[asterisk-users] CDR and Queue Reporting windows application looking for Beta testers!

[Token PBX]

Hi!

I've been on this list for over 3 years and this is my first post.

We have a reporting application for Asterisk that is soon to be in beta.
It's a windows application that generates reports from log files (CDR and
queue).
It has a drag and drop approach to report creating.  There is a pivot grid
and you just put the data that you want in a report, where you want it.
We're currently taking beta sign-ups.
@ http://samreports.com

Regards.

Mihaela MJ.
http://samreports.com
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Re: [asterisk-users] Gigaset 450IP loses registration

[Token PBX]

Hi!

I have the same phone with the same problems:

1. Asterisk box does not have fixed IP address, but dyndns name.
2. Phone is at a different location, connected to a router/ADSL modem
Siemens Gigaset  (with option not to disconnect from internet ever - set
on).
3. Inside asterisk LAN, phone didn't loose connection ever.
4. In sip.conf  NAT is set
5. In phone settings NAT is set also, and sip proxy is set to asterisk's box
dyndns name.
6. When phone is seen as unreachable by asterisk box, and router is reset on
remote location, phone reregisters.

Any help is appreciated.
Thnx


Mihaela MJ.





On 7/3/07, gincantalupo <[EMAIL PROTECTED]> wrote:


Hi Olivier,
I forgot to mention it is a C450IP.
But if you have some hint on S maybe it can help me. Perhaps it is some
configuration...I tried with qulify=no as I read on a web page without
success.

Thank you.

Giorgio Incantalupo


Olivier wrote:
> Is it a S 450IP ou C 450IP ?
> 
>
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Re: [asterisk-users] Sample Config.

[Token PBX]

Hi!



I don't understand  what you mean by : „configure voice part on it", but I
can give general guidelines:



First you setup SPA3000 web UI:

1) Line1 Tab:



Sip settings:

  SIP port : 5060



Proxy and Registration:

  Proxy: Asterisk IP



Subscriber Information:

  Display Name: FXS_username

  Password: FXS password

  User ID: FXS_username



2) PSTN Line Tab:



SIP Settings:

  SIP port: 5061



Proxy and Registration:

  Proxy: Asterisk IP



Subscriber Information:

  Display Name: FXO_username

  Password: FXO_password

  User ID: FXO_username



Dial Plans:

  Dial Plan 1: ()(may be any other dial plan)



VoIP-To-PSTN Gateway Setup:

  VoIP-To-PSTN Gateway Enable: Yes

  Line 1 VoIP Caller DP: 1 (or any other setup like Dial Plan 1)



VoIP Users and Passwords (HTTP Authentication)

  VoIP User 1 Auth ID: asterisk

  VoIP User 1 DP: 1(same as above)



PSTN-To-VoIP Gateway Setup:

  PSTN-To-VoIP Gateway Enable: Yes





Then Asterisk sip.conf:



[ FXO_username]

disallow=all

allow=alaw

type=friend

fromuser= FXO_username

username= FXO_username

secret= FXO_password

host=dynamic

dtmfmode=rfc2833

canreinvite=no

qualify=1000

context=incoming

port=5061



[FXS_username]

disallow=all

allow=alaw

type=friend

username= FXS_username

secret= FXS_password

host=dynamic

dtmfmode=rfc2833

canreinvite=no

qualify=1000

context=outgoing

Best regards
Mihaela MJ


On 1/26/07, Jonson Player <[EMAIL PROTECTED]> wrote:


Hello,
I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to
configure voice part on it. I cannot get it if I can use like peer for my
asterisk. Please help me with some tips.
Thank you guys.

Regards,
Jonson.

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Re: [asterisk-users] Cisco 7970 Unprovisioned

[Token PBX]

Hi everyone!

I just want to thank everybody. My phone works now and  just a little hint:
set qualify=no in sip.conf  of your phone's extension.

Best regards
Mihaela MJ

On 1/21/07, Token PBX <[EMAIL PROTECTED]> wrote:




On 1/20/07, Pavel Jezek <[EMAIL PROTECTED]> wrote:
>
> you have probably something wron in config file and phone refuses to
> configure,
> here is my minimalistic file for 7941/61, you can try...
>
> 
> SIP
> admin
> admin
> 
> 
>D-M-Y
>Central Europe Standard/Daylight Time
>
> 
> ntpserver
> 
>
> 
> 
>
>   
>  
> 
>2000
>5060
>5061
> 
> asteriskserver
>  
>   
>
> 
> 
>
> 
> 
>true
> 
> false
> g729a
> 0
> SIP
> 
>
>   9
>   SIP 999
>   asteriskserver
>   999
>   yourname
>   999
>   xxx
>   
>
>
>   21
>   Helpdesk
>   5880
>
> 
> DRdialplan.xml
> 
>
> 
> admin
> 
>
> SIP41.8-2-1S
>
>
> 1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37
>
>
> 
> 
> 
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Hi!

Here's my configuration file:



true
SIP
user
pass



  Default
  
CMLocal
D.M.Y
W. Europe Standard/Daylight Time
  

  

  

  
2000
  
  My Asterisk IP

  

  

  
Enable
Enable
true
My Asterisk IP
2000

2000

2000
  

  -1
  Default
  Default





true
2
 

  
  
false
false
1
0
1
0
0
1
0
1
1,7
08:30
11:30
01:00
1
1
  


1136931633-57191cee-5ffc-4342-b286-4246b4991890

  
English_United_States
1
en_US
1.0.0.0-1
iso-8859-1
  

  United_States
  
United_States
64
1.0.0.0-1
  

  1
  120
  
  
  
  
  
  
  
  96
  0
  96
  0

  

  3804
  ccm-beta-5-1

  

  
  false




  My Asterisk IP
  5060
  My Asterisk IP
  5060
  My Asterisk IP
  5060
  true



  true
  x-cisco-serviceuri-cfwdall
  x-cisco-serviceuri-pickup
  x-cisco-serviceuri-opickup
  x-cisco-serviceuri-gpickup
  x-cisco-serviceuri-meetme
  x-cisco-serviceuri-abbrdial
  false
  2
  true
  true
  2
  2
  0
  true



  6
  10
  180
  3600
  5
  120
  120
  5
  500
  4000
  70
  true
  None


  1
  false
  true
  false
  false
  none
  101
  3
  avt
  false
  false
  3
  0
  My Asterisk IP
  My company's name.
  2
  false
  15000

10

  true
  16384
  32766

  

  
9
extension
My Asterisk IP
5060
extension
extension

  2

3
extension
extension password
false
3
*97
4
5
extension

true
false
false
true

  

  
  21
  Some name
  Some tel number
  

  

  5060
  184
  0
  dialplan.xml
  SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml



1136931633-57191cee-5ffc-4342-b286-4246b4991890




I bought this phone from a former client who provided me with 8.0.3 SIP
firmware *.cop file and that was it. It's all I have. I don't have Cisco
tech support account or anything like that. I thought it might leave a good
impression on perspective clients seeing this phone operational on my desk.

Thanks again.
Mihaela MJ




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Re: [asterisk-users] Question about FXO/FXS device.

[Token PBX]

Hi!

I have several SPA3000 devices (older versions of SPA3102) and they are
working OK, sound quality is good. It is very configurable to the slightest
details. I use it whenever I need just one or two FXO ports, like for small
scale PSTN integration, or for connecting some other equipment that requires
FXO like GSM Gateway.
Best regards

Mihaela MJ

On 1/17/07, Jonson Player <[EMAIL PROTECTED]> wrote:


Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about 
SPA3102.
What you guys think about it. Is ok, is working with asterisk, can i use it
like voip peer. Thank you for your advice.

Jonson.
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Re: [asterisk-users] Cisco 7970 Unprovisioned

[Token PBX]

On 1/20/07, Pavel Jezek <[EMAIL PROTECTED]> wrote:


you have probably something wron in config file and phone refuses to
configure,
here is my minimalistic file for 7941/61, you can try...


SIP
admin
admin


   D-M-Y
   Central Europe Standard/Daylight Time
   

ntpserver

   


   
  
 

   2000
   5060
   5061

asteriskserver
 
  
   





   true

false
g729a
0
SIP

   
  9
  SIP 999
  asteriskserver
  999
  yourname
  999
  xxx
  
   
   
  21
  Helpdesk
  5880
   

DRdialplan.xml



admin


SIP41.8-2-1S


1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37





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Hi!

Here's my configuration file:



true
SIP
user
pass



 Default
 
   CMLocal
   D.M.Y
   W. Europe Standard/Daylight Time
 

 
   
 
   
 
   2000
 
 My Asterisk IP
   
 
   
 

 
   Enable
   Enable
   true
   My Asterisk IP
   2000
   
   2000
   
   2000
 

 -1
 Default
 Default




   
   true
   2


 
 
   false
   false
   1
   0
   1
   0
   0
   1
   0
   1
   1,7
   08:30
   11:30
   01:00
   1
   1
 


1136931633-57191cee-5ffc-4342-b286-4246b4991890

 
   English_United_States
   1
   en_US
   1.0.0.0-1
   iso-8859-1
 

 United_States
 
   United_States
   64
   1.0.0.0-1
 

 1
 120
 
 
 
 
 
 
 
 96
 0
 96
 0

 
   
 3804
 ccm-beta-5-1
   
 

 
 false




 My Asterisk IP
 5060
 My Asterisk IP
 5060
 My Asterisk IP
 5060
 true



 true
 x-cisco-serviceuri-cfwdall
 x-cisco-serviceuri-pickup
 x-cisco-serviceuri-opickup
 x-cisco-serviceuri-gpickup
 x-cisco-serviceuri-meetme
 x-cisco-serviceuri-abbrdial
 false
 2
 true
 true
 2
 2
 0
 true



 6
 10
 180
 3600
 5
 120
 120
 5
 500
 4000
 70
 true
 None


 1
 false
 true
 false
 false
 none
 101
 3
 avt
 false
 false
 3
 0
 My Asterisk IP
 My company's name.
 2
 false
 15000

10

 true
 16384
 32766

 

 
   9
   extension
   My Asterisk IP
   5060
   extension
   extension
   
 2
   
   3
   extension
   extension password
   false
   3
   *97
   4
   5
   extension
   
   true
   false
   false
   true
   
 

 
 21
 Some name
 Some tel number
 

 

 5060
 184
 0
 dialplan.xml
 SK50719900-3bee-4594-bc3f-6400e1a33bf0.xml



1136931633-57191cee-5ffc-4342-b286-4246b4991890




I bought this phone from a former client who provided me with 8.0.3 SIP
firmware *.cop file and that was it. It's all I have. I don't have Cisco
tech support account or anything like that. I thought it might leave a good
impression on perspective clients seeing this phone operational on my desk.

Thanks again.
Mihaela MJ
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Re: [asterisk-users] Cisco 7970 Unprovisioned

[Token PBX]

On 1/20/07, Jon Farmer <[EMAIL PROTECTED]> wrote:


Are you setting the TFTP server address in the DHCP?

Are you checking the TFTP log to see what files the phone is requesting
and not finding?

Regards

Jon

Jon Farmer
Telford, Shropshire, UK





Hi Jon!


Yes I checked log, and phone requested and loaded all required files and
then some:
It also requested file: "CTLSEP-MAC.tlv",  that has something to do with
license.

Since it couldn't find it returned error and continued to load "SEP-
MAC.cnf.xml" .

Phone booted with SIP firmware but did not load any of the settings from
"SEP-MAC.cnf.xml". I checked that from phone's display.  None of the
settings were loaded, no sip proxy address, phone label, SIP lines etc.. All
was blank. Just some dynamically assigned settings were set like DHCP
address, phone's IP and such.



I followed instructions from wiki voip-info when building  "SEP-
MAC.cnf.xml".



Please help and thanks.



Mihaela MJ


- Original Message 

From: Token PBX <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Sent: Saturday, 20 January, 2007 1:01:25 PM
Subject: [asterisk-users] Cisco 7970 Unprovisioned

Hi!

I did manage to load phone with SIP image : SIP70.8-0-3S, made
SEP-MAC.cnf.xml, but phone never read the configuration from it.
On the screen it's written "Unprovisioned", and phone is not trying to
register with asterisk.

Please help!!

MihaelaMJ

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[asterisk-users] Cisco 7970 Unprovisioned

[Token PBX]

Hi!

I did manage to load phone with SIP image : SIP70.8-0-3S, made
SEP-MAC.cnf.xml, but phone never read the configuration from it.
On the screen it's written "Unprovisioned", and phone is not trying to
register with asterisk.

Please help!!

MihaelaMJ
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[Asterisk-Users] PDA softphone....

[pbx]

I have downloaded SJPhone - and well.. it does connect to my system,
however popping audio is heard when i dial my music on hold extension...

the quality is really really bad..

i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is
that sufficient? The codecs for sjphone are fixed at 64000.. i could not
change those values.

has anyone had successful attempts with something better?

Thanks...

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Re: [Asterisk-Users] iaxmodem

[pbx]

mgetty dump:


10/25 11:23:17 IAX  tio_get_rs232_lines: TIOCMGET failed: Invalid argument
10/25 11:23:17 # data dev=ttyIAX, pid=15158, caller='none',
conn='DIRECT', name='', cmd='/bin/login', user='Fedora Core release 3
(Heidelberg)'

--
10/25 11:23:36 IAX  mgetty: experimental test release 1.1.31-Jul24
10/25 11:23:36 IAX  check for lockfiles
10/25 11:23:36 IAX  locking the line
10/25 11:23:36 IAX  tio_get_rs232_lines: TIOCMGET failed: Invalid argument
10/25 11:23:36 IAX  WARNING: DSR is off - modem turned off or bad cable?
10/25 11:23:36 IAX  lowering DTR to reset Modem
10/25 11:23:36 IAX  TIOCMBIC failed: Invalid argument
10/25 11:23:36 IAX  clean_line: only 500 of 4095 bytes logged
10/25 11:23:37 IAX  waiting...


i have in my /etc/inittab:

/sbin/mgetty ttyIAX -F -r /dev/ttyIAX

the -F is for Fax only and the -r is do not send modem init

Then on the iaxmodem output i get a bunch of:
  Timestamp: 12001ms  SCall: 06850  DCall: 00012 [192.168.1.1:4569]
Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: PONG
   Timestamp: 12001ms  SCall: 00012  DCall: 06850 [192.168.1.1:4569]
   Unknown IE 046  : Present
   Unknown IE 047  : Present
   Unknown IE 048  : Present
   Unknown IE 049  : Present
   Unknown IE 050  : Present
   Unknown IE 051  : Present

Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK
   Timestamp: 12001ms  SCall: 06850  DCall: 00012 [192.168.1.1:4569]
Unable to pass the full buffer onto the device file. -1 bytes of 4
written.Unable to pass the full buffer onto the device file. -1 bytes of 2
written.Unable to pass the full buffer onto the device file. 12 bytes of
25 written.Unable to pass the full buffer onto the device file. -1 bytes
of 12 written.Unable to pass the full buffer onto the device file. -1
bytes of 2 written.Unable to pass the full buffer onto the device file. -1
bytes of 4 written.Unable to pass the full buffer onto the device file. -1
bytes of 2 written.Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 005 Type:
IAX Subclass: HANGUP


Anyways.. it's a nice idea... and if spandsp supported data...it would be
terrific!!!


> [EMAIL PROTECTED] wrote:
>
>>The comment below makes me wonder could ttyIAX be configured to answer
>>mgetty?
>>
>>
>
> Although I've not tried mgetty with IAXmodem, the intent was to make
> this possible (for faxing), yes.
>
>>I have made the mgetty talk to ttyIAX however, as soon as a ring comes
>>into th eextension , mgetty shuts down... so I cannot keep the signal up.
>>
>>
>
> What does the mgetty logging say about what it's doing?
>
>>I tried to use the pppd daemon directly with ttyIAX and it said that the
>>link is in serial loopback disconnecting.
>>
>>Would using iaxModem be feasable for a pppd dialin, or how could I use
>>mgetty with pppd to start it?
>>
>
> spandsp (which is used by IAXmodem) does not currently support data calls.
>
> Lee.
>
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Re: [Asterisk-Users] iaxmodem

[pbx]

The comment below makes me wonder could ttyIAX be configured to answer
mgetty?

I have made the mgetty talk to ttyIAX however, as soon as a ring comes
into th eextension , mgetty shuts down... so I cannot keep the signal up.

I tried to use the pppd daemon directly with ttyIAX and it said that the
link is in serial loopback disconnecting.

Would using iaxModem be feasable for a pppd dialin, or how could I use
mgetty with pppd to start it?

thanks


>
> For hylafax to answer a call, you need to use faxgetty.. Add this 2
> lines to your /etc/inittab  and run   "init q"  to force a reload:


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Re: [Asterisk-Users] TDMoE and Badness in Kernel

[pbx]

2.6.12


> On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
>> I received some postings back, as I was trying to do the same thing.
>>
>> it' is a problem with Kernel 2.6... 2.4 works fine .. this is the
>> summary
>> I got from reading the posts before.
>>
>> I hope that helps... I dont have the ability to go DOWn in kernel to
>> 2.4..
>>
>
> the wiki suggested that it was a problem with softirq.c in the kernel
> and that this was fixed at some point.  What 2.6 version are you running
> that you have this problem?
>
> --
> Trixter http://www.0xdecafbad.com Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
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Re: [Asterisk-Users] TDMoE and Badness in Kernel

[pbx]

I received some postings back, as I was trying to do the same thing.

it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary
I got from reading the posts before.

I hope that helps... I dont have the ability to go DOWn in kernel to 2.4..



> I'm going to poll the group one more time on this one. I have posted
> this before and didn't get any takers.
>
> Digium advises that I should just do IAX in place of TDMoE but I don't
> have that luxury. I have a very complex dial plan built around the TDMoE
> functionality and it would be very difficult/expensive to rewrite it.
> This has always worked excellent on 2.4 but now that we need to upgrade
> to 2.6 I'm getting all kinds of headaches. I'm willing to pay a
> consultant to work this out for me. Please contact me off list if
> interested
>
> The following is my original message:
>
> "Badness in local_bh_enable at kernel/softirq.c on 2.6.X"
>
> I'm seeing this on Kernel 2.6.+ implementations, namely Centos 4.1, FC4
> machines while trying to do TDMoE trunks between two machines.
> 2.4 Kernel operates fine on the same hardware
>
> I'm compiling zaptel-1.0.9.2 as per instructions in README.Linux26 +
> README.udev. I've also tried CVS head zaptel.
>
> Here are some references where the issue has been reported before but
> I've yet to find a documented solution;
>
> http://lists.digium.com/pipermail/asterisk-users/2005-February/091867.html
>
> http://bugs.digium.com/view.php?id=5126
>
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

That did it... Thank you.

putting / after the register line caused it to not
register any more... and i would get error

server1.goiax.com/ could not be found.

anyways.. thanks for your help guys :)


> Replace
> [goiax]
> with
> [PHONENUMBER]
>
> username= don't work for users in IAX channel.
>
> On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
>> That is What I stated in the email.. my GOIAX #. not the DID #.
>>
>> That is not the issue.
>>
>> > for the incoming context put your goiax.com  phone
>> > number
>> > not the free DID number but the other one.
>> >
>> > On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>> >>
>> >> Trixter:
>> >>
>> >> Thanks for the guide to setting this up:... I have tried the below
>> >> configuration with my settings, and when I place /goiax-in after my
>> >> register command, my register statement fails.
>> >>
>> >> If i remove it. I get a Rejected connect attempt from goiax's server
>> IP,
>> >> trying to reach 's@'
>> >>
>> >> I have put my GoIAX # in default, local, as the extension, and
>> nothing.
>> >>
>> >> I dont know where to look next on why i'm getting the rejected
>> connect
>> >> attempt.
>> >>
>> >> Thanks..
>> >>
>> >> ./Ben
>> >>
>> >> > On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> >> >> Can anybody post a step by step setup guide, please?
>> >> >
>> >> > Its like anything else once you have signed up ...
>> >> >
>> >> > in iax.conf
>> >> > register =>
>> >> PHONENUMBER:[EMAIL PROTECTED]/goiax-in> >> PROTECTED]/goiax-in>
>> >> >
>> >> > [goiax]
>> >> > type = peer
>> >> > host = server1.goiax.com 
>> >> > context = default
>> >> > secret = PASSWORD
>> >> > allow = gsm
>> >> > ;allow = ulaw
>> >> > ;disallow = all
>> >> > notransfer = yes
>> >> > qualify = yes
>> >> > auth = md5
>> >> > username = PHONENUMBER
>> >> >
>> >> >
>> >> > replace PHONENUMBER with the 8782 number you were issued.
>> Replace
>> >> > PASSWORD with your password from you account signup.
>> >> >
>> >> > Then in extensions.conf
>> >> > ; for outbound
>> >> > exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
>> >> > exten => _1NX,2,Busy
>> >> > exten => _1NX,102,Congestion
>> >> > exten => _1NX,202,playback(tt-weasels)
>> >> >
>> >> > ; for inbound
>> >> > exten => goiax-in,1,DO WHATEVER HERE
>> >> >
>> >> > asterisk -rx reload
>> >> >
>> >> > you should be set.
>> >> >
>> >> >
>> >> > --
>> >> > Trixter http://www.0xdecafbad.com Bret McDanel
>> >> > UK +44 870 340 4605 Germany +49 801 777 555 3402
>> >> > US +1 360 207 0479 or +1 516 687 5200
>> >> > FreeWorldDialup: 635378
>> >> > ___
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>> >> Easynews.com--
>> >> >
>> >> > Asterisk-Users mailing list
>> >> > Asterisk-Users@lists.digium.com
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> > To UNSUBSCRIBE or update options visit:
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >>
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>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
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>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >
>> >
>> >
>> > --
>> > Tom Vile
>> > Baldwin Technology Solutions, Inc
>> > Consulting - Web Design - VoIP Telephony
>> > www.baldwintechsolutions.com 
>> > Phone: 518-631-2855 x205
>> > Phone: 845-652-4578 x205
>> > Phone: 978-203-3848 x205
>> > Fax: 518-631-2856
>> > ___
>> > --Bandwidth and Colocation sponsored by Easynews.com --
>> >
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>> > Asterisk-Users@lists.digium.com
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

That is What I stated in the email.. my GOIAX #. not the DID #.

That is not the issue.

> for the incoming context put your goiax.com  phone
> number
> not the free DID number but the other one.
>
> On 10/19/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>>
>> Trixter:
>>
>> Thanks for the guide to setting this up:... I have tried the below
>> configuration with my settings, and when I place /goiax-in after my
>> register command, my register statement fails.
>>
>> If i remove it. I get a Rejected connect attempt from goiax's server IP,
>> trying to reach 's@'
>>
>> I have put my GoIAX # in default, local, as the extension, and nothing.
>>
>> I dont know where to look next on why i'm getting the rejected connect
>> attempt.
>>
>> Thanks..
>>
>> ./Ben
>>
>> > On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> >> Can anybody post a step by step setup guide, please?
>> >
>> > Its like anything else once you have signed up ...
>> >
>> > in iax.conf
>> > register =>
>> PHONENUMBER:[EMAIL PROTECTED]/goiax-in> PROTECTED]/goiax-in>
>> >
>> > [goiax]
>> > type = peer
>> > host = server1.goiax.com 
>> > context = default
>> > secret = PASSWORD
>> > allow = gsm
>> > ;allow = ulaw
>> > ;disallow = all
>> > notransfer = yes
>> > qualify = yes
>> > auth = md5
>> > username = PHONENUMBER
>> >
>> >
>> > replace PHONENUMBER with the 8782 number you were issued. Replace
>> > PASSWORD with your password from you account signup.
>> >
>> > Then in extensions.conf
>> > ; for outbound
>> > exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
>> > exten => _1NX,2,Busy
>> > exten => _1NX,102,Congestion
>> > exten => _1NX,202,playback(tt-weasels)
>> >
>> > ; for inbound
>> > exten => goiax-in,1,DO WHATEVER HERE
>> >
>> > asterisk -rx reload
>> >
>> > you should be set.
>> >
>> >
>> > --
>> > Trixter http://www.0xdecafbad.com Bret McDanel
>> > UK +44 870 340 4605 Germany +49 801 777 555 3402
>> > US +1 360 207 0479 or +1 516 687 5200
>> > FreeWorldDialup: 635378
>> > ___
>> > --Bandwidth and Colocation sponsored by
>> Easynews.com--
>> >
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>> > Asterisk-Users@lists.digium.com
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> > To UNSUBSCRIBE or update options visit:
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>> --
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Tom Vile
> Baldwin Technology Solutions, Inc
> Consulting - Web Design - VoIP Telephony
> www.baldwintechsolutions.com 
> Phone: 518-631-2855 x205
> Phone: 845-652-4578 x205
> Phone: 978-203-3848 x205
> Fax: 518-631-2856
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Re: [Asterisk-Users] DID setup from goiax.com

[pbx]

Trixter:

Thanks for the guide to setting this up:... I have tried the below
configuration with my settings, and when I place /goiax-in after my
register command, my register statement fails.

If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'

I have put my GoIAX # in default, local, as the extension, and nothing.

I dont know where to look next on why i'm getting the rejected connect
attempt.

Thanks..

./Ben

> On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
>> Can anybody post a step by step setup guide, please?
>
> Its like anything else once you have signed up ...
>
> in iax.conf
> register => PHONENUMBER:[EMAIL PROTECTED]/goiax-in
>
> [goiax]
> type= peer
> host= server1.goiax.com
> context = default
> secret  = PASSWORD
> allow   = gsm
> ;allow  = ulaw
> ;disallow   = all
> notransfer  = yes
> qualify = yes
> auth= md5
> username= PHONENUMBER
>
>
> replace PHONENUMBER with the 8782 number you were issued.  Replace
> PASSWORD with your password from you account signup.
>
> Then in extensions.conf
> ; for outbound
> exten => _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
> exten => _1NX,2,Busy
> exten => _1NX,102,Congestion
> exten => _1NX,202,playback(tt-weasels)
>
> ; for inbound
> exten => goiax-in,1,DO WHATEVER HERE
>
> asterisk -rx reload
>
> you should be set.
>
>
> --
> Trixter http://www.0xdecafbad.com Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
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RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN

[pbx]

David:

Also port 1:2 is a good idea to forward to the server as well..


> David,
> Yes, I've also forwarded port 4569 to the server.
> Since the router is forwarding to the server, I cannot
> forward it to the client as well -- however, as the
> client isn't going out past the LAN, it shouldn't
> matter... unless there's something else going on that
> I don't know about.
> Thanks
> Wolfgang
>
>
> --- David J Carter <[EMAIL PROTECTED]> wrote:
>
>> Wolfgang wrote:  -
>>
>> I've already sunk several hours into this without
>> any
>> real progress, so I'd really appreciate any help  My
>> task is simple -- establish a connection between a
>> softphone on XP ProSP2 to a Asterisk server on Linux
>> FC4 over a LAN through a Netgear router. The server
>> will then go out to a PSTN termination service.
>>
>> Thus far, the PSTN termination connection works fine
>> -- I've opened up 4569 with iptables, and forwarded
>> 4569 to the server IP.  I am not, however, having
>> any
>> luck connecting the softphone to the server.
>>
>> I can telnet, ftp, and http to the server, but not
>> IAX2. Iaxping times out, registration by Idefisk and
>> Firefly also times out.
>>
>> The server fails to see the client as well.
>>
>> Here's a portion of my iax.conf:
>>
>> [client]
>> type=friend
>> username=client
>> secret=**
>> host=192.168.1.40
>> context=clientcon
>>
>> and extensions.conf:
>>
>> [clientcon]
>> exten => 2278,1,Dial(IAX2/client)
>>
>>
>>
> ==
>> You say you have 4569 configured in iptables, what
>> about the netgear router?
>>
>> Have you port forwarded 4569 there?
>>
>> Dave
>>
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>>
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>
>
>
>
>
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[Asterisk-Users] CDR problem with DST Channel

[pbx]

I have 3 different SIP extensions in my DIAL string.

i.e. I have HOME_PHONES_TO_RING=SIP/2000&SIP/2001&SIP/2002

so in my Extensions file i have Dial(${HOME_PHONES_TO_RING},30,tTr)

So... when the home phone line rings, all three phones ring.

Anyways.. the problem is.. in the CDR log, sometimes the log entry shows
2000, sometimes 2001, sometimes 2002

Only extension 2000 answered the call, yet, 2001 is listed as the
answering channel, or 2002 is listed as the answering channel? The LASTAPP
column all show Hangup, and disposition shows ANSWERED.

Is there a way to to force a flush to the CDR to make it reflect the
correct phone that answered?

Thanks

./Ben

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Re: [Asterisk-Users] RE: faxing to/from asterisk - new scripts

[pbx]

I'm game for using them /and testing them.

Ben..

> Roman:
>
> I created two bash scripts called Mail2Fax and Fax2Mail for use with the
> asterisk sever.
>
> They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
> They
> make using these apps a lot easier, including being able to mail to
> [EMAIL PROTECTED] for outgoing faxes and then extracting phone numbers from
> the
> subject line!  (Makes it easy to use with Sendmail without complex rules /
> virtual user tables).
>
> They also include error logs, parameter checking, etc.
>
> Let me know if you want them
>
> Michelle Dupuis
> Technical Support Specialist
> Oxford Consulting Group Ltd.
> Making IT work for your business...
>
> T: (519) 672-8238
> E:   [EMAIL PROTECTED]
> W:   www.ocg.ca
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Re: [Asterisk-Users] Auto-assign CallerID for all my FXS Interfaces

[pbx]

In an answer about the voicemail box extension, I have 3 different sipura
boxes linked to the same voicemail account?

Sipura 1 = Sipura 2000 - with line 1 being my home phone
Sipura841 - Extension 2 = also my home phone
Sipura 2 = Sipura 1001 = phone in the bedroom - being the home phone

so i can pick up any one of the three and get the stutter tone and check
voicemail.

I have the seprate ones, becuase after hours (after 10pm) i dont want to
ring the bedroom phone), etc..

$.02


> As far as I can tell, in order to have caller ID show up for calls from
> other
> internal phones, I have to set the caller ID on each channel in
> zapata.conf.
> This is tedious, and redundant (since Asterisk knows which extension is
> making the call, and it could look up the name from the voicemail
> configuration--if the extension matched the mailbox).
>
> Is there a way to set the caller ID for these calls from the voicemail
> information so I don't have to duplicate all the names?  Is there some
> variable or function that will return a voicemail name for an extension?
> Would this functionality be pretty trivial to implement?
>
> Also, the stutter dialtone mapping to mailboxes requires the same
> duplication. However, I can't see how Asterisk would know that a zap
> channel
> "belongs to" a single extension, so I can understand this requirement.
>
> --
> Shaw Terwilliger <[EMAIL PROTECTED]>
> SourceGear LLC
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Re: [Asterisk-Users] TDMOE Badness in kernel...

[pbx]

Right... I had seen the multiple issues, on the flip side, the only
solution was NOT to use Kernel 2.6.+...

So... I'm happy that the behaviour is reproduceable (from what I have seen
from my steps, to that of others, and other Distros..)

Anyone out there have the magic wand to make it work with 2.6.+?

Thanks..


> I'm seeing the same behavior on a Debian system with 2.6.12.
> I have two systems with Digium Quad T1s in each and I trunk them with
> TDMoEThis always worked great on 2.4 and up to 2.6.8 but beyond that
> it either spits out copious amounts of kernel badness and paralyzes the
> system completely or gives the same results as you mentioned with
> constant Alarms/Alarm Clears.
>
> I've mentioned it on this forum before as well.
>
> On Wed, 2005-10-05 at 09:49 -0700, [EMAIL PROTECTED] wrote:
>> Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel
>> versions?
>>
>> I'm having the issue that is in the Mantis bug database with badness
>> with
>> the kernel.
>>
>> My Story:
>>
>> I can get the dynamic span to come up and show OK in the zttool on both
>> machines. However i get errors every second (Warning: detected alarm on
>> channel 1... then channel 2...)
>>
>> And then the next second, i get : alarm cleared on channel 1,... channel
>> 2... etc...
>>
>> No call will go through across the link becuase of the alarms.
>>
>> It looks as if 2.4 Kernel works, but it would be a lot of work to go
>> back
>> in time.
>>
>> Can anyone give me some direction on this.
>>
>> I have setup IAX2 between the two machines, but I would like the ability
>> to use Dial(Zap/group number/Exten)
>>
>> I havent found a solution in reading through the wiki's about doing
>> something similar with IAX.. i.e (Dial/IAX2/group number/$exten)...
>> There
>> are some scripts and macros that require you to code variables and check
>> status of each trunk etc but it would be nice to use a group with
>> IAX,
>> and in the IAX.conf place iax in groups... (unless i just havent' found
>> it)...
>>
>> Help?
>>
>>
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[Asterisk-Users] TDMOE Badness in kernel...

[pbx]

Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions?

I'm having the issue that is in the Mantis bug database with badness with
the kernel.

My Story:

I can get the dynamic span to come up and show OK in the zttool on both
machines. However i get errors every second (Warning: detected alarm on
channel 1... then channel 2...)

And then the next second, i get : alarm cleared on channel 1,... channel
2... etc...

No call will go through across the link becuase of the alarms.

It looks as if 2.4 Kernel works, but it would be a lot of work to go back
in time.

Can anyone give me some direction on this.

I have setup IAX2 between the two machines, but I would like the ability
to use Dial(Zap/group number/Exten)

I havent found a solution in reading through the wiki's about doing
something similar with IAX.. i.e (Dial/IAX2/group number/$exten)... There
are some scripts and macros that require you to code variables and check
status of each trunk etc but it would be nice to use a group with IAX,
and in the IAX.conf place iax in groups... (unless i just havent' found
it)...

Help?


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Re: [Asterisk-Users] IODBC instead of UNIXODBC

[pbx]

I would check your /etc/ld.so.conf file and make sure that you have the
library path for the IODBC libraries in there...

Then run ldconfig

and try reloading asterisk again.



> Hello.
>
> It's possible to use IODBC instead UNIXODBC with realtime?
> As I see, the res Makefile load a odbcinst.h file, but
> in IODBC there's not this file.
> I change the res Makefile (iodbcinst.h instead odbcinst.h)
> and the make create the res_odbc.so.
>
> But when asterisk boot it don't start showing:
>
> [res_odbc.so]Oct  4 10:24:43 WARNING[9748]: loader.c:314 __load_resource:
> libiodbc.so.2: cannot open shared object file: No such file or directory
> Oct  4 10:24:43 WARNING[9748]: loader.c:543 load_modules: Loading module
> res_odbc.so failed!
>
> There is something else I have do?
>
> Thanks.
>
> JS
>
>
>
>
>
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Re: [Asterisk-Users] Adding Cepstral to Asterisk

[pbx]

the app_cepstral.c file had a problem that it was trying use

#include "../asterisk.h"

I had to force it to where asterisk.h was located... in my case it was in
/usr/src/asterisk/include
so i changed the #include to say

#include "/usr/src/asterisk/include/asterisk.h" and then it would compile
through with no problems



> Wojciech Tryc wrote:
>
>> I am not following...
>> Why would you need to integrate Cepstral directly into Asterisk? Just
>> to be able to call it as Asterisk app from your dialplan? I am running
>> Cepstral and calling it through the System call.
>>
> You could try the howto located here:
>
> http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt
>
> for cepstral integration into asterisk. It makes it app_cepstral, instead
> of using system calls.
>
> Mat

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[Asterisk-Users] TDMoE help with Alarms...

[pbx]

I have configured TDMoE sucessfully and I am able to make a Zap connection
from one box to the other.

The question I have is..

I'm getting repeated errors every second on both systems..

Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 1: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 2: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 3: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 4: No Alarm
Oct  3 09:53:16 WARNING[4409]: chan_zap.c:6252 handle_init_event: Detected
alarm on channel 5: No Alarm
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 1
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 2
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 3
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 4
Oct  3 09:53:16 NOTICE[4409]: chan_zap.c:6247 handle_init_event: Alarm
cleared on channel 5


What is causing these errors?

When i do a zttool it shows that there are no errors...

Thanks...




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Re: [Asterisk-Users] Adding Cepstral to Asterisk

[pbx]

Then did you do a make clean / make / make install?

Then do "show applications" at the CLI prompt after you have restarted
asterisk.

"service asterisk stop"
"service asterisk start"

...

> I downloaded Cepstral to my Asterisk Box.  I did the install and let it
> install to /opt/swift.
>
> I brought down a new CVS-HEAD as of today 10/1.
>
> I added APPS+=app_cepstral.so into the Makefile in
> /usr/src/asterisk/apps/Makefile
>
> Like:
>
> # Obsolete things...
> #
> #APPS+=app_sql_postgres.so
> #APPS+=app_sql_odbc.so
> APPS+=app_cepstral.so
> #
>
> I did this piece but wasn't sure exactly what part of the Makefile I was
> to
> add it in so I added it in here:
>
> Towards the top of the file where it talks obsolete programs are commented
> out.
> And then after the section that compiles voicemail add:
>
> app_cepstral.so: app_cepstral.c
>   $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift
> -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include
>
> Make sure the $(CC) line starts with a tab, not spaces.
>
>
> I didn't see a lot about voicemail:
>
> app_sql_odbc.so: app_sql_odbc.o
> $(CC) $(SOLINK) -o $@ $< -lodbc
>
> app_cepstral.so: app_cepstral.c
> $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift
> -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include
>
> look:   look.c
> $(CC) -pipe -O6 -g look.c -o look -lncurses
>
>
> I checked the /etc/ld.so.conf file for a line like: opt/swift/lib in the
> file.  It wasn't there so I added it:
>
> include ld.so.conf.d/*.conf
> /opt/swift/lib
>
>
> I ran ldconfig when I was done.
>
> I can't see that Cepstral was added into Asterisk and I was wondering what
> I
> have done wrong that it doesn't work.
>
> Thanks.
>
>
>
>
>
>
>
>
>
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[Asterisk-Users] IAX2 Group dialing.... Is there something in the horizon?

[pbx]

Since the search engine on voip-info.org is not working correctly with old
links, etc..

I was curious if there is some hidden talent in the IAX2 outbound dialing?

What I'm asking about is:

Dial(IAX2/g1/${EXTEN})

Is there a way to set up groups like the above command using either SIP or
IAX2 protocols like you can do with Zap?

Thanks.



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Re: [Asterisk-Users] Asterisk - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?

[pbx]

YUP!!!

I guess working on this at 11:30pm at night, really need your wits about
you... Thanks... did that and it worked like a champ..

One other question I have the app_dtmftotext.c file is located at the
root of spandsp... however, when I had this file in my Makefile to
compile, and then have asterisk load it, asterisk complains on it.. Is it
really needed?

Thanks...



>
> I've been bitten by this before.  You've installed a newer version of
> SpanDSP over an older version.  Remove the spandsp libraries in the
> /usr/local/lib folder and re-install SpanDSP.
>
> Doug
>
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Re: [Asterisk-Users] Asterisk - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?

[pbx]

> On Sat, Oct 01, 2005 at 08:32:34AM -0700, [EMAIL PROTECTED] wrote:
>> I'm trying to put together a package of asterisk-head, spandsp, and
>> app_rx,tx fax.
>>
>> I can get everything to compile:
>>
>> spandsp-0.0.2pre20
>> asterisk-head (cvs co -r HEAD asterisk)
>> the app_rx/tx from soft-switch.org in the 1.1 folder
>>
>> However, asterisk complains that there is unused symbols when running
>> /usr/sbin/asterisk -vvvgc
>
> In which module? Are you sure it is not a left-over? (check dates, or
> book-keeping oof you package-management system, if you use one)


it's in the module app_rxfax.so, and if i comment out that one in
/etc/asterisk/modules.conf then it will compain about app_txfax.so.

ARGH

anyways...

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[Asterisk-Users] Asterisk - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?

[pbx]

I'm trying to put together a package of asterisk-head, spandsp, and
app_rx,tx fax.

I can get everything to compile:

spandsp-0.0.2pre20
asterisk-head (cvs co -r HEAD asterisk)
the app_rx/tx from soft-switch.org in the 1.1 folder

However, asterisk complains that there is unused symbols when running
/usr/sbin/asterisk -vvvgc

ARGH..

Does someone have a package with files that I could try?

I would greatly appreciate it.

Thanks.


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[Asterisk-Users] CALLERID to Sipura Devices (or others for that matter).. CVS-Latest Version

[pbx]

This is probably generic to any device... However..

Incoming callerid is working with number only.

If I try for any reason to use the function Set(Callerid(name)="blah
blah") it will then send only the outgoing extension as the callerid to
the phone that i s connected to the sipura device...

I also have a Sipura-841 that the same behaviour is happening.

I upgraded to the lastest Sipura 2000 firmware 3.1.15 and that did make no
difference...

But If I try to use the new functions i get the following errors:

-- Executing Set("SIP/2001-d523", "CallerID(number)=1231231230") in
new stack
Sep 25 10:15:56 ERROR[20733]: pbx.c:1380 ast_func_write: Function CallerID
not registered

I cannot use callerid_rewrite or any callerid mangling script at this point..

I have ugpraded to use and test the ODBC_Storage for voicemail - and that
is working great, but now If I go back I have to change a bunch of stuff
to go back to 1.0.9.

HELP...Is there anything that I can do to start using my scripts for
callerid_rewrites again?

Thanks.





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[Asterisk-Users] CVS-HEAD and Caller ID -- Pulling my hair out!

[pbx]

I have looked into this callerid problem now for a few hours.

1) Caller id on a sipura-2000 now shows:

cidname
2000

Where cidname is the new outputted formate from the cid_rewrite agi script
and 2000 is the exten number.

In looking at the Dial() application,

option "o"
  'o' -- Original (inbound) Caller*ID should be placed on the outbound
leg of the call
 instead of using the destination extension (old style
asterisk behavior)

I tried using this in my Dial string as an option but it wouldn't work, i
was getting the same Information..

Before I would have

Ben Cell
1234567890

appear on the sipura-2000 attached telephone.

The only thing i can get now is just

1234567890
2000 to appear on the phone

And when I try to use the new function set(CallerId(Name)="name") it says
that that function is not registered..

However when I use the SetCidName and SetCidNum i get this:

Sep 22 21:15:03 WARNING[25515]: app_setcidnum.c:72 setcallerid_exec:
SetCIDNum is deprecated, please use Set(CALLERID(number)=value) instead.
-- Executing SetCIDName("IAX2/66.234.228.170:4569-2", "xx") in
new stack
Sep 22 21:15:03 WARNING[25515]: app_setcidname.c:70 setcallerid_exec:
SetCIDName is deprecated, please use Set(CALLERID(name)=value) instead.


ARGH!  it's a never ending Circle

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Re: [Asterisk-Users] AGI Script to interact with ACCESS Databse and Set CID info on the fly.

[pbx]

ACCESS supports ODBC driven connections..



> Well guys here comes the fun part. I have a Microsoft access (VBA)
> application that interfaces with my SQL database. This app pulls of info
> from the SQL record and than picks up the phone and dials that locations
> number. I have purchased a few hundred NpaNxx's for my own use. I want get
> into too much detail there but no worries this is legal. I need to change
> my

> question is how do we get Access to speak to an AGI script. Has anyone
> done
> anything like this? Thanks a lot for reading but this will be a fun one.
>


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[Asterisk-Users] ODBC Voicemail WEB Retrieval V1.1

[pbx]

Hi All.

After some input, I created a V1.1 version of my ODBC VM retrieval from
the ODBC_Storage

It now uses either Mysql or unixODBC drivers to connect to the database

I didn't have php compiled with unixODBC so i had to recompile it in

"./configure --with-unixODBC --with-mysql --with-apxs2=./blah/blah/blah"
make
make install

after all that it worked.

see the Readme and changelog

http://www.itsngroup.com/software/asterisk/downloads/

The older version (1.0) only had MySQL support.

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[Asterisk-Users] ODBC VM Playback from Web Page

[pbx]

Hi all..

After some further research I have come up with a quick and dirty way to
playback the "longblob" recordings from the ODBC database for those of you
that are running the ODBC storage for voicemail.

Have a look

http://www.itsngroup.com/software/asterisk/downloads/ODBC_VM_1.0.tar

A little way that I can give back to such an awesome project

Ben..

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[Asterisk-Users] ODBC Voicemail WEB Retrieval

[pbx]

Ok.

I was sucessful in installing ODBC storage

I'm using MySQL in the backend as it is. but all my drivers are now ODBC.

I am running asterisk-cvs head as of last night 9/19/05

My question is this... the old voicemail.cgi script that allowed checking
voicemail no longer works etc, and never did work for me without a static
voicemail.conf file.

Anyways.. that aside... how does one retrieve the longblob object from the
database and present it to the user (upon authentication) via a website.

I'd be happy to help someone with the www/php/mysql integration but I just
dont know how to get blob's out and save to a temp file out of a database.

Thanks

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[Asterisk-Users] CallerID Num and Name setting to Asterisk.. Problem

[pbx]

I thought that I would try this on iax.conf as well however I still get

asterisk asterisk as the callerid name and num.

I have the latest CVS as of 8/17/05

Has anyone have this working with iax incoming?

Thanks
Ben

> That worked.  The following line also got rid of "asterisk" without
> entering any custom info:
>
> callerid=
>
> Thank you,
> Hugh
>
> On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
>> In the [default] section of sip.conf put:
>>
>> callerid=unavailable
>
>

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Re: [Asterisk-Users] Blank CIDName or CIDNum = "asterisk"

[pbx]

I thought that I would try this on iax.conf as well however I still get

asterisk asterisk as the callerid name and num.

I have the latest CVS as of 8/17/05

Has anyone have this working with iax incoming?

Thanks
Ben

> That worked.  The following line also got rid of "asterisk" without
> entering any custom info:
>
> callerid=
>
> Thank you,
> Hugh
>
> On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
>> In the [default] section of sip.conf put:
>>
>> callerid=unavailable
>
>

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[Asterisk-Users] Toll Call Voicemail Ring Timeout (new module????)

[pbx]

Remember in the good ol days when answering machines were smart enough to
know when there was a message on the machine, and it would pick up after 2
rings rather than 4? (that is, if you knew how to turn it on - that
required to know how to set the time on your VCR to avoid the flashing
12:00:00)

Hahaha. Jokes aside.

I have come up with a way to do this but it's a kludge:

1 - Read in the variables (normal_ring_timeout)
2 - HasVoiceMail_Timeout
3 - run HasVoicemail
4 - If not have voicemail set ring_timeout = normal_ring_timeout
5 - If Have voicemail - it's jumps to priority + 101
6 - Set ring_timeout = hasvoicemail_timeout
7 - Ring extensions for ring_timeout
8 - blah blah blah...

I was wondering if someone had come up with their own way / better way /
maybe custom app for something like this.

Thanks

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Re: [Asterisk-Users] follow me configuration web page??

[pbx]

I would like to do the same thing, and the easiest would be to use MySql
and a web connector :..

I can help.

Ben


> Does anybody have an example follow-me configuration web page code
> written in either php or perl that can write out the follow-me config
> into the asterisk files?
>
> I'd like to setup something on our office voip server that I can
> change as needed via a web page rather than writing the script by
> hand.
>


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Re: [Asterisk-Users] Blank CIDName or CIDNum = "asterisk"

[pbx]

I thought that I would try this on iax.conf as well however I still get

asterisk asterisk as the callerid name and num.

I have the latest CVS as of 8/17/05

Has anyone have this working with iax incoming?

Thanks
Ben

> That worked.  The following line also got rid of "asterisk" without
> entering any custom info:
>
> callerid=
>
> Thank you,
> Hugh
>
> On Thu, 2005-08-11 at 03:19 +0100, Tony Hoyle wrote:
>> In the [default] section of sip.conf put:
>>
>> callerid=unavailable
>
>

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Re: [Asterisk-Users] New digium TE406 & 411

[pbx]

We will start  installing TE411 next week, I'll keep the list informed !
jack


Eric Rees wrote:



Has anyone on the list tried one of these new cards with built-in echo
cancellation?


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[Asterisk-Users] CallerID rewrite php AGI Script

[pbx]

Hello all.

I am looking for the great callerID rewrite script that does the 411
lookup and then stores the information in a database.

If there is information in the Database for the callerid coming in, then
use that and pass it along to the phone.

I lost my entire system hard drive this week, and slowly rebuilding. This
script wasn't in the most recent backup :( :( :(

Please help :)

thanks.

Ben

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Re: [Asterisk-Users] Using 2 x DSL

[VoIP-PBX]

Bernard Cresencia wrote:


Hi Henry,
 
Saw your question on the lists - you can get a name-brand (Symantec) 
for  around CAD$1000 - see the following:
http://www.symantecstore.com/dr/sat1/ec_MAIN.Entry17c?CID=0&SID=49998&SP=10007&PN=5&PID=645999&CUR=124&DSP=&PGRP=0&CACHE_ID=0 
<http://www.symantecstore.com/dr/sat1/ec_MAIN.Entry17c?CID=0&SID=49998&SP=10007&PN=5&PID=645999&CUR=124&DSP=&PGRP=0&CACHE_ID=0>
 
The good thing about it is that it will do load balancing and 
bandwidth aggregation. It also has VPN built in, and they're currently 
running a promo that throws in a Wireless B/G card to turn it into a 
wireless access point (worth CAD$152)


Or if that's too expensive, there is a much cheaper alternative:
http://www.xincom.com/twr502.html which goes for just over CAD$250. 
It's similar to the Symantec but does not have the name brand.
 
Personally, I haven't heard much about these but it should perform as 
advertised.
 
Best regards,

Bernard
*/VoIP-PBX <[EMAIL PROTECTED]>/* wrote:

Hi all, my client wants to double his bandwidth by using 2 x DSL
lines
into one Asterisk network
How can I do this ?

Thanks

Henry
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Thanks Bernard, I should have ask you  first, even the cheap dual WAN 
boxes seem to load balance

and compared to the cost of a T1 this is a very good "bang for his buck".

Thanks again  


TTFN Henry




 


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[Asterisk-Users] Using 2 x DSL

[VoIP-PBX]

Hi all, my client wants to double his bandwidth by using 2 x DSL lines 
into one Asterisk network

How can I do this ?

Thanks

Henry
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Re: [Asterisk-Users] rxfax not answering

[pbx]

This is not the problem

There is an example for faxing. you should set some other parms. ie.
unique filename etc., you also need permissions etc.

Look at NVBackground detect from Newman Telecom. Use the wiki and search
for NVFaxDetect

HTH

> Hello i would like to receive incoming faxes thru' asterisk as tiff
> files thru' the rxfax application.
>
> I setup extensions 101 like this
>   exten=> 101,1,rxfax(/tmp/fax.tif)
>
> then from CLI i run:
>   dial 101
> and rxfax send me his "scream" about the fax ^^
>
> instead when i send a real fax from a faxmachine to that extension
> my 101+rxfax is executed but it just "does nothing"
>
> the call is originated by a FAX on PSTN and received via VoIP by
> asterisk using a/u law codec
>
> i think that is my VoIP provider that has some fax problem.
>
> Is this the problem or there maybe other solutions?
>
> Thank you, Antonio
>
>
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[Asterisk-Users] RXFax and Hangup context Question.

[pbx]

Hello:

I have been using the asterisk system now for almost 5 months. I'm very
happy with it's performance, however I have been using the RXFAX on and
off for the last month or so. I gave up after a while and just had it
route to my analog fax machine in the fax context.

However, I have always had the implementation that it would go to a
macro.. macro-faxreceive, and to the rxfax of the tif file etc etc etc.

Then, at the hangup context, it would run the system command to actually
email the TIF file.

One thing though, when the line hangs up it executes the hangup from the
original source of the call. If the original source of the call was an IVR
menu, it would execute the hangup context for that rather than the hangup
context for the macro.
My question is, is this normal behaviour that it executes the hangup
context of the first context of the call, and not the hangup context of
the current context that the call is executing through?

When a fax is received, it "goto's" the fax context, as described by the
CLI output, but when the hangup occurs, it never executes the system
command?

[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten => s,2,SetVar(EXTEMAIL=${FAXTOEMAIL})
exten => s,3,SetVar(EXTNAME=YourName)
exten => s,4,SetVar(EXTCOMPANY=YourCompany)
exten => s,5,RxFax(${FAXFILE}.tif)
exten => h,1,System(/usr/local/bin/faxmail "${CALLERIDNUM}" "${CALLEDFAX}"
"${EXTNAME}" "${EXTEMAIL}" "${FAXFILE}" "${EXTCOMPANY}")

This is the example that i'm following.

I have in the original context (where the call comes in)
context = fax, priority 1, macro-faxreceive

should i just create  a generic context [faxreceive]
and do a GOTO instead of running the macro? Does a macro jump out when the
hangup is done? is that why this is happening?

thanks...



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[Asterisk-Users] Caller ID Routing using VoicePulseConnect

[pbx]

I have a question for those of you out there using VoicePulseConnect for
incoming did

I have in my Realtime extensions Database
(the x's are replaced with my phone number)
context = voicepulse-in-01
exten = xx/
Priority=1
app=NoOp
appdata = Incoming call with no callerid on xx

However it never triggers

I also tried using one of my other providers (voipjet for outbound) and
calling myself - i set the outbound callerid to nothing and it defaults to
202556

so i tried the above with the difference in
exten = xx/202556
and this example doesn't match either!
This is the only inbound iax provider that i have.

This is the way it's supposed to work? correct? as i have read on the wiki.

Thanks


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RE: [Asterisk-Users] Realtime+IAX2 and RSA

[pbx]

I have found that in the iax_buddies database, if a field doesn't exist...
then just create it :) then it works.

The schemas have been outdated it seems from the "create table blah blah
blah"

so if it is in the IAX.conf file, and it's not in the iax_buddies table
structure, add the field.

Tada!


>>-- Messaggio Originale --
>>Date: Thu, 2 Jun 2005 03:35:26 +0200
>>From: [EMAIL PROTECTED]
>>To: asterisk-users@lists.digium.com
>>Subject: [Asterisk-Users] Realtime+IAX2 and RSA
>>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>>
>>
>>Anyone had Realtime working with IAX2 and RSA authentication to connect
> two
>>PBXs, please? It seems that inkeys/outkey fields are not read at all and
>>the following warning is logged when dialing:
>>
>>Jun  2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to
>> authenticate
>> to XXX.XXX.XXX.XXX
>>
>>Using iax.conf it perfectly works. Maybe a bug in Realtime?
>>
>>TIA,
>>
>>Alex
>
>
> Maybe it will be useful to someone else:
> http://bugs.digium.com/view.php?id=4431
>
> Cheers,
>
> Alex
>
> __
> TISCALI ADSL 1.25 MEGA a soli 19.95 euro/mese
> Solo con Tiscali Adsl navighi senza limiti di tempo
> a meno di 20 euro al mese e in piu' telefoni senza
> pagare il canone Telecom. Scopri come
> http://abbonati.tiscali.it/adsl/sa/1e25flat_tc/
>
>
>
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Re: [Asterisk-Users] Re: Asterisk Box as a Router, Firewall and DHCP Server

[pbx]

Ditto That:)

Thats what i use!
>
> What's wrong with :-
>
>host W2K {
> hardware ethernet   00:30:1B:AC:39:E3;
> fixed-address   192.168.1.130;
> }
>
> this box always gets the same IP and I know who's got what.
>


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Re: [Asterisk-Users] A Way to Write DTMF Digits as text to CDR?

[pbx]

You could simply put something in your dialplan / ivr that allows for
mysql insert into the cdr..



> I've gotten my CDR working the way I like, but I am looking to customize
> it a bit.  I have set up an IVR menu, which works great.  I would like to
> be able to capture the prompted DTMF digits pressed by callers, to my CDR
> database but I don't see any AGI or Asterisk commands that allow one to
> customize the CDR contents.  Am I thinking about this on the wrong track?
> If someone calls sales for instance, and presses 44364 for their PO number
> when prompted, I just want to have a text record of the digits they
> pressed in my CDR so I can easily view it.  No trying to do database
> lookups or screen pops from it or anything fancy, I'm trying to eat an
> elephant one bite at a time.  Anyone have a solution for that?
>
> I hope I'm not being a pest by asking a question every other day, but the
> responses I've gotten have been very helpful.  I'm trying to learn as much
> as I can from the array of documentation, and I swear I'm only asking when
> I feel like I've exhausted what I could find.
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RE: [Asterisk-Users] Sipura 3000 - fax passthrough?

[pbx]

I'm not sure if the Sipura3k allows for that. Somewhere along the line,
the line has to be answered...

and if you have the same codec allow= in your sip.conf for both
extensions, * wont transcode the stream, it would do a passthru.

I have the reverse as well, i have the fax machine do the dialplan through
asterisk, however the context for the fax only sends it back out the
sipura3k box Dial(SIP/192.168.1.254:5061/blahblah)

And it allows for CDR.

if the internal fax machine is busy on an incoming fax call, then i go to
rxfax as a backup.

Ben

> But that requires asterisk handles the call, it has to be g711 encoded,
> and
> all of that mess. What I want is a fax call is directly passed through to
> the fax machine, and voice is send to Asterisk via VOIP.
>
> Chris Mason
> www.anguillaguide.com
> Tel:  (305) 704-7249 Fax: (815)301-9759
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> [EMAIL PROTECTED]
>> Sent: Tuesday, May 31, 2005 6:16 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] Sipura 3000 - fax passthrough?
>>
>> What do you want to do if they aren't fax's? dial a DID or something?
>>
>> I have the similar thing... I have both ports. the FXO, and
>> FXS registering with my asterisk server..
>>
>> Then, the default dialplan is 
>>
>> so it will dial immediately 123456789 in my extensions.conf
>>
>> 123456789 answers the line
>> and then it runs NVFaxDectect from newman telecom, and if
>> it's a fax, it jumps to the fax priority and i then dial
>> ${FAX} where FAX=SIP/> fax>
>>
>> and walla, the fax machine answers and all is well.
>>
>>
>>
>> > I have installed two Sipura 3000's on my office pbx as a test, they
>> > work well an have some great features including fax detect,
>> but I was
>> > hoping to allow incoming faxes on the FXO port to be detected and
>> > passed through to the FXS port. Am I mistaken or does this
>> work, and how?
>> >
>> > Chris Mason
>> > Anguilla
>> >
>> > ___
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Re: [Asterisk-Users] Sipura 3000 - fax passthrough?

[pbx]

What do you want to do if they aren't fax's? dial a DID or something?

I have the similar thing... I have both ports. the FXO, and FXS
registering with my asterisk server..

Then, the default dialplan is 

so it will dial immediately 123456789 in my extensions.conf

123456789 answers the line
and then it runs NVFaxDectect from newman telecom, and if it's a fax, it
jumps to the fax priority and i then dial ${FAX} where FAX=SIP/

and walla, the fax machine answers and all is well.



> I have installed two Sipura 3000's on my office pbx as a test, they work
> well an have some great features including fax detect, but I was hoping to
> allow incoming faxes on the FXO port to be detected and passed through to
> the FXS port. Am I mistaken or does this work, and how?
>
> Chris Mason
> Anguilla
>
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Re: [Asterisk-Users] ISO Suggestions for Multiple Inbound Voicepulse Lines

[pbx]

Easy:

I just had to do the same thing recently.

in your inbound context of [voicepulse-in-01]
you have something like this in your extensions.conf

exten => 1234567890,1,NoOp(incoming call on ${EXTEN})
exten => 1234567890,2,Dial(blah blah blah)
exten => 9876543210,1,NoOp(IncomingCall on ${EXTEN})
exten => 9876543210,2,Dial(halb halb halb)

etc etc tc...
i did this .. as i'm waiting for my 2nd number to port over to voicepulse...

Ben

> I'm looking to set up multiple inbound Voicepulse Connect lines and have
> Asterisk route them direct to different IVR or Voicemail based on the
> inbound number that is called.  Unfortunately, I just can't see how one
> would go about identifying the number that is being called.  Has anyone
> been able to do something like this with Voicepulse?
>
> I appreciate any assistance.
>
> Phil
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Re: [Asterisk-Users] ISO Suggestions for Multiple Inbound Voicepulse Lines

[pbx]

Easy:

I just had to do the same thing recently.

in your inbound context of [voicepulse-in-01]
you have something like this in your extensions.conf

exten => 1234567890,1,NoOp(incoming call on ${EXTEN})
exten => 1234567890,2,Dial(blah blah blah)
exten => 9876543210,1,NoOp(IncomingCall on ${EXTEN})
exten => 9876543210,2,Dial(halb halb halb)

etc etc tc...
i did this .. as i'm waiting for my 2nd number to port over to voicepulse...

Ben

> I'm looking to set up multiple inbound Voicepulse Connect lines and have
> Asterisk route them direct to different IVR or Voicemail based on the
> inbound number that is called.  Unfortunately, I just can't see how one
> would go about identifying the number that is being called.  Has anyone
> been able to do something like this with Voicepulse?
>
> I appreciate any assistance.
>
> Phil
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[Asterisk-Users] How to timeout using AGI.

[pbx]

How does one process / capture a timeout that has happened in using an AGI
script.. Preferably PHP.

I know you can set the wait timeout for a certain time, but how does the
script continue?

Thanks

Ben

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Re: [Asterisk-Users] spandsp issue

[pbx]

The patch is not that much.

I just opened it up - and then typed in the patch information into the
Makefile manually.

I have to do this for another application, so it wasn't that hard to do.

Even if i tried to "copy / paste" it would complain (the make process) on
the app_txfax.so : app_txfax.c
 $solink blah blah blah

seems it needs a TAB on the next line, and a simple cut and paste doesn't
preserve that, so it compains about missing a separator..

so i just typed it in manually - Power of VI :)

HTH

Ben

> I'm trying to compile spandsp and Asterisk but the patch makefile hunks
> don't even resemble any parts of the asterisk Makefile. I am using the
> latest version of both. Has anyone else run into this problem?
>
> -Mark
>
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RE: [Asterisk-Users] LiveVOIP

[pbx]

I have an open ticket with them, and I have had better quality now that
they saw something wrong with my account on their end.

Anyways, the next question is in iax_peers / iax_buddies /
iax_friends, etc etc in realtime.

In the Codec order (allow) i have "g729;ulaw;ilbc;gsm"
However, LiveVOIP only uses Ulaw if ulaw is anywhere in the codec prefs.

You can see in the IAX2 debug below the problem. However, if i only
specify g729 then g729 is used.

I have the ticket open with the trouble shooter ticket.

I can say though that since they made some change the my calls have been
going through much nicer.

I dont use the IP address only . i use a context in my iax_buddies to
control the codecs.

 Timestamp: 670008ms  SCall: 00119  DCall: 1 [216.118.117.46:4569]
-- Executing NoOp("SIP/3000-fe9c", "Making outbound long-distance call
xx")
-- Executing SetCallerID("SIP/3000-fe9c", "xx")
-- Executing Dial("SIP/3000-fe9c",
"IAX2/user:[EMAIL PROTECTED]/xxx")
-- doing lookup for '217.160.244.186'
-- doing lookup for '217.160.244.186'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
   Timestamp: 8ms  SCall: 5  DCall: 0 [217.160.244.186:4569]
   VERSION : 2
   CALLED NUMBER   : xx
   CODEC_PREFS : (g729|gsm|ulaw)
   CALLING NUMBER  : xx
   CALLING PRESNTN : 0
   CALLING TYPEOFN : 0
   CALLING TRANSIT : 0
   LANGUAGE: en
   USERNAME: user
   FORMAT  : 256
   CAPABILITY  : 63750
   ADSICPE : 2
   DATE TIME   : 2005-05-23  18:49:22

-- Called user:[EMAIL PROTECTED]/xx
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
   Timestamp: 1ms  SCall: 00077  DCall: 5 [217.160.244.186:4569]
   AUTHMETHODS : 3
   CHALLENGE   : 184957130
   USERNAME: user

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
AUTHREP
   Timestamp: 00101ms  SCall: 5  DCall: 00077 [217.160.244.186:4569]
   MD5 RESULT  : cf1ba900bacc277bcb85f6e5785e4f39

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT
   Timestamp: 00092ms  SCall: 00077  DCall: 5 [217.160.244.186:4569]
   FORMAT  : 4

-- Call accepted by 217.160.244.186 (format ulaw)
-- Format for call is ulaw
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
   Timestamp: 00092ms  SCall: 5  DCall: 00077 [217.160.244.186:4569]
Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: HANGUP
   Timestamp: 01211ms  SCall: 5  DCall: 00077 [217.160.244.186:4569]
   CAUSE CODE  : 0

-- Hungup 'IAX2/livevoip-out-5'
  == Spawn extension (home, xxx, 3) exited non-zero on
'SIP/3000-fe9c'

> Agreed, takes forever for calls to go through, they say they can get DIDs
> in
> a few business days, they can't.  Their quality has gone down hill fast.
> They talk a lot but can't deliver in my opinion.
>
>

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[Asterisk-Users] LiveVOIP

[pbx]

Is anyone having problems with LiveVOIP for outbound calls since their
network upgrade a week ago?

Ever since the network upgrade, it takes 2-3 times MINIMUM in order for a
call to go from my system to theirs.

I haven't changed any configs on my side, it just says "

"call accepted by blah blah blah"
and stays there for about 25 seconds, then comes back and says no one is
available to answer at this time.

When  a call does go through, it gives back the message "call is making
progress blah blah blah"



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RE: [Asterisk-Users] CallerID name lookup AGI script

[pbx]

That was it!
Told me exactly what i needed.

in the php file it's looking for /usr/bin/php
mine was in /usr/local/bin/php

ARGH! so simple

Thanks for that though!!

it's working Great!!!


> Run ./cid_rewrite.php from the the shell to see where it's failing.
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
>> Sent: Friday, May 20, 2005 10:04 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: RE: [Asterisk-Users] CallerID name lookup AGI script
>>
>>
>> I have been trying to get this script to work as well cid_rewrite
>>
>> However this is what the CLI reports:
>> -- Executing EAGI("Zap/1-1", "cid_rewrite/cid_rewrite.php|us")
>> -- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php
>> AGI Tx >> agi_request: cid_rewrite.php
>> AGI Tx >> agi_channel: Zap/1-1
>> AGI Tx >> agi_language: en
>> AGI Tx >> agi_type: Zap
>> AGI Tx >> agi_uniqueid: 1116600703.79
>> AGI Tx >> agi_callerid: 619xxx  <-- it is getting caller
>> id number AGI Tx >> agi_calleridname: unknown AGI Tx >>
>> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >>
>> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >>
>> agi_dnid: unknown AGI Tx >> agi_rdnis: unknown AGI Tx >>
>> agi_context: incoming-zap AGI Tx >> agi_extension: s AGI Tx
>> >> agi_priority: 2 AGI Tx >> agi_enhanced: 1.0 AGI Tx >>
>> agi_accountcode: AGI Tx >>
>> -- AGI Script cid_rewrite.php completed, returning 0
>> -- Executing Dial("Zap/1-1", "IAX2/4000|20|rtT")
>> -- Called 4000
>> -- Call accepted by 129.46.90.210 (format gsm)
>> -- Format for call is gsm
>>
>> However, it does not have any information being returned.
>>
>> I have edited the agi_config.php to point to where the
>> information is. But it just is not getting anything.
>>
>> Help???...
>>
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RE: [Asterisk-Users] CallerID name lookup AGI script

[pbx]

I have been trying to get this script to work as well
cid_rewrite

However this is what the CLI reports:
-- Executing EAGI("Zap/1-1", "cid_rewrite/cid_rewrite.php|us")
-- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php
AGI Tx >> agi_request: cid_rewrite.php
AGI Tx >> agi_channel: Zap/1-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1116600703.79
AGI Tx >> agi_callerid: 619xxx  <-- it is getting caller id number
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: incoming-zap
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 1.0
AGI Tx >> agi_accountcode:
AGI Tx >>
-- AGI Script cid_rewrite.php completed, returning 0
-- Executing Dial("Zap/1-1", "IAX2/4000|20|rtT")
-- Called 4000
-- Call accepted by 129.46.90.210 (format gsm)
-- Format for call is gsm

However, it does not have any information being returned.

I have edited the agi_config.php to point to where the information is. But
it just is not getting anything.

Help???...

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RE: [Asterisk-Users] GOTO statement in Realtime-Extensions not workinglike expected

[pbx]

I was just going to ask this same question

Is this the normal behavior that you have to do, jump back to the .conf file?

It is how I have it configured, but it's more a "hybrid" than a true
realtime system.

Thanks

> Use the Goto statement with '|' instead of ','. And make tables for each
> context you have in the extensions.conf file.
>   One thing I noticed is using Goto in real time extensions causes
> the jump back to the extensions.conf file.
>   So first jump to extensions.conf and then specify another switch
> statement. But make  a new table for each context in the extensions.conf.
>
>
>
>
>
> Regards:
> Bharat M. Sarvan
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Thursday, May 19, 2005 3:24 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] GOTO statement in Realtime-Extensions not
> workinglike expected
>
> Hi .. When I use the GoTo statement in realtime to goto a priority only
> ... E.g. Goto(3) then there's no problem
>
> But, If I try to jump to another context ... E.g.
> Goto(othercontext,${EXTEN},3) then it doesn't work
>
> If I process the same statement in extensions.conf things go well
>
> Are there things broken regarding GoTo in combination with Realtime
> Extensions ?
>
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[Asterisk-Users] MeetMe -1 return Code - howto

[pbx]

I was searching for help on how to handle the errors that are returned
from the MeetMe application.

for instance.

1) if a user tries to join a conference that is locked, allison says that
the conference is locked and then comes back to the dialplan, however it
does not continue down the dialplan.

I have a meetme command on Priority 8, and the CLI says that it returned
non zero (as the wiki states it would be -1).

I tried to use 109 priority thinking it is 101+n, but i cannot have it go
back in time to priority 7, becaue it would just loop.

How does one handle this?

When i'm in a conference, and I press the pound key, it does exit out
nicely and continue down the dialplan, i have some NoOp on the next
priority after the MeetMe command and it flows nicely from there.

   -- Accepting a maximum of 4 digits.
-- Playing 'conf-getconfno' (language 'en')
-- User entered '1234'
-- Executing NoOp("SIP/2001-e4e7", "user entered conf no: 1234")
-- Executing MeetMe("SIP/2001-e4e7", "1234|p")
-- Playing 'conf-locked' (language 'en')
  == Spawn extension (ivr-join-conf, s, 8) exited non-zero on 'SIP/2001-e4e7'

I also get the error if they enter a conf # that is not configured.
Allison says that that is an invalid conference number, please try again,
and the same error:

-- Accepting a maximum of 4 digits.
-- Playing 'conf-getconfno' (language 'en')
-- User entered '5678'
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-4", "user entered conf
no: 5678")
-- Executing MeetMe("IAX2/[EMAIL PROTECTED]:4569-4", "5678|p")
  == Parsing '/etc/asterisk/meetme.conf': Found
-- Playing 'conf-invalid' (language 'en')
  == Spawn extension (ivr-join-conf, s, 8) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569-4'
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569-4'


However my dialplan has an 8 and a 109 priority both are NoOp.. and they
are never triggered.

Thanks for some help on this in advance...




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Re: [Asterisk-Users] Mysql cmd with Asterisk Problems

[pbx]

The information that is below is from the CLI window.
the # for ${resultid} only appears if you using the .conf file. If you
realtime with mysql resultid has no value and thus is blank.

I have copied and pasted exact same line from the .conf file to the
appdata field in mysql and i get the same result. the ${resultid}
reference is on the fetch line, but it doesnt have a value!


> [EMAIL PROTECTED] wrote:
>
>> -- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-13", "Fetch
>> fetchid ivr_password")
>
>> -- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-15", "Fetch
>> fetchid 6 ivr_password") in new stack
>
>
> I don't see the "6" in the realtime extensions debug. Did you forget
> the
> ${} reference?
>
> -Matthew
>
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Re: [Asterisk-Users] Mysql cmd with Asterisk Problems

[pbx]

Hi.. .and thank you for your help.

I just tried your example.. and yes it did return what i wanted (with my
information)

One thing I need to add here i'm using mysql-realtime configuration.
So when I'm running using mysql extensions I get this:

-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-13", "Connect connid
localhost dbuser dbpass dbname")
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-13", "Query resultid
5 SELECT ivr_password FROM users WHERE ivr-id = 1234")
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-13", "Fetch fetchid 
ivr_password")
May 18 10:59:43 WARNING[12640]: app_addon_sql_mysql.c:113 find_identifier:
Identifier 0, identifier_type 2 not found in identifier list
May 18 10:59:43 WARNING[12640]: app_addon_sql_mysql.c:328 aMYSQL_fetch:
aMYSQL_fetch: Invalid result identifier 0 passed
  == Spawn extension (macro-mmisd-login, s, 9) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569-13' in macro 'mmisd-login'
  == Spawn extension (ivr-mmisd-login, 1, 12) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569-13'

When I use extensions.conf  hard coded i get:
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-15", "Query resultid
5 SELECT ivr_password FROM users where ivr_id=1234") in new stack
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-15", "Fetch fetchid 6
ivr_password") in new stack
May 18 10:57:41 WARNING[12640]: app_addon_sql_mysql.c:316 aMYSQL_fetch:
ast_MYSQL_fetch: numFields=1
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-15", "Clear 6") in
new stack
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-15", "Disconnect 5")
in new stack
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-15", "ivr_password =
") in new stack

It works..

So how do I make this work in using mysql extensions!???

Thanks



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[Asterisk-Users] RTFriendsCache=yes help Voicemail MWI help

[pbx]

A while back I converted back to static conf files from a database setup.

However I decided to tackle it again.

The problem that I was experiencing, was, there was no stutter tone on my
sipura 2000 or 3000 when there was a voicemail left at either extension
when I was using mysql setup for peers and voicemail.

I have 2 contexts... home, office in my voicemail configuration
I now use VoicemailMain([EMAIL PROTECTED]) context being office, or home.. and
that all works.

It's just the stutter tone that does not work.

Someone suggested, and i also found in the wiki to place
rtfriendscache=yes in the sip.conf file, and I have tried that as well.
Still no avail.

In the mysql record for the registration i have in the mailbox field i.e
[EMAIL PROTECTED] (the sipura extension is 2000) and the context is home. but
still no stutter tone.

If I use the static voicemail.conf file and sip.conf file the stutter tone
works.

Thanks in advance

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[Asterisk-Users] Mysql cmd with Asterisk Problems

[pbx]

Hello all:
I am trying to use the mysql command to retrieve information from a mysql
database.
my example here was formed from using the wiki reference to using the
mysql command.

The problem is with the fetch command.
Here is the macro code:
Mysql(QueryString=SELECT\ ivr-password\ from\ users\ where\ ivr-id=${userid})
Mysql(Query r ${connid} ${QueryString})
Mysql(Fetch fetchid ${r} dbuserpass)
Mysql(Clear ${resultid})
Mysql(Disconnect ${connid})

However, it never gets past the fetch line. and ${r} is not showing
anything either from che CLI window.
I usesd the mysqlasteri web page to make the command escape character
happy, etc. I have tried putting \' around each item, etc. However The
same problem comes back with the fetch line. I have tried to use
mysql(fetch fetchid ${r} ivr-password) thinking the variable that is
coming out of the DB has to be named the same, etc. but it doesn't matter.
${r} is blank in the fetch command when I know there is a valid record.
The ${connid} has a value in it as it's being passed.

I have only been able to find the mysql cmd example on the wiki, and no
other. I program mysql with php all the time, but i dont understand the
errors that it is returning..
Thank you.

Output is below:

-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "")
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "")
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "Authenticate user
now: userid: 1234 - pass: ")
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "Passing to
macro-mmisd-login")
-- Executing Macro("IAX2/[EMAIL PROTECTED]:4569-1",
"mmisd-login|1234|")
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "Entered
Macro-mmisd-login")
-- Executing NoOp("IAX2/[EMAIL PROTECTED]:4569-1", "Passed in userid:
1234 - userpass: ")
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-1", "Connect connid
localhost mmisd-ivr-user mmisd-ivr-pass ivr-db")
-- Executing SetVar("IAX2/[EMAIL PROTECTED]:4569-1",
"QueryString=SELECT\ ivr-password\ from\ users\ where\ ivr-id=1234")
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-1", "Query r 18
SELECT\ ivr-password\ from\ users\ where\ ivr-id=1234")
-- Executing MYSQL("IAX2/[EMAIL PROTECTED]:4569-1", "Fetch fetchid 
dbuserpass")
May 18 07:01:05 WARNING[7114]: app_addon_sql_mysql.c:113 find_identifier:
Identifier 0, identifier_type 2 not found in identifier list
May 18 07:01:05 WARNING[7114]: app_addon_sql_mysql.c:328 aMYSQL_fetch:
aMYSQL_fetch: Invalid result identifier 0 passed


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Re: [Asterisk-Users] BYOD provider other than broadvoice

[pbx]

There are plenty on the wiki...

> Are there any BYOD providers out that that people have had positive
> experiences with? I have broadvoice and they suck lately.  Anyony have
> anyone with a good amount of peers and not a lot of downtime?
> --
> Michael Lyszczek
> New York, NY, 10282
> NEW EMAIL : [EMAIL PROTECTED]
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Re: [Asterisk-Users] Unable to specify channel 1: No such device

[pbx]

It looks like from the ztcfg output that you are trying to reference port
3, which is if your looking at the card with modules up. the one on the
far right.

Move the module to the far left... and that will be channel 1

The far left, is the TOP port on the card.

HIH


> Hi, I did not find any useful information to configure a Wildcard
> TDM400P with a FXO card. I've tried everithing, I tried configure it
> using the cvs and the information from digium page, I tried to
> configure it using
> debian packages, I tried to configure with kernels 2.4.30 and 2.6.11, I
> even
> switched the mother board (I tried 3 motherboards).
>


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[Asterisk-Users] sipura ATA, VMI, and Realtime

[pbx]

I have had issues with using * in Realtime for dialplan, voicemail,
sip_buddies, sipfriends, etc... sip this and that.

The problem that I have is the VMI doesn't work on my Sipura ata's.. I
have a 1001, 2000, and 3000.

The VMI only works when I configure the sip registration in the sip.conf
file.
(i.e. mailbox=).

In the database (mysql) - i have also set the mailbox =  and the phone
never gets a VMI.

Anyone else have this problem...

I have the Sipuras set up for VMI when New VM arrives. The VM Available
setting drives me nutz when it short rings the phone every 30 min on new
VM.

Thanks...

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[Asterisk-Users] Linux & Asterisk

[Asterisk Pbx]

Hi all,

I am thinking in implementing asterisk into my buisness. I heard all
sorts of good things about it. The question im asking my self is what
linux distribution is best to use? Do you know what distribution they
use for their asterisk training?

Thanks for reading me.

PBXER
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Re: [Asterisk-Users] TE110P experiance

[pbx]

[EMAIL PROTECTED] wrote:

Hello to all,
I would like to ask some Digium TE110P users if they can share experiance
about this card. I put in service card yesterday but I noticed following
(strange) behaviar:
- if I have to reboot my computer my zaptel driver fail to start and
produce this error:
 ZT_SPANCONFIG failed on span 1: No such device or address (6)
- to solve this problem I have to power cycle my computer and in all cases
this brings up card!
- does anybody have any info about this hardware, example there are two LED
- what is the meaning of these LEDs. I bought this card and got anly card
without any papers (just bill :-( )
Regards,
[EMAIL PROTECTED]
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This is a well known bug ( can't remeber the number) but
the Card Identification  on the  bus PCI change (for some reason) and 
the driver is not able to find the card anymore,
try to add the line marked with a +  in the wcte11cxp.c (in zaptel 
source), and recompile your driver.

If you like tio verify the bug
start your your system from power down
do a  lspci  -v  locate "Tiger Jet Network "
look the id
load zaptel driver
do lspci -v
ans see the difference
Hope this help

static struct pci_device_id t1xxp_pci_tbl[] = {
   { 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long) 
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long) 
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long) 
"Digium Wildcard TE110P T1/E1 Board" },
   { 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long) 
"Digium Wildcard TE110P T1/E1 Board" },
+ { 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long) 
"Digium Wildcard TE110P T1/E1 Board" },
   { 0 }
};



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Re: [Asterisk-Users] TE410P problem (Looping UP Span 1...)

[pbx]

Scott Stingel wrote:
Sid-
Try connecting one port to another.  Note that one of the ports must 
be set up as "cpe" and the other as "net" in zapata.conf when you loop 
them together like this.

A suitable crossover cable for testing can be constructed by cutting 
up a CAT 5 cable, and connecting:
Pin 1 <--> Pin 4 on the other end
Pin 2 <--> Pin 5
Pin 4 <--> Pin 1
Pin 5 <--> Pin 2

You should get green's on both the connected channels if your zaptel 
and zapata configurations are ok, and if you've run both modprobe and 
ztcfg as documented.

Good luck
Scott Stingel
President
EVT, Inc.
www.evtmedia.com

Sid wrote:
Hi list,
 
We have been trying to configure a Quad Span T1 card in a system 
running RH9. We have followed the instructions in the Wiki and 
searched the mailing lists, but so far havent got any success. Cable 
is connected to the first span, and module is loaded. Without loading 
the module the LED glows in red colour, but the moment we load 
module, it goes off. (No red or green) .
 
We ran zttool and tried to run a loop test, but zttool simply hung 
with the message 'Looping UP Span 1...'. We had to terminate zttool 
with 'kill'.  Here is the output of the lsmod command. Can someone 
shed some light on this?
 
Thanks,
-Sid
 
Module  Size  Used byNot tainted
wcusb  20128   0  (unused)
wct4xxp54272   0  (unused)
zaptel182432   0  [wcusb wct4xxp]
 
tail -f /var/log/messages
Jan  6 14:54:32 localhost kernel: TE410P: Launching card: 0
Jan  6 14:54:32 localhost kernel: TE410P: Setting up global serial 
parameters
Jan  6 14:54:32 localhost kernel: Found a Wildcard: Wildcard 
TE410P-Xilinx
Jan  6 14:54:32 localhost kernel: usb.c: registered new driver wcusb
Jan  6 14:54:32 localhost kernel: Wildcard USB FXS Interface driver 
registered
Jan  6 14:54:33 localhost kernel: Registered tone zone 0 (United 
States / North America)
Jan  6 14:54:33 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS
Jan  6 14:54:33 localhost zaptel: Running ztcfg:  succeeded
Jan  6 14:55:07 localhost kernel: TE410P: Span 1 configured for ESF/B8ZS
Jan  6 14:55:07 localhost kernel: Registered tone zone 0 (United 
States / North America)
 


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Try this,
Start  the linux box
unload zaptel driver by running  modprobe -r wct4xxp  and modprobe -r zaptel
at this point you should see running red led (1 at the time)  on all 4 ports
Config your zaptel.conf with  appropriate span ( say  2)  the ex below 
is for EUROISDN
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16

span=2,1,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
--- Now reload the driver by modprobe wct4xxp
( no need to run modprobe zaptel)
and run ztcfg - and check channels numbering and status
you should see red blinking light on span 1, and 2
now connect a cross cable like recommanded by Scott Stingle
start  asterisk with the appropriate  zapata.conf
you shouls see green llight if your config is correct
Guck
Jack
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Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

[pbx]

Eric Bishop wrote:
Well it's clear now that this is not an isolated issue. Has anyone
been in touch with Digium about this issue? I have logged a support
issue with them, but  thus far have not received a response. Anyone
had better luck with Digium support and the Compaq/HP G4 server
series?
On Wed, 5 Jan 2005 18:05:22 +0100, Tais M. Hansen <[EMAIL PROTECTED]> wrote:
 

On Monday 03 January 2005 19:34, [EMAIL PROTECTED] wrote:
   

Has anyone had success using a TE410P card in an HP-Compaq DL380 G4
server?
 

We're struggeling with the same thing right now. We have several TE410Ps
working on DL380G3s, but have so far been unsuccessful in getting it to work
on the G4.
Our G4 config is dual xeon 3.6ghz, 2gb ram, kernel 2.6.10 and 2.4.28.
zaptel and wct4xxp modules loads fine. At this point the flashing red lights
on the wct4xxp are turned off. zttool shows all spans are OK, no matter if
there are anything plugged in.
--
Regards,
Tais M. Hansen
ComX Networks A/S
Tel: +45-70257474
Fax: +45-70257374
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I just installed few TE410P in ml330/350 G3 dual Xeon 3.6ghz  without 
any problem  (kernel 2.6.8)
Coulds you describe the problem more specificaly.
Jack


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Re: [Asterisk-Users] Alcatel PBX

[pbx]

[EMAIL PROTECTED] wrote:
Dear Users,
i have the following scnario.
1. Alcatel PBX with e1 module
2. asterisk server with 2 digium e100p cards. 1 connected to pstn e1, 1
connected to alcatel pbx.
i m having problem in outgoing from alcatel.
incoming from pstn -> asterisk -> alcatel working fine, but outgoing from
alcatel -> asterisk -> pstn or any sip extensions not working. it hangs up the
line as soon as i answer the call. i have generated dialtone via playtones but
it has also issue.
when i connect pstn e1 line directly to altacel e1 module, it works fine, but
behind asterisk it hangups.
any body have good idea ?
further details can be provided if u need more.
regards.
-Neo


This message was sent using IMP, the Internet Messaging Program.
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It might be a good Idea, to at least send your config file zaptel, 
zapata,and extension
and  the message  that you got on the console  when  the problem occurs
... Regards,
Jack

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Re: [Asterisk-Users] compiling error

[pbx]

Read Asterisk install,
You need to install libssl package

Wesley Jay Deypalan wrote:
Hi,
I tried to compiled Asterisk on a Mandrake 9.2 and I encountered this error
command: make
after compiling for sometime then this error appeared
gcc -g  -o asterisk -Wl,-E  io.o sched.o logger.o frame.o loader.o
config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o
manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o
autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o
aestab.o aeskey.o utils.o  editline/libedit.a db1-ast/libdb1.a
stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -lssl
/usr/bin/ld: cannot find -lssl
collect2: ld returned 1 exit status
make: *** [asterisk] Error 1
I'm not really knowledgeable in compiling. What does this mean? Did I
missed something?
TIA,
Wesley
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Re: [Asterisk-Users] Can some bady help me ???

[pbx]

the commande is make;make install
not make:make install
you  typed ":" instead of ";"
Rodney Acosta Coya wrote:
i found de file Makefile but i dont now what to do with  it
look at this

inux:/inst/pbx/asterisk-1.0.0 # make:make install
bash: make:make: command not found
linux:/inst/pbx/asterisk-1.0.0 # ls
.  README.fpm  apps   callerid.c  config.c  editline
io.c   pbx  sounds
.. SECURITYast_expr.y cdr configs   enum.c
keys   pbx.csounds.txt
.version   acl.c   astconf.h  cdr.c   contrib   file.c
loader.c   poll.c   srv.c
BUGS   aescrypt.c  asterisk.8.gz  channel.c   db.c  formats
logger.c   privacy.cstdtime
CREDITSaeskey.casterisk.c channelsdb1-ast   frame.c
make_build_h   redhat   tdd.c
ChangeLog  aesopt.hasterisk.h chanvars.c  dlfcn.c   fskmodem.c
manager.c  res  term.c
HARDWARE   aestab.casterisk.sgml  cli.c   dns.c image.c
md5.c  rtp.ctranslate.c
LICENSEagi astman codecs  doc   images
mkdep  sample.call  ulaw.c
Makefile   alaw.c  astmm.ccoef_in.h   dsp.c include
muted.csay.cutils.c
README app.c   autoservice.c  coef_out.h  ecdisa.h  indications.c
muted.conf.sample  sched.c
linux:/inst/pbx/asterisk-1.0.0 # Makefile install
bash: Makefile: command not found
linux:/inst/pbx/asterisk-1.0.0 # nmake install
bash: nmake: command not found
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Re: [Asterisk-Users] Music on Hold on Debian 2.6 help wanted

[pbx]

Not realy sure, but it seems that you are missing Zaptel timing,
Just Google for ZTDUMY or viop-org,  ZTDUMY take TDM timming from a 
USB-DEVICE
this might be your problem...

I am running in  under root!
Jack

Joost Kraaijeveld wrote:
Hi Jack,
[EMAIL PROTECTED] schreef:
 

Joost , I am running 2.6.8  SMP Sarge/Debian on a HP ml330 an have no
problem. The Zaptel hardware is e T410P
Are you running without Zaptel Hardware ?
   

Yep. I have two Winbond ISDN cards in the machine. 

I am relieved that someone succeeded in running it on (almost) the same kernel. 
I almost (?) gave up hope.
BTW: do you run Asterisk as root or as asterisk?
Groeten,
Joost 
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Re: [Asterisk-Users] Music on Hold on Debian 2.6 help wanted

[pbx]

Joost , I am running 2.6.8  SMP Sarge/Debian on a HP ml330 an have no 
problem.
The Zaptel hardware is e T410P

Are you running without Zaptel Hardware ?
Jack

Joost Kraaijeveld wrote:
Hi all,
For some reason Music On Hold does not work. I have searched the internet for 
solutions but found nothing that helped.
I use Asterisk 1.0.1 and mpg123 0.59r on Debian 2.6.7-1-386 (Sarge). mpg123 works on the 
commandline (I get sound from the soundcard). If I start Asterisk, two instances of 
mpg123 are started with it (ps shows: mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3). Calling an extension that just 
plays the music on hold only shows "Unable to start music on hold (class 'default') 
on channel SIP/softel1-df8f" in the logfiles and the connection is terminated.
Is there anyone who has the same configuration and that has music on hold 
working? Does anyone has any ideas about solving this problem? Or  is it just 
not possible (yet) with this distribution?
BTW: I do get a warning related to music on hold during asterisk's startup that 
may be related (?):
On  [res_musiconhold.so]Nov 18 09:39:24 VERBOSE[1077055616]:  [res_musiconhold.so] 
=> (Music On Hold Resource)
NG[1077055616]: Unable to open pseudo channel for timing...  Sound may be 
choppy.
TIA
Joost 

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Re: [Asterisk-Users] Can't install the mfcr2 support correctly

[Pbx]

Dear Khaled,
I thing you must read the documentation a little bit more deapely!
does zaptel compile ok ?
which kernel are you using ?
have you configure the zaptel.conf file
which parameters are you using for r2 signaling ?
refer to this page  as guide for starting 
http://www.asterisk.org/index.php?menu=download read the all page
no  lights aster wct4xxp means span is not configured!
good luck !

Jack

- Original Message - 
From: "Abdelghani Khaled" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Sunday, October 31, 2004 12:04 PM
Subject: [Asterisk-Users] Can't install the mfcr2 support correctly


Hi Mr Jack, hi everybody
Thank you for your answer for my message titled "can't run ztcfg". I tried 
what you proposed me and the error I told about is not signaled. However I 
still have problems to get mfc/r2 support running.

I refered to the mfcr2 support documentation available in the opencall.org 
website http://www.opencall.org/installing-mfcr2.html .

The digium card I have is the TE410P
After installing the zaptel driver using the following commands:
make clean
make install
make config
After adding the necessary lines to the zaptel.conf file.
The problem is that when I execute the following command :
modprobe wct4xxp
which is necessary to run ztcfg correctly (as you told me) I get the 4 
ports lights of the card off.
Yesterday I got them red.

And when I connect an E1 to a port I don't get a green light but I get 
either a red light (or no light at all) if the light was already red or no 
light if the light was already off.

I hope that my message is clear and that someone will help me.
_
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Re: [Asterisk-Users] can't run ztcfg

[Pbx]

Hello ,
You need to check if the module relative to your card  is loaded with lsmod
depending on the card you have will need to  start the following command
modprobe wcfxs   for tdm400p
or
modprobe wcfxo - for fxs card ( x100p)
or
modprobe wct1xxp  for single E1/T1 or
modprob wct4xxp for quad E1/T1
hope this helps
Jack

- Original Message - 
From: "Abdelghani Khaled" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 27, 2004 4:27 PM
Subject: [Asterisk-Users] can't run ztcfg


Hi everybody,
I am algerian and I am trying to install the zaptel driver.
I bring the source from the cvs server. I build the source using the make 
clean; make install; command. When running the ztcfg file I get the 
following:

Notice: Configuration file is /etc/zaptel.conf
line 140: Unable to open master device 'dev/zap/ctl'
I'd like someone to help me resolving that problem or explain me the 
signification of that notice.

_
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[Asterisk-Users] TDM400P - TE405P- configuration issue

[Pbx]

Hello, 
 
I just installed a Asterisk box with A 
TDM400P  and a TE405P
 
Installing the TE405P  alone in the (connected 
to Ericcson -E1 euro ISDN ) everythings works great! 
 
installing the TDM400P alone ( connected to analog 
bundle works great)! 
 
 
Configuring both card  gives problem. 

 
if I configure TDM400P first in zaptel.conf 
chaanles 1,2 and 3,4( FXO/FXS) and modeprobe wcfsx and than wct4XXP
( channel from 5 on) the TE405P is ignored by the 
system ( no red light, no life on the card) 
 
Now if I configure  the T405P first (channels 
1, ) and TDM400P behind loading WCT4XXP first and than WCFXS
the TE405 works fine, an the TDM400P  looks 
dead ( no green lights) 
 
Does some of you have a configuration sample ? 

Or any Idea on whats going on 
 
intensive Googeling gives me no 
result
 
Merci 
 
Jack
 
 
 
 
 
 
 
 
 
Does any one in this list have a zapata.zaptel.con 
sample file 
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[Asterisk-Users] REPOSTED: Problems patching Makefile in apps directory

[PBX Portela]

Dear Friends,I'm trying to install the spandsp in my Asterisk box, 
in order to userxfax and txfax.Well,  following the instruccions i 
try to patch the Makefile within theasterisk/apps source directory and found 
the following error message[EMAIL PROTECTED] apps]# patch < 
Makefile.patchpatching file MakefileHunk #1 FAILED at 35.Hunk #2 
FAILED at 68.2 out of 2 hunks FAILED -- saving rejects to file 
Makefile.rejMay you help me?Thank you in 
advance.Juanjo
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[Asterisk-Users] Problems patching Makefile in apps directory

[PBX Portela]

Dear Friends,

I'm trying to install the spandsp in my Asterisk box, in order to use
rxfax and txfax.
Well,  following the instruccions i try to patch the Makefile within the
asterisk/apps source directory and found the following error message

[EMAIL PROTECTED] apps]# patch < Makefile.patch
patching file Makefile
Hunk #1 FAILED at 35.
Hunk #2 FAILED at 68.
2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej

May you help me?

Thank you in advance.
Juanjo

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[Asterisk-Users] Phone numbers in SPAIN

[PBX Portela]

I am looking for a provider that will provide a 
phone number via IAX, IAX2 or SIP using numbers in Barcelona or Tarragona or 
even Other City in Spain. May be a 902 number.
 
Thanks in advance.

[Asterisk-Users] Re:FATAL: Module zaptel not found.

[PBX Portela]

The output of my lsmod doesn´t seems to have the zaptel and wcfxo modules,
And the output of cat /proc/interrupts is:

   CPU0
  0:  103001742  XT-PIC  timer
  1:   1134  XT-PIC  i8042
  2:  0  XT-PIC  cascade
  5:  37511  XT-PIC  Ensoniq AudioPCI
  8:  1  XT-PIC  rtc
  9:  0  XT-PIC  ohci_hcd
 10:4979359  XT-PIC  eth0
 11:  10285  XT-PIC  aic7xxx
 12:   3070  XT-PIC  i8042
 14: 824226  XT-PIC  ide0
 15:485  XT-PIC  ide1
NMI:  0
ERR:  0

And the kudzu detected the card as a modem.
When i made "make" in /usr/src/zaptel gave me error.

May you help me.

Kind regards,
Juanjo
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[Asterisk-Users] FATAL: Module zaptel not found.

[PBX Portela]

Dear Sirs,

I'm running an Asterisk 0.9.1 in a Fedora Core 2 box.
I installed a X100P card on my box and when i try to load modules i am
rejected.
When i type modprobe zaptel my Fedora respond : "FATAL: Module zaptel not
found." . The same uccurs when i type "modprobe wcfxo"

May you help me.

Thank you in advance

Juanjo
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[Asterisk-Users] Sipura, Asterisk, *0, and Call Waiting

[PBX Tech]

I have searched through all the lists and found that some people have had 
luck with flash hook, then *0 to answer a call waiting call.

I have an Asterisk server with one FXO card, the dialtone for the fxo card 
is providing by another pbx called a Definity.

When I am on the Sipura, and another call comes in, I hear the call-waiting 
indicator, when I flash hook I just hear tone, if I dial *0 I just hear dead 
air.

Its like the Asterisk isnt flashing the fxo line.

Any suggestions?

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RE: [Asterisk-Users] Remote reload Cisco 7960

[PBX]

Title: Message



One 
way maybe not the best, but can work.
 
If you 
are pulling down the config from a tftp server.  Turn on telnet to the 
phone.  Create a perl script to telnet and reboot the phone.  When the 
phone boots back up it will grab the new config from the tftp 
server.
 
-gcc

  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of B. J. 
  BomarPosted At: Friday, January 16, 2004 1:13 PMPosted 
  To: Asterisk User GroupConversation: [Asterisk-Users] Remote 
  reload Cisco 7960Subject: [Asterisk-Users] Remote reload Cisco 
  7960
  Does anyone have a 
  working way of having a Cisco 7960 reload its config remotely.  I have 
  tried some of the scripts that I have found on the web, but to no avail.  
  Thanks for the help.
   
  B. 
  J.
   
   
   
   

RE: [Asterisk-Users] Programming an unlocked ADSI phone?

[PBX]

I wanted to give you some guidance on the configuration of the phone
Here is sniplet of configuration Aastra 390 and 480 Phones...


In an ADSI script for the 1st Slot:
;
; Asterisk default ADSI script
;
;
; Begin with the preamble requirements
;
DESCRIPTION "Asterisk PBX" ; Name of vendor
VERSION 0x00   ; Version of stuff
;SECURITY "_AST"   ; Security code
SECURITY 0x9BDBF7AC; Security code
FDN 0x000F ; Descriptor number
In an ADSI script for the 2nd Slot:
;
; Asterisk default ADSI script
;
;
; Begin with the preamble requirements
;
DESCRIPTION "Asterisk PBX" ; Name of vendor
VERSION 0x00   ; Version of stuff
;SECURITY "_AST"   ; Security code
SECURITY 0x78921D49; Security code
FDN 0x85EFD9DA ; Descriptor number

You wouldn't need to program any other slots than these 2, and the 1st
slot is the only one that MUST contain programming. This is because the
first slot is triggered when the phone rings or when the phone is placed
off-hook. The second slot (the Self Launching slot) is triggered when
the phone has had no activity for a certain amount of time. Programming
in this slot can be identical to slot 1, or it can be completely
different, such as for advertising purposes. 

How to clear the ADSI Scripts from the phone

Hit options.
Choose Time/Date
Set the time to Jan 1 12:00am
Hit done
Hit done
Hit options
Hit Mute or Flash 
A display giving the CPE ID and other stuff will appear
QUICKLY press the # key

Hope this helps.

-gcc


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Posted At: Wednesday, December 31, 2003 2:58 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Programming an unlocked ADSI phone?
Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone?


Tim,

Thanks for your continued participation in this thread.

The truth is, it's not clear to me how to delete a service ... the
services menu only allows me to 'Select' or 'Quit'.

It's also not clear to me how you managed to get Comedian Mail
downloaded to Slot 1 without unlocking it with a code.

When you did the ADSIProg originally (presumably you only did this
once), did you download to both slots, or just the second one?

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax

- Original Message - 
From: "Tim Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 31, 2003 2:42 PM
Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone?


Have you tried deleting those services and adding them back in by
checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg


You should be able to just hit the "#" during the call, but you will
also have to make sure you have the |Tt defined in your extensions.conf
file as well.




Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227


-Original Message-
From: Darren Nickerson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 10:02 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone?

Yes, there are 2 available slots. When I push services, I see:

  Services
> Asterisk PBX
   Asterisk PBX
 
 

As I mentioned, the only soft-key I seem to have is 'VMail', and when I
select that I still get:

Comedial Mail
download refused

Services is full

And yet somehow there does seem to be some ADSI functionality later on,
when it falls back to a voice prompt after the download is refused and
I'm navigating the voice prompts, reading the messages.

It troubles me that Asterisk isn't very chatty about the ADSI stuff ...
when I access VMail all I see is:

   -- Starting simple switch on 'Zap/5-1'
-- Executing VoiceMailMain("Zap/5-1", "") in new stack
(here's where the ADSI stuff seems to happen on the phone, download
refused etc)
-- Playing 'vm-login' (language 'en')
[snip rest of log]

There's no indication that any ADSI transactions are going on here, but
that tell-tale tone can be heard and the little rotating animated cursor
on the phone means _something_ is definitely going on at the ADSI level.
Am I missing a debug option that could show me more about what may be
going wrong?

I'll get with Sayson tech support today to see if they can make any
sense of this.

-Darren

--
Darren Nickerson
Senior Sales & Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 ext 8106 office
+1.215.243.8335 fax

- Original M

[Asterisk-Users] AGI - IVR - Time Clock

[PBX]

I wanted to post the beginings of my latest IVR Project for an automated
Time Clock software.

The customer has over 300 Field Reps that call in everytime they arrive
on location and whey they leave that location.  This is handled by the
receptionist now and she logs in them and out of there Time Clock
Software.  Which takes up majority of her day.  The customer has
requested a automated way of handling these request.

AGI Script 

#!/usr/bin/perl

use Asterisk::AGI;
use DBI;

$AGI = new Asterisk::AGI;

my %input = $AGI->ReadParse();
my $callerid = $input{'callerid'};

# Time Clock Questions 

my $empid = $AGI->get_data('employee',-1,5); # Asks for Employee ID
$AGI->stream_file(entered);
$AGI->say_digits($empid);

my $optemp = $AGI->get_data('correct',-1,1);  # Asks if what was
entered is correct otherwise ask question again

if ($optemp != 1) {
employeeid ();
}



my $strid = $AGI->get_data('store',-1,5);  # Asks for Store ID
$AGI->stream_file(entered);
$AGI->say_digits($strid);

my $optstr = $AGI->get_data('correct',-1,1);# Asks if what
was entered is correct otherwise ask question again

if ($optstr != 1) {
storeid ();
}


my $stat = $AGI->get_data('status',-1,1);   #Asks - Login or Logout

if ($stat == 1) {
$AGI->stream_file(login);
}else{
$AGI->stream_file(logout);
}

my $optstat = $AGI->get_data('correct',-1,1); # Asks if what
was entered is correct otherwise ask question again

if ($optstat != 1) {
status ();
}


# Database Connection ###

my $dbh = DBI->connect("DBI:mysql:database=service;host=localhost",
"username", "password",
{'RaiseError' => 1});

$query = "INSERT INTO auto (Callerid, Date, Time, Empid, Strid, Status)
VALUES ('$callerid', sysdate(), sysdate(), '$empid', '$strid',
'$stat')";

$sth = $dbh->prepare($query);
$sth->execute();

$sth->finish();


$dbh->disconnect;



$AGI->stream_file(beep);


### Sub Routines 

sub employeeid {

my $empid = $AGI->get_data('employee',-1,5);
$AGI->stream_file(entered);
$AGI->say_digits($empid);

my $optemp = $AGI->get_data('correct',-1,1);

}


sub storeid {

my $strid = $AGI->get_data('store',-1,5);
$AGI->stream_file(entered);
$AGI->say_digits($strid);

my $optstr = $AGI->get_data('correct',-1,1);

}


sub status {

my $stat = $AGI->get_data('status',-1,1);

if ($stat == 1) {
$AGI->stream_file(login);
}else{
$AGI->stream_file(logout);
}

my $optstat = $AGI->get_data('correct',-1,1);


--

Page to view data that was entered

There is 3 Tables involved...

1. Data that is entered by user
2. Employeed Name -> Employee ID
3. Store Named -> Store ID


-




Service Express Time Clock




  
Service Express Time
Clock
  
  
Caller ID
Store Name
Employee Name
Date
Time
Login / Logout
  
";

for ($i=0; $i<$number; $i++) {
$v_callid = mysql_result($result, $i, "CallerID");
$v_store = mysql_result($result, $i, "name");
$v_emp = mysql_result($result, $i, "employee");
$v_date = mysql_result($result, $i, "Date");
$v_time = mysql_result($result, $i, "Time");
$v_stat = mysql_result($result, $i, "Status");

if ($v_stat == '1') {
$v_stat_chgn = "Login";
}else{
$v_stat_chgn = "Logout";
}

// print "$v_callid, $v_date, $v_time, $v_empid, $v_strid,
$v_stat, $v_stat_chgn";
echo "";
echo "$v_callid";
echo "$v_store";
echo "$v_emp";
echo "$v_date";
echo "$v_time";
echo "$v_stat_chgn";
echo "";
}


mysql_close();

?>





Again this is a basic script.  The next step is for the AGI script to
integrate with the Time Clock software so there is no interaction other
than the field reps when they call in to login or logout.

-gcc
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[Asterisk-Users] SIP / FXS - MOH

[PBX]

Is there anway to do MOH on a FXS extension like what is done using SIP.
There has to be a way within manager or something, to send this call to
MOH and then retreive the call.

I need to set this up, so users are just hitting one button to put
callers on hold and one or another button to retrieve the users.

-gcc
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[Asterisk-Users] SIP - Ringback

[PBX]

I am new to the sip side of things and have a question regarding
ringback.  I don't hear ringback when using the sjphone softphone when
dialing internal extensions.  It's fine when dialing outside over the
pstn.

Is this a issue of the softphone, configuration or sip in general?

Thank you,

-gcc
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RE: [Asterisk-Users] SIP / X-ten Softphone

[PBX]

Andrew---

Thank you for the Sjphone insight.  I am liking this much better.  I can
register and call a extension now.  I am having other issues. I'm sure
it's just config issues though.

Thanks...  

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Posted At: Thursday, December 18, 2003 11:16 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] SIP / X-ten Softphone
Subject: Re: [Asterisk-Users] SIP / X-ten Softphone


As with any debugging, you should try steps seperately...

- Original Message -----
From: "PBX" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, December 18, 2003 10:42 PM
Subject: [Asterisk-Users] SIP / X-ten Softphone

> [1005]
> type=friend
> secret=1005
> host=dynamic
> mailbox=1005
> context=default

Add user=1005 to this sip.conf definition.

>
> X-Lite Softphone
>
> Network->Out Bound SIP Proxy: (IP of *)
>
> SIP Proxy->Default->Enabled: Yes
> SIP Proxy->Default->UserName: 1005
> SIP Proxy->Default->Password: 1005
> SIP Proxy->Default->Domain/Realm: (IP of *)
> SIP Proxy->Default->SIP Proxy: (IP of *)
> SIP Proxy->Default->Send Internal IP: Always

Take a look at some screenshots of X-Lite configs. I remember there
being a line or two that wanted to have :5060 after the IP.

As a side note, I gave up on X-Lite. I found it difficult to navigate
the configuration menus, and there was a background hum that isn't there
if I use sjphone, diax, or iaxcomm/iaxclient.


Andrew Thompson http://aktzero.com/

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[Asterisk-Users] SIP / X-ten Softphone

[PBX]

I know this has been covered more times than to mention and this is
where I got most of my info from... But I am having issues with this.  I
can't seem to get the phone to register with *.  This is being tested on
a internal network right now.

Here is the setup -

sip.conf

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0  ; Address to bind to
context = default   ; Default for incoming calls
allow = all

[1005]
type=friend
secret=1005
host=dynamic
mailbox=1005
context=default

extension.conf

; SIP

exten => 1005,1,Dial,SIP/1005|15
exten => 1005,2,Voicemail2(u1005)
exten => 1005,102,Voicemail2(b8200)

X-Lite Softphone

Network->Out Bound SIP Proxy: (IP of *)

SIP Proxy->Default->Enabled: Yes
SIP Proxy->Default->UserName: 1005
SIP Proxy->Default->Password: 1005
SIP Proxy->Default->Domain/Realm: (IP of *)
SIP Proxy->Default->SIP Proxy: (IP of *)
SIP Proxy->Default->Send Internal IP: Always

But the only thing I every get is discovering firewall and Discovered
Full Cone Nat Firewall.

Any ideas...

-gcc
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RE: [Asterisk-Users] Any Ideas

[PBX]

Ok...

Let me give a better example.

A caller calls in and a user picks up the phone.  Then the user needs to
put the caller on hold so he can go check on something.  He would like
to press the hold button on the phone and hang the receiver up.  He can
do this, but the caller never hears MOH.  The user does what he needs to
do and comes back and picks up the receiver and press hold to release
the caller from hold.


I would like this functionality - but for the caller to hear MOH.  You
mentioned I could do some redirects via the manager interface to get the
call back if I just put in out in an extension playing MOH.

-gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Posted At: Thursday, December 18, 2003 6:44 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Any Ideas
Subject: Re: [Asterisk-Users] Any Ideas


Hi!

> I need to come up with a solution that the user can place the caller 
> on hold, the caller here MOH and the user hang the receiver up.  Just 
> as if they hit, the hold button on the phone.  This can be done, using

> ADSI if need be.

What you are trying to do doesn't seem to make much sense. First of all 
it sounds like you *really* want to do call parking. Secondly, if you 
hang up, then what are you going to do with the caller? Why not right 
away hang up on the caller - or do you want to collect phone fees from 
him while having him listen to MOH indefinitely? :->

Here's one way to do it: Create an extension that looks like

exten => 333,1,Answer
exten => 333,2,MusicOnHold(default)

and then use # to transfer the caller to that extension. Unless you use 
the manager interface (redirect) or some smart scripting/ dialplan
layout 
you won't be able to get back to that caller though. But you didn't say 
that you need to do that. ;->

Cheers, Philipp


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[Asterisk-Users] Any Ideas

[PBX]

Has anyone come up with any ideas on how to place a call on hold and
have them use MOH, with out having to park the call?  This is using a
analog phone.  I know you can hit flash or # but that just gives me dial
tone.

I need to come up with a solution that the user can place the caller on
hold, the caller here MOH and the user hang the receiver up.  Just as if
they hit, the hold button on the phone.  This can be done, using ADSI if
need be.

Thanks for any insight,

-gcc
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RE: [Asterisk-Users] On Hold - Talked about before

[PBX]

I should have stated this.  Is there any Analog phones that can do this.

-gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Posted At: Tuesday, December 09, 2003 11:57 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] On Hold - Talked about before
Subject: Re: [Asterisk-Users] On Hold - Talked about before


At 08:45 PM 12/9/2003, you wrote:
>Ok - Here is where I am at.  I know this topic has been discussed 
>before, but never a solid answer was set in place.  Is anyone aware of 
>any phones that can put a caller on hold and the caller hear MOH by the

>user pressing the hold button.  I understand most phones are only 
>muting the speaker and handset.

The SNOM phones can do this, and are also excellent phones generally. 
Install the 1.6x software build for now; the 2.x build changes their 
behavior a bit and breaks MOH with asterisk. This is being worked on.

--Ernest

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[Asterisk-Users] On Hold - Talked about before

[PBX]

Ok - Here is where I am at.  I know this topic has been discussed
before, but never a solid answer was set in place.  Is anyone aware of
any phones that can put a caller on hold and the caller hear MOH by the
user pressing the hold button.  I understand most phones are only muting
the speaker and handset.

I need to find a solution for this either this way or by implementing
something in ADSI.  But I have been unable to find any type of HOLD
pattern in ADSI.  A user can park the call and that in it self has
benefits.  But if a user just wants to put a caller on hold and then
come back and retrieve the caller from hold by hitting the hold button
again - how is this done?

And this has nothing to do with transfering the call.  I want the user
to be able to press "A" hold button and hang the phone up and the caller
hear MOH.

If any one can point me in the right direction that would be great.

-gcc
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RE: [Asterisk-Users] AGI (IF/ELSE)

[PBX]

Ok.. I was thinking about this.. It is not a very wise decsion to put
the user input in a loop.. So how could I do some error checking outside
of the loop?

-gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Thursday, November 27, 2003 9:19 PM
Posted To: Asterisk User Group
Conversation: AGI (IF/ELSE)
Subject: [Asterisk-Users] AGI (IF/ELSE)


I need some help with some statements.

#!/usr/bin/perl

use Asterisk::AGI;

$AGI = new Asterisk::AGI;

my %input = $AGI->ReadParse();
my $callerid = $input{'callerid'};

if ($optemp != 1) {

my $empid = $AGI->get_data('employee',-1,5);
$AGI->stream_file(entered);
$AGI->say_digits($empid);

my $optemp = $AGI->get_data('correct',-1,1);

}else{

my $strid = $AGI->get_data('store',-1,5);
$AGI->stream_file(entered);
$AGI->say_digits($strid);

my $optstr = $AGI->get_data('correct',-1,1);
}

exit;

I can't seem to figure out what I am doing wrong.  When the script is
run. The user puts in there employee ID and then hears it back to them.
Then they are asked if this is correct press 1 for yes or 9 for no.  If
they press 1, it should go onto the next piece of the script

But if I press 1 the script ends Any ideas 

Thanks, 

-gcc
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