Hi,
 
  i'm currently using Originate command on AMI, i can call a certain
  channel like a SIP user SIP/1000 then once 1000 is answered it dials out
  to amobile or landline.
 
  Would just like to know if i can use AMI to dialout to a mobile or
  landline first (instead of SIP user) and once answered, dial another
  mobile or landline again.
 
  If not is it possible to call a macro from the AMI? i think i can
  probably use AGI for this, but i don't know if i can call a macro from
  the AMI command.

 Asterisk will always dial the channel first then the
 context/extension/application etc.

 Couple of things to bear in mind - you won't be able to tell if the call
 is answered if you are using analogue lines.

 Action: Originate
 Channel: Zap/g1/12345
 Context: extensions
 Extension: 1000
 Priority: 1

 The above will call 12345 and when connected (either when the call
 starts with analogue or when it is connected with digital) it will go to
 extension 1000 in the context extensions, where you would have something
 like:

 [extensions]
 exten = _1XXX,1,Dial(SIP/${EXTEN})

 --
 Kind Regards,

 Matt Riddell
 Director
 ___

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 http://www.venturevoip.com/news.php (Daily Asterisk News - html)
 http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)



A much more scalable way to do this is to create and then FTP or move .call
files to the proper directory.  Depends how much you plan on banging on the
AMI.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
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Re: [asterisk-users] Dial-out via AMI

On 24/04/2009 10:19 a.m., Steve Totaro wrote:
 A much more scalable way to do this is to create and then FTP or move
 .call files to the proper directory.  Depends how much you plan on
 banging on the AMI.

Maybe, but the Asterisk Manager is happy with 10 calls per second and if 
your controlling process is spread across Asterisk machines you can do 
hundreds of calls per second.  If you're doing what it seems he is, I'd 
agree that call files may be easier, but I'm not sure it scales better. 
  How many call files can you put in a directory, is he using a hard 
drive or compact flash (max writes).

I've never actually done any proper tests of the comparison between a 
call file and a manager originate.  I would have thought they were 
pretty much the same, albeit that you're adding an extra layer of 
complexity with the call files.

-- 
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

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Re: [asterisk-users] Dial-out via AMI

On Thu, Apr 23, 2009 at 6:43 PM, Matt Riddell li...@venturevoip.com wrote:

 On 24/04/2009 10:19 a.m., Steve Totaro wrote:
  A much more scalable way to do this is to create and then FTP or move
  .call files to the proper directory.  Depends how much you plan on
  banging on the AMI.

 Maybe, but the Asterisk Manager is happy with 10 calls per second and if
 your controlling process is spread across Asterisk machines you can do
 hundreds of calls per second.  If you're doing what it seems he is, I'd
 agree that call files may be easier, but I'm not sure it scales better.
  How many call files can you put in a directory, is he using a hard
 drive or compact flash (max writes).

 I've never actually done any proper tests of the comparison between a
 call file and a manager originate.  I would have thought they were
 pretty much the same, albeit that you're adding an extra layer of
 complexity with the call files.

 --
 Kind Regards,

 Matt Riddell
 Director
 ___

 http://www.venturevoip.com (Great new VoIP end to end solution)
 http://www.venturevoip.com/news.php (Daily Asterisk News - html)
 http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)


I have in VERY intensive AMI usage environments.  You can easily drop 100 or
more call files in the spool dir and no worries.  @ ~100 calls it may take a
second or so to ring (all SIP extensions).

I have no doubt that it is the more stable and scalable way to go.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
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Re: [asterisk-users] BLINDTRANSFER and SIP hardphones

2009-04-23 Thread Kevin P. Fleming

Olivier wrote:

 When a SIP hardphone is transfering a call while ringing (caller and
 callee don't speak to each other) using phone's Transfer key, it seems
 BLINDTRANSFER remains empty.
 Though I can see a 302 MOVED TEMPORARILY message coming in.

If the person performing the transfer has dialed the transferee's number
and hears the call ringing, that is not a blind transfer, it is an
attended to transfer to a call that hasn't been answered yet. There
won't be any variables set for blind transfer, as it isn't one.

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

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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread Rob Hillis

Darrick Hartman wrote:
 Rob Hillis wrote:
   
 Daily?  No.  However, after implementing a weekly restart of Asterisk,
 I've found the instance of lockups and CPU utilisation spikes have
 decreased significantly.
 

 Unless you're using some unstable modules, there really should be no 
 need to restart Asterisk.  Is there a certain activity that is causing 
 these lockups?  I have low power systems which haven't had Asterisk 
 restarted in months many times.  Granted, these are mostly low call 
 volume systems, but unless there is a memory leak, you should not needed 
 to restart the Asterisk process.  (my guess is one of the modules you 
 are using has some sort of problem).

This particular system isn't low power - it's a full blown server. 
Since I don't work at this place, I don't know what people are doing at
the time the system freezes up.

It's been some time since I updated Asterisk at this site, so they're
probably running version 1.4.17 - 1.4.20 there. (it's a voluntary
organisation where I've since become sick of (a) the politics and (b)
their expectation that I drop what I'm doing to help them, regardless of
whether I'm at work or not)

If I were to do things again, I'd be running Astlinux on a net 5501 with
an integrated hard drive (for voicemail/IVR and so on)  Only time I've
ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
time to upgrade Astlinux.


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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread Darrick Hartman

Rob Hillis wrote:
 Darrick Hartman wrote:
 Rob Hillis wrote:
   
 Daily?  No.  However, after implementing a weekly restart of Asterisk,
 I've found the instance of lockups and CPU utilisation spikes have
 decreased significantly.
 
 Unless you're using some unstable modules, there really should be no 
 need to restart Asterisk.  Is there a certain activity that is causing 
 these lockups?  I have low power systems which haven't had Asterisk 
 restarted in months many times.  Granted, these are mostly low call 
 volume systems, but unless there is a memory leak, you should not needed 
 to restart the Asterisk process.  (my guess is one of the modules you 
 are using has some sort of problem).
 
 This particular system isn't low power - it's a full blown server. 
 Since I don't work at this place, I don't know what people are doing at
 the time the system freezes up.
 
 It's been some time since I updated Asterisk at this site, so they're
 probably running version 1.4.17 - 1.4.20 there. (it's a voluntary
 organisation where I've since become sick of (a) the politics and (b)
 their expectation that I drop what I'm doing to help them, regardless of
 whether I'm at work or not)

Ah.  That's probably the issue.  There were some significant bugs in 
some of the releases in that range.

 If I were to do things again, I'd be running Astlinux on a net 5501 with
 an integrated hard drive (for voicemail/IVR and so on)  Only time I've
 ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
 time to upgrade Astlinux.

That's what we like to hear!  Did you update to the latest version (0.6.5)?

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Re: [asterisk-users] AMD Not Working

On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell li...@venturevoip.com wrote:

 On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
  Hi All,
 
  I am trying to use the AMD (Answering Machine Detect).
  But it is not sending the AMD_Status as either
  the Human or Machine, it hangs up in middle.

 I'd say that the remote end of the call is hanging up - do a SIP debug
 so you can see what happens - the best way to test things like this is
 by calling your own number - that way you can guarantee it doesn't hang
 up :)

 --
 Kind Regards,

 Matt Riddell
 Director
 ___

 http://www.venturevoip.com (Great new VoIP end to end solution)
 http://www.venturevoip.com/news.php (Daily Asterisk News - html)
 http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)



You can also run Orecx on the localhost (for very small production or lab
systems) or on a different host via mirrored switch port and then listen to
all calls (SIP and other VoIP), or RTPTap via Sangoma cards).

I have done this many times to catch intermittent problems that are
continuously reported by users but cannot be readily reproduced.  I just ask
that the user log the time of the call and what they experienced, then I can
listen to the recording, ascertain all the critical info that users leave
off trouble reports, and figure out the commonalities.  Obviously, all due
notice/permission and/or legal disclosures should be made/given before
recording anything.

It is great for troubleshooting (and yes, calls do get crossed and all kinds
of other strangness in Asterisk, you know, what you write off as user error
:-)

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
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[asterisk-users] Dahi-tools Compilation on Ubuntu/Xen

2009-04-23 Thread i...@ameri.me

Hi all,

I'm trying to compile dahdi-tools on Ubuntu 8.04 on Xen (Amazon EC2 to be 
exact). dahdi-linux compiled and installed successfully, after which I do the 
following to install dahdi-tools:

wget 
http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
tar xzvf dahdi-tools-current.tar.gz
cd dahdi-tools*
./configure
make
make install
make config


Everything seems to go well, the installation is successful, and I can start 
the dahdi service after this and test it and it all seems fine.

The issue then is that my whole system becomes unusable. It seems the libc6 
files go missing. calls to most programs, such as apt-get result in the 
following error:

apt-get: /usr/local/lib/libstdc++.so.6: no version information
available (required by apt-get)
apt-get: /usr/local/lib/libstdc++.so.6: no version information
available (required by /usr/lib/libapt-pkg-libc6.7-6.
so.4.6)
.
.
.
apt-get: relocation error: apt-get: symbol
_ZSt16__ostream_insertIcSt11char_traitsIcEERSt13basic_ostreamIT_T0_ES6_PKS3_i,
version GLIBCXX_3.4.9 not defined in file libstdc++.so.6 with link
time reference


I have also tried configuring dahdi-tools with

./configure cc=gcc-4.0.2

GCC 4.0.2 is the version of gcc with which my kernel source is compiled, and 
it is installed on the system.

But the results are the same. An unusable system.

I'm pretty sure I'm missing something obvious here, but I fail to see what. 
Any hints/advice will be greatly welcome.

Cheers
-- 
Aryan Ameri

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Re: [asterisk-users] Dahi-tools Compilation on Ubuntu/Xen

2009-04-23 Thread Tzafrir Cohen

On Fri, Apr 24, 2009 at 09:54:15AM +1000, i...@ameri.me wrote:
 Hi all,
 
 I'm trying to compile dahdi-tools on Ubuntu 8.04 on Xen (Amazon EC2 to be 
 exact). dahdi-linux compiled and installed successfully, after which I do the 
 following to install dahdi-tools:
 
 wget 
 http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
 tar xzvf dahdi-tools-current.tar.gz
 cd dahdi-tools*
 ./configure
 make
 make install
 make config
 
 
 Everything seems to go well, the installation is successful, and I can start 
 the dahdi service after this and test it and it all seems fine.
 
 The issue then is that my whole system becomes unusable. It seems the libc6 
 files go missing. calls to most programs, such as apt-get result in the 
 following error:
 
 apt-get: /usr/local/lib/libstdc++.so.6: no version information
 available (required by apt-get)
 apt-get: /usr/local/lib/libstdc++.so.6: no version information
 available (required by /usr/lib/libapt-pkg-libc6.7-6.
 so.4.6)

/usr/local/lib/libstdc++ ???

Did you put it there?

Please provide the output of the following:

ls -l /lib/libstdc++* /usr/local/lib/libstd++*

grep . /etc/ld.so.conf /etc/ld.so.conf.d/*

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

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Re: [asterisk-users] Asterisk routine maintenance activities

2009-04-23 Thread Rob Hillis

Darrick Hartman wrote:
 If I were to do things again, I'd be running Astlinux on a net 5501 with
 an integrated hard drive (for voicemail/IVR and so on)  Only time I've
 ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
 time to upgrade Astlinux.
 

 That's what we like to hear!  Did you update to the latest version (0.6.5)?

Is the Pope Catholic?

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Re: [asterisk-users] Dahi-tools Compilation on Ubuntu/Xen

2009-04-23 Thread Aryan Ameri

On Fri Apr 24 2009 10:33:28 GMT+1000 (EST) Tzafrir Cohen 
tzafrir.co...@xorcom.com wrote:
 Everything seems to go well, the installation is successful, and I can start 
 the dahdi service after this and test it and it all seems fine.

 The issue then is that my whole system becomes unusable. It seems the libc6 
 files go missing. calls to most programs, such as apt-get result in the 
 following error:

 apt-get: /usr/local/lib/libstdc++.so.6: no version information
 available (required by apt-get)
 apt-get: /usr/local/lib/libstdc++.so.6: no version information
 available (required by /usr/lib/libapt-pkg-libc6.7-6.
 so.4.6)
 
 /usr/local/lib/libstdc++ ???
 
 Did you put it there?

No. I didn't put anything there. The system is working fine before installing 
dahdi-tools, and becomes unusable with these /usr/local/lib/libstdc++ after 
that. This is on a fresh install of Ubuntu 8.04 with no other software 
installed but the minimum base. Something is going wrong here.

 Please provide the output of the following:
 
 ls -l /lib/libstdc++* /usr/local/lib/libstd++*

ls: cannot access /lib/libstdc++*: No such file or directory
ls: cannot access /usr/local/lib/libstd++*: No such file or directory

 grep . /etc/ld.so.conf /etc/ld.so.conf.d/*

/etc/ld.so.conf:include /etc/ld.so.conf.d/*.conf
/etc/ld.so.conf.d/i486-linux-gnu.conf:# Multiarch support
/etc/ld.so.conf.d/i486-linux-gnu.conf:/lib/i486-linux-gnu
/etc/ld.so.conf.d/i486-linux-gnu.conf:/usr/lib/i486-linux-gnu
/etc/ld.so.conf.d/libc6-xen.conf:hwcap 0 nosegneg
/etc/ld.so.conf.d/libc.conf:# libc default configuration
/etc/ld.so.conf.d/libc.conf:/usr/local/lib

I think I should also mention that after installing dahdi-tools and doing make 
config, I got the following output:
-
DAHDI has been configured.

If you have any DAHDI hardware it is now recommended you
edit /etc/dahdi/modules in order to load support for only
the DAHDI hardware installed in this system.  By default
support for all DAHDI hardware is loaded at DAHDI start.

I think that the DAHDI hardware you have on your system is:


And that's it. It just returns to the bash shell after that.

Regards,
-- 
Aryan Ameri

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[asterisk-users] Hangup Detection After Originate (Asterisk Manager API)

2009-04-23 Thread Saurabh Nirkhey

I  have written an asterisk manager client which creates an outbound call
using Asterisk manager API's Originate action.
when the call is connected I run 3 applications on it.
1)read a dtmf digit from user
2)A customized application which I have written,(It plays something to user)
3)Hangup

If user hangs up while app 2(see above) is executing I get a 'Event Hangup'
from asterisk in my manager client .
But if app2 is over and asterisk executes Hangup (app3),It never sends any
packet to my client regarding Hangup of the call.

I have given all permissions to manager user in manager.conf.
Can somebody help me?

Thanks  Regards
===
(-:  Saurabh   :-)
===

French is the language of love,For everything else there is 'C'   

Every search begins with beginner's luck and ends with the victor being
severly tested
-Paulo Coehlo
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[asterisk-users] cheap CHEAP ata

2009-04-23 Thread David fire

hi i need many cheaps atas or some very cheap way to connect analogs phones
to asterisk
what do you recomend? i searches and only find solutions like 40 U$D (in the
states, here in argentina is like 80 U$D) per phone any links or something?
thanks!
David

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Re: [asterisk-users] cheap CHEAP ata

2009-04-23 Thread Aryan Ameri

On Fri Apr 24 2009 14:03:14 GMT+1000 (EST) David fire ddf...@gmail.com wrote:
 hi i need many cheaps atas or some very cheap way to connect analogs 
 phones to asterisk
 what do you recomend? i searches and only find solutions like 40 U$D (in 
 the states, here in argentina is like 80 U$D) per phone any links or 
 something?
 thanks!
 David

ATAs are price-competitive when you are connecting a couple of analogue 
phones, but they quickly lose their advantage if you want to use many analogue 
phones.

The cheapest way to achieve what you want to achieve is to build an Asterisk 
box yourself, put in a Digium or Sangoma card and put as many FXS modules on 
the board as you require. You can usually find cards which accept daughter 
boards which mean you can even install additional FXS modules on the same card.

Cheers
-- 
Aryan Ameri

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Re: [asterisk-users] cheap CHEAP ata

2009-04-23 Thread David fire

thanks for your answer but the boards are too expensive.
4 fxs digium card is like 600.
the 12 fxs open vox board (i have one) is like 40U$D per phone. (in the
states).
i need cheap!!!
David

2009/4/24 Aryan Ameri i...@ameri.me

 On Fri Apr 24 2009 14:03:14 GMT+1000 (EST) David fire ddf...@gmail.com
 wrote:
  hi i need many cheaps atas or some very cheap way to connect analogs
  phones to asterisk
  what do you recomend? i searches and only find solutions like 40 U$D (in
  the states, here in argentina is like 80 U$D) per phone any links or
  something?
  thanks!
  David

 ATAs are price-competitive when you are connecting a couple of analogue
 phones, but they quickly lose their advantage if you want to use many
 analogue
 phones.

 The cheapest way to achieve what you want to achieve is to build an
 Asterisk
 box yourself, put in a Digium or Sangoma card and put as many FXS modules
 on
 the board as you require. You can usually find cards which accept daughter
 boards which mean you can even install additional FXS modules on the same
 card.

 Cheers
 --
 Aryan Ameri

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Re: [asterisk-users] BLINDTRANSFER and SIP hardphones