Re: [asterisk-users] TCP Trigger on incoming call request

[Daniel Isenmann]

Thank for the hint. I will have a look into it.

Daniel

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von isr...@gmail.com
Gesendet: Freitag, 6. Mai 2011 15:22
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] TCP Trigger on incoming call request

Look at function CURL 

-Original Message-
From: Daniel Isenmann daniel.isenm...@seetec.de
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 6 May 2011 13:04:09 
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] TCP Trigger on incoming call request

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[asterisk-users] how to play music when dial fail or time out

[John Wu]

Hi all,
I need to support this feature. When caller dial if the dial fail or no
answer from the
called number then play a music. So how to achieve that?

Thanks!
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[asterisk-users] Slightly OT: Android phone as sip-gw?

[Jay R. Worthington]

Hi,

i have some spare (read: Boss get's a new one every few month ;)) Android
Phones laying around. Does someone know a way of using them as a mobile
gateway for asterisk? I could not find any SIP-Gateway in the Market, and i
don't think it's possible to use the GSM Audio directly with something like
chan_datacard...

Regards,

Jay
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[asterisk-users] OUTBOUND CALLER ID

[mahesh katta]

Hi,
THIS IS IN DUBAI.

I am having PRI line with 100 DID's (00-99) and when we call to any landline
or mobile number then it shows us our board number or pilot number (i.e
4663000 means 00).. As i give all the extensions a particular DID, so people
from outside world can call them. The problem is the CALLERID ... When we
call from any of other extension PSTN line carries out our pilot number (i.e
4663000 means 00) rather than its own DID number.

I have already talked with my service provider and he said that they have
activated it from their end..

Incoming DID's have been configured successfully and probelm is just for the
outgoing caller ID that it shows always is pilot number 

-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] OUTBOUND CALLER ID

[A J Stiles]

On Monday 09 May 2011, mahesh katta wrote:
 Hi,
 THIS IS IN DUBAI.

 I am having PRI line with 100 DID's (00-99) and when we call to any
 landline or mobile number then it shows us our board number or pilot number
 (i.e 4663000 means 00)..

In the context through which outgoing calls are placed, you need a step which 
sets the caller ID number.  For instance, part of our dialplan maps external 
phone numbers with the local part 707060 to 707072 to internal extensions 301 
to 312 respectively.  Our E1 provider also requires us to include the STD 
code, minus the leading zero, for the town we are in -- and will silently 
anonymise the call if we try to send a caller ID that does not belong to us.

So for outgoing calls, we have something like

[ts-outgoing]
exten = _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
exten = _0., 2, Set(CALLERID(num)=${STD}${localno})


-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

[Ishfaq Malik]

On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
 Are you not seeing issues with *8 call pick up then ?

Nope, I double checked it after seeing someone saying they had issues
with it and it is fine on the installation I have.

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] RTP Path and t or T option

[Dovid Bender]

Hi,

I have been away from the list for a bit so please forgive me if this has 
already been covered.

From what I understand if I have the t or T option in my dial string then the 
RTP must go through Asterisk since we need to know if any DTMF was pressed.

If both users and the server are not behind NAT and we set up for the RPT to go 
direct if I am using INFO for DTMF, when using the t or T option does the RTP 
go direct ?

I am asking since from what I understand out of band DTMF was created so that a 
leg in the middle can get the DTMF with out having to be in the RTP stream.

Further more in 1.8.X where there is IPv6 support if using IPv6 with DTMF info 
shouldn't the RTP go direct ?

I am asking more then complaining (since I do not have multiple IPv4 IP's to 
test this). If this is not how Asterisk works I would like to see if this can 
be patched so we can save RTP long trips.

Regards,

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Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

[Dovid Bender]

Jay,

AFAIK you can only use chan_datacard with a specific USB modem (I can not 
recall the name at the moment).

I have tried it and it works well for both Audio and SMS messages.

You may want to try chan_mobile which works over blue tooth.

Regards,

Dovid

  - Original Message - 
  From: Jay R. Worthington 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, May 09, 2011 10:47
  Subject: [asterisk-users] Slightly OT: Android phone as sip-gw?


  Hi,

  i have some spare (read: Boss get's a new one every few month ;)) Android 
Phones laying around. Does someone know a way of using them as a mobile gateway 
for asterisk? I could not find any SIP-Gateway in the Market, and i don't think 
it's possible to use the GSM Audio directly with something like chan_datacard...

  Regards,

  Jay




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Re: [asterisk-users] how to play music when dial fail or time out

[Dovid Bender]

John,

You want to do it only after it fails ?

If so you can do something like.

Exten = _X., 1, Dial(SIP/${EXTEN}@PEER)
Exten = _X., 2, GotoIf($[${DIALSTATUS} = ANSWER]?10)
Exten = _X., 4, MusicOnHold()
Exten = _X., 10, Hangup

  - Original Message - 
  From: John Wu 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Monday, May 09, 2011 10:41
  Subject: [asterisk-users] how to play music when dial fail or time out


  Hi all,
  I need to support this feature. When caller dial if the dial fail or no 
answer from the
  called number then play a music. So how to achieve that?

  Thanks!



--


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Re: [asterisk-users] Blacklist with *30

[Dovid Bender]

Alejandro,

What GUI are you using ? I don't think Asterisk comes with *30 to ban calls.

Regards,

Dovid

- Original Message - 
From: Alejandro Cabrera Obed aco1...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Friday, May 06, 2011 23:51
Subject: [asterisk-users] Blacklist with *30



Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon't get the menu but I get a new dial tone.

What happen please ??? What can I do to solve this ???

Thanks a lot,

Alejandro

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Re: [asterisk-users] OUTBOUND CALLER ID

[mahesh katta]

Sir ,

this is not working


On Mon, May 9, 2011 at 1:52 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:

 On Monday 09 May 2011, mahesh katta wrote:
  Hi,
  THIS IS IN DUBAI.
 
  I am having PRI line with 100 DID's (00-99) and when we call to any
  landline or mobile number then it shows us our board number or pilot
 number
  (i.e 4663000 means 00)..

 In the context through which outgoing calls are placed, you need a step
 which
 sets the caller ID number.  For instance, part of our dialplan maps
 external
 phone numbers with the local part 707060 to 707072 to internal extensions
 301
 to 312 respectively.  Our E1 provider also requires us to include the STD
 code, minus the leading zero, for the town we are in -- and will silently
 anonymise the call if we try to send a caller ID that does not belong to
 us.

 So for outgoing calls, we have something like

 [ts-outgoing]
 exten = _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
 exten = _0., 2, Set(CALLERID(num)=${STD}${localno})


 --
 AJS

 Answers come *after* questions.

 --
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-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Sebastian Arcus]

Oh yeah - love your idea :-)

So just to clarify - I take it the Cisco phones (at least the 7940) are 
supposed to be run with a tftp server available at all time - not only 
during the initial configuration? Just making sure I didn't miss 
something obvious in the documentation. Can somebody confirm please.


It's just that the Grandstream phones I use can load firmware and 
configs over http or tftp - but they are happy to work without it once 
they are configured.


Sebastian


On 05/09/2011 02:59 AM, C.J. Adams-Collier wrote:

Run more of your systems as diskless.  Make your tftp setup
indispensable :)

On Sun, 2011-05-08 at 22:37 +0100, Sebastian Arcus wrote:

Hi James,

Thanks for the reply. I'm not concerned about performance. But I've
learned that every extra daemon software on a server comes with its
security caveats. I would feel much better about not having another one
to worry about and keep an eye on.

Sebastian


On 05/08/2011 10:30 PM, James Miller wrote:

I have my tftp daemon running all the time and it really doesnt affect
the performance of the machine. Is there a reason why you want to shut
it down?

“I see blindness more as the ability and sight
more as the disability, I only see that which
is within a person.”
Patrick Henry Hughes - 2009


On 5/8/2011 5:19 PM, Sebastian Arcus wrote:

Hi all,

Sorry for posting here - but I figured there are many people with
Cisco IP phones here - and I use them with Asterisk :-)

I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK,
loaded the SIP configuration files OK, they work with Asterisk just fine.

My question is - will I have to keep on running the tftp server for
them for ever and ever? Isn't there any option for them to just use
the settings they have already loaded form the tftp server - so that I
can kill tftpd on my server machine? I tried doing that, and then the
phones stop booting, going in a loop looking for the tftpd server.

It seems a bit pointless, having to run the tftpd daemon all the time
- although I've already loaded the firmware and configurations I want.

Thank you,

Sebastian

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Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

[randulo]

On Mon, May 9, 2011 at 9:47 AM, Jay R. Worthington
jayrworthing...@gmail.com wrote:
 gateway for asterisk? I could not find any SIP-Gateway in the Market, and i

Portech has made GSM and CDMA gateways for years - nothing that works
with your old Android phones, though.

http://www.portech.com.tw/p3-product1.asp?Cid=6


:r

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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Doug Lytle]

Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp 
server available at all time


That is my experience.  But, if you're running tftp under Linux, then 
it's probably spawned by xinetd and won't be running unless the service 
is requested.


Doug

--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

[Alec Davis]

 
 Are you not seeing issues with *8 call pick up then ?
 --
 Thanks, Phil
 

https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues,
particulary when you have pickupsounds enabled.

Alec 
 


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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

[Satish Patel]

Thanks to all for reply,

I have already put 1.8 in production. Actually we are using basic  
function so I hope we are good and fingurs cross.


--
Sent from my iPhone

On May 9, 2011, at 7:18 AM, Alec Davis siva...@paradise.net.nz wrote:



Are you not seeing issues with *8 call pick up then ?
--
Thanks, Phil



https://reviewboard.asterisk.org/r/1185/ helps with *8 pickup issues,
particulary when you have pickupsounds enabled.

Alec



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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Sebastian Arcus]

On 05/09/2011 12:02 PM, Doug Lytle wrote:

Sebastian Arcus wrote:

Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time


That is my experience. But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be running unless the service
is requested.


Thanks for the reply. No, I run tftpd directly from rc.local script (on 
Slackware). That's fine - I just wanted to know I wasn't doing something 
wrong. If everybody else is in the same boat - I'll just be along for 
the ride then :D


Sebastian



Doug



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[asterisk-users] OT - Which Android handset with Wifi-only ?

[Olivier]

Hi,

I would be curious to play with an Android phone with Wifi-only capability.
My plan is to install Bria on it and see if it could be used within a couple
of WiFi access points, as a high-end wireless phone.

Which handset would you recommend ?

Regards
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Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

[aster...@ck-lee.com]

Chan_datacard can selected model of huawei usb modem for voice and sms.

Chan_mobile on  the other hand use bluetooth connection for voice. I have not 
tried sms. For mobile phone, it seeMs nokia is quite good.

--- Sent with System SEVEN - the new generation of mobile messaging

-original message-
Subject: Re: [asterisk-users] Slightly OT: Android phone as sip-gw?
From: Dovid Bender asteriskus...@dovid.net
Date: 09/05/2011 17:28

Jay,

AFAIK you can only use chan_datacard with a specific USB modem (I can not 
recall the name at the moment).

I have tried it and it works well for both Audio and SMS messages.

You may want to try chan_mobile which works over blue tooth.

Regards,

Dovid

  - Original Message - 
  From: Jay R. Worthington 
  To: asterisk-users@lists.digium.com 
  Sent: Monday, May 09, 2011 10:47
  Subject: [asterisk-users] Slightly OT: Android phone as sip-gw?


  Hi,

  i have some spare (read: Boss get's a new one every few month ;)) Android 
Phones laying around. Does someone know a way of using them as a mobile gateway 
for asterisk? I could not find any SIP-Gateway in the Market, and i don't think 
it's possible to use the GSM Audio directly with something like chan_datacard...

  Regards,

  Jay




--


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Re: [asterisk-users] Slightly OT: Android phone as sip-gw?

[Sebastian Arcus]

On 05/09/2011 08:47 AM, Jay R. Worthington wrote:

Hi,

i have some spare (read: Boss get's a new one every few month ;))
Android Phones laying around. Does someone know a way of using them as a
mobile gateway for asterisk? I could not find any SIP-Gateway in the
Market, and i don't think it's possible to use the GSM Audio directly
with something like chan_datacard...


I used older Nokia phones with chan_mobile/chan_bluetooth for incoming 
calls. I vaguely believe that the documentation stated some of them 
could be used for outgoing calls as well. I believe it depends on which 
Bluetooth headset stack the actual phone implements. I'm afraid I don't 
know if any of the Android phones have the right Bluetooth 
implementation to do outgoing calls.


I'm actually quite curious about the results - could you post here if 
you find a way please.


Sebastian



Regards,

Jay



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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

[mgraves]

Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.

Michael Graves
mgraves  mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgra...@mstvp.onsip.com
skype mjgraves

  Original Message 
 Subject: [asterisk-users] OT - Which Android handset with Wifi-only ?
 From: Olivier oza_4...@yahoo.fr
 Date: Mon, May 09, 2011 7:10 am
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 
 
 Hi,
 
 I would be curious to play with an Android phone with Wifi-only capability.
 My plan is to install Bria on it and see if it could be used within a couple
 of WiFi access points, as a high-end wireless phone.
 
 Which handset would you recommend ?
 
 Regardshr--
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[asterisk-users] 40sec between dial execution and sending SIP request

[Pezhman Lali]

Dear
I have a small pbx with asterisk 1.6.2.16.
I have a funny problem, there is exactly 40sec between dial execution and
sending first invite packet on sip.
do you have any idea where the problem is ?

Best regards

-- 
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Re: [asterisk-users] Blacklist with *30

[Alejandro Cabrera Obed]

Dear Dovis, I'm using Elastix and the dialplan comes with this line:

*30,1,Goto(app-blacklist-add,s,1)

Any idea ??? Thanks a lot.

2011/5/9 Dovid Bender asteriskus...@dovid.net:
 Alejandro,

 What GUI are you using ? I don't think Asterisk comes with *30 to ban calls.

 Regards,

 Dovid

 - Original Message - From: Alejandro Cabrera Obed
 aco1...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, May 06, 2011 23:51
 Subject: [asterisk-users] Blacklist with *30


 Dear, when I dial *30 in order to get instructions to blacklist an
 extension, Idon't get the menu but I get a new dial tone.

 What happen please ??? What can I do to solve this ???

 Thanks a lot,

 Alejandro

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-- 
Alejandro Cabrera Obed
aco1...@gmail.com
www.alejandrocabrera.com.ar

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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

[randulo]

On Mon, May 9, 2011 at 2:20 PM,  mgra...@mstvp.com wrote:
 Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.

Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not?

:r

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Re: [asterisk-users] Blacklist with *30

[Dovid Bender]

Try the Elastix forums.

- Original Message - 
From: Alejandro Cabrera Obed aco1...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, May 09, 2011 15:35
Subject: Re: [asterisk-users] Blacklist with *30



Dear Dovis, I'm using Elastix and the dialplan comes with this line:

*30,1,Goto(app-blacklist-add,s,1)

Any idea ??? Thanks a lot.

2011/5/9 Dovid Bender asteriskus...@dovid.net:

Alejandro,

What GUI are you using ? I don't think Asterisk comes with *30 to ban 
calls.


Regards,

Dovid

- Original Message - From: Alejandro Cabrera Obed
aco1...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 06, 2011 23:51
Subject: [asterisk-users] Blacklist with *30



Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon't get the menu but I get a new dial tone.

What happen please ??? What can I do to solve this ???

Thanks a lot,

Alejandro

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www.alejandrocabrera.com.ar

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[asterisk-users] Ustream feed as MOH

[Dovid Bender]

Hi,

Has anyone ever tried getting the Audio of ustream (ustream.tv) in to Asterisk 
for MOH ?

Regards,

Dovid
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Re: [asterisk-users] Background music during a call

[Ioan Indreias]

I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room - in the example we have only one room = 1234

If you prefix the dialled extension with 1 = you will have a lovely
chat. With 2 - cursing chat.

HTH,

Ioan

===

[from-internal]

exten = _1XXX,1,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
exten = _2XXX,1,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))

[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)

exten = love-a,1,Set(__MOH=love)
exten = love-a,2,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten = love-b,1,Goto(chat,100)

exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)

exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten = curse-b,1,Goto(chat,100)

exten = fake,1,Answer
exten = fake,2,MusicOnHold(${__MOH})

exten = chat,1,Goto(100)
exten = chat,2,MeetMe(1234,dx1q)

exten = chat,100,MeetMe(1234,daAx1q)

exten = h,1,MeetMeAdmin(1234,K)
===

On Mon, May 9, 2011 at 8:20 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
 Will this work:
 exten= 123,1,Meetme(1234)
 exten= 123,n,Hangup()
 exten= 5000,1,Dial(Local/123@bk_music/n,,m())
 exten= 5000,2,Goto(bk_music,123,1)

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[asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Hi !

We curently have a centos 5 / asterisk 1.4 server that we have some DTMF 
problems with. It has a Sangoma A104d card and only port one is used to 
connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for 
modem access and port 3 is connected for data communication via PPP.


Now, I want to freshen this setup to something newer. So I installed a 
Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers 
and an A101 card I had laying around.


I did a test this weekend and pluged in our PRI in that test server. I never 
got succeded to have a call trough. When I dialed in, the call is hanged 
up with :


Channel 1/1, span 1 got hanup, cause 6
Spawn extension (ael-default, s, 3) exited non-zero on 
'DAHDI/i1/NPANXX-2'

Hungup 'DAHDI/i1/NPANXX-2'

Here's my dahdi/system.conf :

loadzone=us
defaultzone=us

#Sangoma A101 port 1 [slot:0 bus:6 span:1] wanpipe1
span=1,1,0,esf,b8zs
bchan=1-23
echocanceller=mg2,1-23
hardhdlc=24

my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with :

switchtype=national
pridialplan=unknown
signalling=pri_cpe
group=1
channel = 1-23

as the last non-commented lines.

So, for one thing, the card I have in my test server doesn't have an 
hardware echo canceller, but it's still enabled in my wanpip setting. Could 
that be a source of problem ?


Other than that, is there anything obvious I've missed ? 



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Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.

[stephen.hindmarch]

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Jason Parker
 Sent: 06 May 2011 20:01
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.
 
 On 05/06/2011 01:30 PM, Bob Beers wrote:
  Not sure if this will work, but I'd try adding, before line 86:
 
  #Workaround for PAE
  %if %{paevar} == PAE
  Provides: kmod-dahdi-linux
  %endif
 
  Can't actually test it myself, sorry.
 
  - Bob
 
 
 You'd probably want to modify the kmodtool that comes with it, to just always
 provide kmod-dahdi-linux.
 

Jason,

I solved by editing dahdi-linux.spec to change the Requires to expect the PAE 
variant, but I think Bob's idea is better.

I'm not sure what the kmodtool is. Can you enlighten?

Steve Hindmarch

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Re: [asterisk-users] Background music during a call

[Ioan Indreias]

Updated dialplan: fix a typo when using MOH variable and now you have
truly dynamic conference rooms.

Have fun,
Ioan.

+
exten = _[12]XXX,1,Set(__MM=${EPOCH})
exten = _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
exten = _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))

[chat-room]
exten = love,1,Goto(love-a,1)
exten = love,2,Goto(love-b,1)

exten = love-a,1,Set(__MOH=love)
exten = love-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten = love-b,1,Goto(chat,100)

exten = curse,1,Goto(curse-a,1)
exten = curse,2,Goto(curse-b,1)

exten = curse-a,1,Set(__MOH=curse)
exten = curse-a,n,Dial(Local/fake@chat-room,,G(chat-room,chat,1))

exten = curse-b,1,Goto(chat,100)

exten = fake,1,Answer
exten = fake,2,MusicOnHold(${MOH})

exten = chat,1,Goto(100)
exten = chat,2,MeetMe(${MM},dx1q)

exten = chat,100,MeetMe(${MM},daAx1q)

exten = h,1,MeetMeAdmin(${MM},K)
+

On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias indre...@gmail.com wrote:
 I have tested the following dialplan and it could be used as a
 starting point. What you have to resolve is how to generate different
 MeetMe conference room - in the example we have only one room = 1234

 If you prefix the dialled extension with 1 = you will have a lovely
 chat. With 2 - cursing chat.

 HTH,

 Ioan

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Re: [asterisk-users] Trying out a new version with sangoma card

[Jim Dickenson]

Make sure the firmware on the card is latest. I had a problem, not like your, 
and flashing the card to the latest firmware resolved it.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On May 9, 2011, at 6:11 AM, Nicolas Ross wrote:

 Hi !
 
 We curently have a centos 5 / asterisk 1.4 server that we have some DTMF 
 problems with. It has a Sangoma A104d card and only port one is used to 
 connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for 
 modem access and port 3 is connected for data communication via PPP.
 
 Now, I want to freshen this setup to something newer. So I installed a 
 Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers 
 and an A101 card I had laying around.
 
 I did a test this weekend and pluged in our PRI in that test server. I never 
 got succeded to have a call trough. When I dialed in, the call is hanged up 
 with :
 
 Channel 1/1, span 1 got hanup, cause 6
 Spawn extension (ael-default, s, 3) exited non-zero on 'DAHDI/i1/NPANXX-2'
 Hungup 'DAHDI/i1/NPANXX-2'
 
 Here's my dahdi/system.conf :
 
 loadzone=us
 defaultzone=us
 
 #Sangoma A101 port 1 [slot:0 bus:6 span:1] wanpipe1
 span=1,1,0,esf,b8zs
 bchan=1-23
 echocanceller=mg2,1-23
 hardhdlc=24
 
 my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with :
 
 switchtype=national
 pridialplan=unknown
 signalling=pri_cpe
 group=1
 channel = 1-23
 
 as the last non-commented lines.
 
 So, for one thing, the card I have in my test server doesn't have an hardware 
 echo canceller, but it's still enabled in my wanpip setting. Could that be a 
 source of problem ?
 
 Other than that, is there anything obvious I've missed ? 
 
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Re: [asterisk-users] how to play music when dial fail or time out

[Enrico Cicconi]

Hi,
have you tried to manage all with dialplane ?

just an example:

[incoming]
**exten = s,1,Dial (SIP/your_called_party,20)
exten = s,n, Playback(music_message)
.



In the first step the call is redirect to the configured called party 
and if without answer (busy, not logged, not answered) ...

... in the second step a music is played.

You can also do other kind of job instead of 'playback' if you need.

Hoping to have helped you, have a nice day

Enrico
www.rdmnet.it
http://www.rdmnet.it/asterisk/104-asterisk-in-pillole-impostare-il-dialplan.html



Il 09/05/2011 09:41, John Wu ha scritto:

Hi all,
I need to support this feature. When caller dial if the dial fail or 
no answer from the

called number then play a music. So how to achieve that?

Thanks!


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Re: [asterisk-users] Trying out a new version with sangoma card

[Nicolas Ross]

Make sure the firmware on the card is latest. I had a problem, not like
your, and flashing the card to the latest firmware resolved it.


I did the upgrade, I will make another test when appropriate.

I will also upgrade my curent card, I am curent at version 25, wich dates  
2007, it might solve our curent problem also...


Regards,



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[asterisk-users] asterisk syntax highlighting for gedit

[Naomi Rosenberg]

Hi,

Apologies if this is a duplicate - been having mail server issues and I don't 
think I managed to send it when I tried this morning.

It seems there is no .conf syntax highlighting script available for gedit. I'm 
thinking of putting one together myself, but don't want to reinvent the wheel.

So I'm just enquiring if anyone knows of one that already exists that i've 
missed.

Thanks

Naomi Rosenberg

www.servicesforasterisk.co.uk




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Re: [asterisk-users] asterisk syntax highlighting for gedit

[Alex Balashov]

On 05/09/2011 10:32 AM, Naomi Rosenberg wrote:


So I'm just enquiring if anyone knows of one that already exists
that i've missed.


Not to inflame editor-related passions, but vim does quite a good job.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] asterisk syntax highlighting for gedit

[Ishfaq Malik]

On Mon, 2011-05-09 at 15:32 +0100, Naomi Rosenberg wrote:
 Hi,
 
 Apologies if this is a duplicate - been having mail server issues and I don't 
 think I managed to send it when I tried this morning.
 
 It seems there is no .conf syntax highlighting script available for gedit. 
 I'm thinking of putting one together myself, but don't want to reinvent the 
 wheel.
 
 So I'm just enquiring if anyone knows of one that already exists that i've 
 missed.
 
 Thanks
 
 Naomi Rosenberg
 
 www.servicesforasterisk.co.uk
 
 
The .ini file type does the job

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Warren Selby]

On Mon, May 9, 2011 at 7:07 AM, Sebastian Arcus s...@open-t.co.uk wrote:



 On 05/09/2011 12:02 PM, Doug Lytle wrote:

 Sebastian Arcus wrote:

 Cisco phones (at least the 7940) are supposed to be run with a tftp
 server available at all time


 That is my experience. But, if you're running tftp under Linux, then
 it's probably spawned by xinetd and won't be running unless the service
 is requested.


 Thanks for the reply. No, I run tftpd directly from rc.local script (on
 Slackware). That's fine - I just wanted to know I wasn't doing something
 wrong. If everybody else is in the same boat - I'll just be along for the
 ride then :D

 Sebastian


When I used to have a Cisco 7941 phone in my home office, I only needed the
tftp server online if I made changes to the configuration.  The phone itself
worked fine without one being online all the time.

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] 40sec between dial execution and sending SIP request

[Warren Selby]

On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote:

 Dear
 I have a small pbx with asterisk 1.6.2.16.
 I have a funny problem, there is exactly 40sec between dial execution and
 sending first invite packet on sip.
 do you have any idea where the problem is ?


Check the dial timeout on your phone itself.  What model phone do you have?

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http://www.selbytech.com
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Re: [asterisk-users] OUTBOUND CALLER ID

[Carlos Rojas]

Hello

Do you set your callerid in the context outgoing?

[outgoing]

exten = _X.,1,Set(CALLERID(num)=4663000)
exten = _X.,n,Dial(..

On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.comwrote:

 Sir ,

 this is not working


 On Mon, May 9, 2011 at 1:52 PM, A J Stiles 
 asterisk_l...@earthshod.co.ukwrote:

 On Monday 09 May 2011, mahesh katta wrote:
  Hi,
  THIS IS IN DUBAI.
 
  I am having PRI line with 100 DID's (00-99) and when we call to any
  landline or mobile number then it shows us our board number or pilot
 number
  (i.e 4663000 means 00)..

 In the context through which outgoing calls are placed, you need a step
 which
 sets the caller ID number.  For instance, part of our dialplan maps
 external
 phone numbers with the local part 707060 to 707072 to internal extensions
 301
 to 312 respectively.  Our E1 provider also requires us to include the STD
 code, minus the leading zero, for the town we are in -- and will silently
 anonymise the call if we try to send a caller ID that does not belong to
 us.

 So for outgoing calls, we have something like

 [ts-outgoing]
 exten = _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
 exten = _0., 2, Set(CALLERID(num)=${STD}${localno})


 --
 AJS

 Answers come *after* questions.

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 --
 Best Regards,

 Mahesh Katta
 *BUZZ**WORKS* Business Services Private Limited
 BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
 (E) Mumbai 400069
 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
 Web http://www.buzzworks.com


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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Cassius Smith]

On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote:

Sebastian Arcus wrote:
 Cisco phones (at least the 7940) are supposed to be run with a tftp
 server available at all time

That is my experience.  But, if you're running tftp under Linux, then
it's probably spawned by xinetd and won't be running unless the service
is requested.

Doug
If you want the users to have access to ringtones and desktop images, they
are dynamically loaded via tftp. So yes, you'll need to keep the tftp
server running. 

HTH
Cassius





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Re: [asterisk-users] asterisk syntax highlighting for gedit

[Naomi Rosenberg]

no worries, i'm not as passionate as some! 

It just happens gedit is the one I've gravitated towards, despite what I'm sure 
are good reasons to use something more hardcore.  And I also fancy the project 
of writing the highlighting script, it would be a nice little job for me and 
I'm sure there will be others who would find it useful once made. I just don't 
want to waste time on it if it's been done already, that's all.

Naomi Rosenberg

www.servicesforasterisk.co.uk

- Original Message -
From: Alex Balashov abalas...@evaristesys.com
To: asterisk-users@lists.digium.com
Sent: Monday, 9 May, 2011 3:33:33 PM
Subject: Re: [asterisk-users] asterisk syntax highlighting for gedit

On 05/09/2011 10:32 AM, Naomi Rosenberg wrote:

 So I'm just enquiring if anyone knows of one that already exists
 that i've missed.

Not to inflame editor-related passions, but vim does quite a good job.

-- Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Occasional call from asterisk

[Bruce B]

Thanks for the input. Long ago the CDR showed asterisk as the CLID but it
doesn't anymore so I am puzzled now how to even stop taking calls because my
CLID is now blank and I can't refuse any call with no CLID.

*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*

Here are some out of place messages I am getting in my logs but nothing out
of norm around the time I get Ghost calls though:
*WARNING[11002] chan_dahdi.c: CallerID returned with error on channel
'DAHDI/2-1'*

*NOTICE[12524] chan_dahdi.c: Got event 17 (Polarity Reversal)...*
*
*
*
DEBUG[12524] chan_dahdi.c: Ignore switch to REVERSED Polarity on channel 4,
state 6

DEBUG[12524] chan_dahdi.c: Ignoring Polarity switch to IDLE on channel 4,
state 6
*


Can someone shed light on these options as to what exactly they do:
hanguponpolarityswitch=yes
answeronpolarityswitch=yes

Hopefully some Asterisk guru can tell us more about what might be happening
as I see this as a situation that can be avoided or at least there should be
a workaround for this.

Regards,



On Mon, May 9, 2011 at 9:50 AM, Brian Henning bhenn...@pineinst.com wrote:

 Hello Bruce,



 I did not find a solution, only advice to lead me to think “huh, well
 that’s annoying but we can deal with it.”  I understand from my users,
 though, that it’s *not* always the case that it’s a phantom call—sometimes
 there really is someone calling.



 Note that I haven’t tried what I’m about to suggest, but you might try
 examining the CALLERID data before dialing the SIP extensions and, if it is
 empty or contains “asterisk,” reset it to something like “not available.”



 Cheers,

 ~Brian



 *From:* Bruce B [mailto:bruceb...@gmail.com]
 *Sent:* Friday, May 06, 2011 10:55 PM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Cc:* bhenn...@pineinst.com

 *Subject:* Re: [asterisk-users] Occasional call from asterisk



 Hi Brian,



 Did you find a solution to your problem? or at least got a working
 dial-plan for it? I have the same problem again as well and want to know
 what to do with the dial-plan to off-set the effect at least since Telco
 says it's not their issue.



 Regards,

 Bruce

 On Thu, Apr 7, 2011 at 5:53 PM, Brian Henning bhenn...@pineinst.com
 wrote:

 Hi,

 Now and then our SIP phones ring with asterisk showing as the caller-ID.
 Upon picking up the receiver, there is about five seconds of silence and
 then the channel is closed (hangup).  Can anyone offer some insight?
  Here's
 relevant snippets from my extensions.conf and Master.csv log:

 This line shows up in Master.csv:


 ,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5
 01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07
 21:37:05,2011-04-07 21:37:16,2011-04-07
 21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444,

 Here's [inbound] from extensions.conf:
 [inbound]
 exten = s,1,Answer
 exten = s,n,Ringing
 exten = s,n,Set(CALLERID(num),9${CALLERID(num)})
 exten = s,n,Dial(SIP/504SIP/506,5,tTgr)
 exten = s,n,Goto(1-${DIALSTATUS},1)
 exten = 1-ANSWER,1,Hangup
 exten =
 _1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr)
 exten = _1-.,n,Goto(2-${DIALSTATUS},1)
 exten = 2-ANSWER,1,Hangup
 exten = _2-.,1,Voicemail(499@default,u)
 exten = _2-.,2,Hangup

 The idea is that first 504 and 506 ring, then if neither of them answer,
 everyone rings.  Works great most of the time.

 I have a hunch that maybe this happens if the inbound caller hangs up while
 the first Dial() is ringing, but I would've expected to see the first Dial
 (to 504 and 506) show up in the Master.csv log, and it's not there.  (The
 preceding line of the log is a call from almost an hour earlier).  In that
 case though I'd expect to see 1-CANCEL in the log instead.  Perhaps if
 the
 caller happens to hang up right between the two Dial() commands?..

 As an aside, the Set(CALLERID...) bit doesn't work.  The idea was to
 prepend
 a 9 so that a SIP user could use the redial feature of the phone's call
 log to return a missed call (automatically including the 9 for outside
 line).  Unfortunately the 9 does not get prepended.

 Thanks in advance for any and all advice!
 ~Brian

 --
  Brian Henning, Software Engineer

/\Pine Research Instrumentation
   //\\   5908 Triangle Drive
  ///\\\  Raleigh, NC 27617
   USA
||
||phone: 919.782.8320
  fax:   919.782.8323
  email: bhenn...@pineinst.com
 --



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[asterisk-users] Free Alarms sound

[amit salunkhe]

Dear All

Can anyone let me know where i can free sound file whcih i can use for
system monitoring alrams.

Regards
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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Sebastian Arcus]

On 05/09/2011 03:40 PM, Warren Selby wrote:
/snip



Thanks for the reply. No, I run tftpd directly from rc.local script
(on Slackware). That's fine - I just wanted to know I wasn't doing
something wrong. If everybody else is in the same boat - I'll just
be along for the ride then :D

Sebastian


When I used to have a Cisco 7941 phone in my home office, I only needed
the tftp server online if I made changes to the configuration.  The
phone itself worked fine without one being online all the time.


That's strange. Mine get stuck on the booting phase, looking for the 
tftp server, if they can't find it there. Even if I change the dhcpd 
option not to pass out any tftp server. Any ideas what did you configure 
differently?


Sebastian





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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Warren Selby]

On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus s...@open-t.co.uk wrote:

 That's strange. Mine get stuck on the booting phase, looking for the tftp
 server, if they can't find it there. Even if I change the dhcpd option not
 to pass out any tftp server. Any ideas what did you configure differently?

 Sebastian


All I can really think is that the 7940 and the 7941 use different firmwares
and configuration files, maybe there was some kind of change between the
two?  Does the phone never time out while looking for a server?

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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

[--[ UxBoD ]--]

- Original Message -
 On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
  Are you not seeing issues with *8 call pick up then ?
 
 Nope, I double checked it after seeing someone saying they had issues
 with it and it is fine on the installation I have.
 

Which release are you running as this is still open 
https://issues.asterisk.org/view.php?id=18654
-- 
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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[Sebastian Arcus]

On 05/09/2011 04:50 PM, Warren Selby wrote:

On Mon, May 9, 2011 at 10:48 AM, Sebastian Arcus s...@open-t.co.uk
mailto:s...@open-t.co.uk wrote:

That's strange. Mine get stuck on the booting phase, looking for the
tftp server, if they can't find it there. Even if I change the dhcpd
option not to pass out any tftp server. Any ideas what did you
configure differently?

Sebastian


All I can really think is that the 7940 and the 7941 use different
firmwares and configuration files, maybe there was some kind of change
between the two?


That's always a possibility.

Does the phone never time out while looking for a server?

Not as far as I can tell. It just goes in a loop. Even if it would 
eventually time out, 2-3 hours to boot a phone would be a bit much :-)


Sebastian

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Re: [asterisk-users] Asterisk 1.8 latest branch safe for production ?

[satish patel]

 Which release are you running as this is still open 
 https://issues.asterisk.org/view.php?id=18654
 -- 
 Thanks, Phil

I am using current SVN branch 1.8 and We aren't using above call pickup 
features. 

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Re: [asterisk-users] Cisco 7940 phone and tftpd provisioning - for ever?

[A J Stiles]

On Monday 09 May 2011, Cassius Smith wrote:
 On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote:
 Sebastian Arcus wrote:
  Cisco phones (at least the 7940) are supposed to be run with a tftp
  server available at all time
 
 That is my experience.  But, if you're running tftp under Linux, then
 it's probably spawned by xinetd and won't be running unless the service
 is requested.

 If you want the users to have access to ringtones and desktop images, they
 are dynamically loaded via tftp. So yes, you'll need to keep the tftp
 server running.

As far as I understand it, inetd / xinetd is just a wrapper which does some of 
the business of every server daemon:  whenever a request comes in on a 
listened-to port, x?inetd invokes an instance of some external program, with 
its STDIN, STDOUT and STDERR already connected transparently to the socket.  
So the server program is spared from having to care about protocols, 
sockets and forking, at the expense that the superserver daemon may take ever 
so slightly longer to do what it has to do on account of having to look stuff 
up in its config.  (Of course, if the external server is written in an 
interpreted language with poor support for sockets, using inetd / xinetd 
might work out just fractionally faster.)

But, surely, the only way to find out if a server is listening on a port is to 
send it a request?  And whether the server is monolithic with its own code to 
deal with incoming connections or started via inetd, it will still equally 
respond to that request.


-- 
AJS

Answers come *after* questions.

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[asterisk-users] conf syntax highlighting for gedit

[Naomi Rosenberg]

Hi,

It seems there is no .conf syntax highlighting script available for gedit. I'm 
thinking of putting one together myself, but don't want to reinvent the wheel. 

So I'm just enquiring if anyone knows of one that already exists that i've 
missed.

Thanks

Naomi Rosenberg

www.servicesforasterisk.co.uk


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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

[Olivier]

2011/5/9 randulo rand...@randulo.com

 On Mon, May 9, 2011 at 2:20 PM,  mgra...@mstvp.com wrote:
  Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.

 Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it
 not?


Yes, of course, all dual-mode phones support WiFi but :
1. I'm not certain those would work without any SIM-card inside
2. those are likely to be more expensive than WiFi-only handset.

See the last iPod touch which is marketed as a Sametime client is quite
cheeper than the iPhone.

To my knowledge, most Android-based WiFi-only machines are tablets.

Cheers
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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?

[Steve Underwood]

On 05/10/2011 12:55 AM, Olivier wrote:


2011/5/9 randulo rand...@randulo.com mailto:rand...@randulo.com

On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com
mailto:mgra...@mstvp.com wrote:
 Lots of Android handsets support wifi, like my G2, aka HTC DesireZ.

Wouldn't ANY modern one have wifi? That would be odd if it didn't,
would it not?


Yes, of course, all dual-mode phones support WiFi but :
1. I'm not certain those would work without any SIM-card inside
2. those are likely to be more expensive than WiFi-only handset.

See the last iPod touch which is marketed as a Sametime client is 
quite cheeper than the iPhone.


To my knowledge, most Android-based WiFi-only machines are tablets.
archos make cheap wifi only android devices, but I'm not sure of the 
small ones have the mic and speaker in the right place for a SIP call.


There are some very cheap Android phones around, while the wifi VoIP 
phones tend to be expensive.


Steve


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[asterisk-users] iax2 issue in asterisk

[satish patel]

Hey guys!

I have issue between iax vs iax2 following is my setup

asterisk-1.2 --IAXAsterisk-1.8

I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8 
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[asterisk-users] high PDD

[Abid Saleem]

Hi,
I am using Asterisk 1.4.17 for my C4 routing but I am experiencing a high pdd 
of around 20 seconds. Could you please help me to reduce it and what could be 
the reason. Thanks
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Re: [asterisk-users] iax2 issue in asterisk

[Gordon Henderson]

On Mon, 9 May 2011, satish patel wrote:



Hey guys!

I have issue between iax vs iax2 following is my setup

asterisk-1.2 --IAXAsterisk-1.8

I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8



Might you be missing

  requirecalltoken=no

in iax.conf in the 1.8 system for calls originating from the 1.2 system?

Gordon

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Re: [asterisk-users] iax2 issue in asterisk

[satish patel]

Awesome! 

root@:~# cat /etc/asterisk/iax.conf | grep requirecalltoken
; By setting 'requirecalltoken=no', call token validation becomes optional for
; that peer/user.  By setting 'requirecalltoken=auto', call token validation 
; can require it from this peer.  So, requirecalltoken is internally set to yes.
; requirecalltoken may only be used in peer/user/friend definitions,
; By default, 'requirecalltoken=yes'.

requirecalltoken=no

Also there was an other issue host=dynamic i set to host=x.x.x.x and it works!


 Date: Mon, 9 May 2011 18:45:05 +0100
 From: gordon+aster...@drogon.net
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] iax2 issue in asterisk
 
 On Mon, 9 May 2011, satish patel wrote:
 
 
  Hey guys!
 
  I have issue between iax vs iax2 following is my setup
 
  asterisk-1.2 --IAXAsterisk-1.8
 
  I am able to call from 1.8 to 1.2 over iax but no from asterisk 1.2 to 1.8
 
 
 Might you be missing
 
requirecalltoken=no
 
 in iax.conf in the 1.8 system for calls originating from the 1.2 system?
 
 Gordon
 
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Re: [asterisk-users] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

[Paul Hayes]

Hi,

It looks to me that the 401 unauth packets aren't getting back to the phones. 
Which suggests a network/router/nat issue rather than anything wrong with the 
asterisk or phone configuration.

Cheers,
Paul.



On 8 May 2011, at 01:59, GNUbie gnu...@gmail.com wrote:

 Hello all,
 
 I have installed the .deb packages of the Asterisk v1.8.3.3 from the
 upstream project on my Debian GNU/Linux Squeeze server and bought the
 Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
 exercise. After setting up everything and trying to fix this problem,
 I am still getting a 401 Unauthorized SIP message. So as of this
 writing, I still cannot successfully REGISTER to my Asterisk box.
 
 Below are the snippets of my Asterisk and SNOM 300 configurations
 including the logs for your reference.
 
 I hope anyone from this community can help me solve this problem. A
 HOWTO of a similar scenario will help a lot.
 
 Thank you in advance.
 
 Regards,
 
 GNUbie
 
 - - - ASTERISK v1.8.3.3 - - -
 
 [ /etc/asterisk/sip.conf ]
 
 [general]
 ...
 ...
 tlsenable=yes
 tlsbindaddr=0.0.0.0
 tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem
 tlscipher=ALL
 tlsclientmethod=tlsv1
 tlsbindport=5061
 externtlsport=5061
 externtcpport=5061
 tcpbindaddr=0.0.0.0
 tcpbindport=5061
 tcpenable=yes
 srvlookup=yes
 
 [361]
 username=361
 secret=***
 callerid=361-tls361
 mailbox=361@family
 context=family
 transport=tls
 port=5061
 type=friend
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=no
 nat=yes
 qualify=yes
 autoframing=yes
 encryption=yes
 
 *CLI core show version
 Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a
 x86_64 running Linux on 2011-04-22 17:50:44 UTC
 
 *CLI sip show settings
 
 Global Settings:
 
 UDP Bindaddress: 0.0.0.0:5060
 TCP SIP Bindaddress: 0.0.0.0:5060
 TLS SIP Bindaddress: 0.0.0.0:5061
 Videosupport: No
 Textsupport: No
 Ignore SDP sess. ver.: No
 AutoCreate Peer: No
 Match Auth Username: No
 Allow unknown access: No
 Allow subscriptions: Yes
 Allow overlap dialing: Yes
 Allow promsic. redir: No
 Enable call counters: No
 SIP domain support: Yes
 Realm. auth: No
 Our auth realm pbx.domain.com
 Use domains as realms: No
 Call to non-local dom.: Yes
 URI user is phone no: No
 Always auth rejects: Yes
 Direct RTP setup: No
 User Agent: Asterisk rocks!
 SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze
 SDP Owner Name: root
 Reg. context: (not set)
 Regexten on Qualify: No
 Caller ID: asterisk
 From: Domain:
 Record SIP history: Off
 Call Events: Off
 Auth. Failure Events: Off
 T.38 support: No
 T.38 EC mode: Unknown
 T.38 MaxDtgrm: -1
 SIP realtime: Disabled
 Qualify Freq : 6 ms
 Q.850 Reason header: No
 
 Network QoS Settings:
 ---
 IP ToS SIP: CS0
 IP ToS RTP audio: CS0
 IP ToS RTP video: CS0
 IP ToS RTP text: CS0
 802.1p CoS SIP: 4
 802.1p CoS RTP audio: 5
 802.1p CoS RTP video: 6
 802.1p CoS RTP text: 5
 Jitterbuffer enabled: Yes
 Jitterbuffer forced: No
 Jitterbuffer max size: 200
 Jitterbuffer resync: 1200
 Jitterbuffer impl: fixed
 Jitterbuffer log: No
 
 Network Settings:
 ---
 SIP address remapping: Enabled using externhost
 Externhost: pbx.domain.com
 externaddr: 11.22.33.44:0
 Externrefresh: 10
 Localnet: 192.168.101.0/255.255.255.0
 
 Global Signalling Settings:
 ---
 Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc)
 Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30
 Relax DTMF: No
 RFC2833 Compensation: No
 Symmetric RTP: No
 Compact SIP headers: No
 RTP Keepalive: 0 (Disabled)
 RTP Timeout: 15
 RTP Hold Timeout: 0 (Disabled)
 MWI NOTIFY mime type: application/simple-message-summary
 DNS SRV lookup: Yes
 Pedantic SIP support: Yes
 Reg. min duration 1800 secs
 Reg. max duration: 3600 secs
 Reg. default duration: 120 secs
 Outbound reg. timeout: 20 secs
 Outbound reg. attempts: 0
 Notify ringing state: Yes
 Include CID: No
 Notify hold state: No
 SIP Transfer mode: open
 Max Call Bitrate: 384 kbps
 Auto-Framing: No
 Outb. proxy: not set
 Session Timers: Refuse
 Session Refresher: uas
 Session Expires: 1800 secs
 Session Min-SE: 90 secs
 Timer T1: 3000
 Timer T1 minimum: 100
 Timer B: 192000
 No premature media: Yes
 Max forwards: 70
 
 Default Settings:
 -
 Allowed transports: UDP
 Outbound transport: UDP
 Context: default
 Force rport: No
 DTMF: rfc2833
 Qualify: 0
 Use ClientCode: No
 Progress inband: Never
 Language:
 MOH Interpret: default
 MOH Suggest:
 Voice Mail Extension: asterisk
 
 *CLI sip show peer 361
 
 * Name : 361
 Secret : Set
 MD5Secret : Not set
 Remote Secret: Not set
 Context : family
 Subscr.Cont. : Not set
 Language :
 AMA flags : Unknown
 Transfer mode: open
 CallingPres : Presentation Allowed, Not Screened
 Callgroup :
 Pickupgroup :
 MOH Suggest :
 Mailbox : 361@family
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit : 0
 Max forwards : 0
 Dynamic : Yes
 Callerid : 361-tls 361
 MaxCallBR : 384 kbps
 Expire : -1
 Insecure : 

[asterisk-users] Rates Importer Tool

[A E [Gmail]]

Hi All,

new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?

Thanks so much in advance
aeg
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Re: [asterisk-users] Rates Importer Tool

[Jason Aarons (AM)]

I know most billing software sell this as a monthly service.  You get cd-rom 
every month where they have collected the published tarrif tables filed with 
the FCC. You load it on the software to analyze call costs.   I'm guessing this 
is a lot of labor hours/manual work thus they charge for providing it.  In 
particular I am thinking of InforTel for Windows.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Monday, May 09, 2011 2:29 PM
To: Commercial and Business-Oriented Asterisk Discussion; Asterisk Users 
Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Rates Importer Tool

Hi All,

new to the list. Wondering if anyone has / knows of, a good rate importer tool 
that can be used to standardize and normalize the ratesheets / rate decks etc. 
obtained from various carriers so they can be analysed and imported into a DB 
or be saved as a CSV or something?

Thanks so much in advance
aeg

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Re: [asterisk-users] Rates Importer Tool

[A E [Gmail]]

On Mon, May 9, 2011 at 3:05 PM, Jason Aarons (AM) 
jason.aar...@dimensiondata.com wrote:

  I know most billing software sell this as a monthly service.  You get
 cd-rom every month where they have collected the published tarrif tables
 filed with the FCC. You load it on the software to analyze call costs.   I’m
 guessing this is a lot of labor hours/manual work thus they charge for
 providing it.  In particular I am thinking of InforTel for Windows.


That's interesting. Wasn't aware of such a thing...if these subscriptions
ad/or software are reasonably priced then we might still be interested in
having a look at it. What specific product of InforTel were you referring
to?
Thanks
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[asterisk-users] Need help defining a stackexchange (i.e. stackoverflow) for telephony

[Simon P. Ditner]

For those of that are fans of stackoverflow.com, and stackexchange.com, 
there's an effort to define a telephony stackexchange site. It's still in 
the definition phase. What it needs to move forwards is more votes on 
on/off topic questions, and perhaps some better questions to vote for or 
against.


If you're interested in helping out, or following the progress, visit:
http://area51.stackexchange.com/proposals/12932/telephony/

Cheers,
spd


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Re: [asterisk-users] Blacklist with *30

[Alejandro Cabrera Obed]

Dear, finally I implement the functionality code *94 in order to
access the blacklist menu from my own extension and put another
extension in the black list of Asterisk.

But after blacklisting a given extension, when I call from that
extension to my own extension the call always rings, it is not denied
by the blacklist.

Why could be the problem the blacklist doesn't work ???

Thanks a lot

2011/5/9 Dovid Bender asteriskus...@dovid.net:
 Try the Elastix forums.

 - Original Message - From: Alejandro Cabrera Obed
 aco1...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, May 09, 2011 15:35
 Subject: Re: [asterisk-users] Blacklist with *30


 Dear Dovis, I'm using Elastix and the dialplan comes with this line:

 *30,1,Goto(app-blacklist-add,s,1)

 Any idea ??? Thanks a lot.

 2011/5/9 Dovid Bender asteriskus...@dovid.net:

 Alejandro,

 What GUI are you using ? I don't think Asterisk comes with *30 to ban
 calls.

 Regards,

 Dovid

 - Original Message - From: Alejandro Cabrera Obed
 aco1...@gmail.com
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Friday, May 06, 2011 23:51
 Subject: [asterisk-users] Blacklist with *30


 Dear, when I dial *30 in order to get instructions to blacklist an
 extension, Idon't get the menu but I get a new dial tone.

 What happen please ??? What can I do to solve this ???

 Thanks a lot,

 Alejandro

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[asterisk-users] Really, really loud ringers

[Justin Sherrill]

Anyone have some recommended equipment for alerting people to calls in a noisy 
environment?

I have Polycom IP550 phones set up in some really noisy environments - our mine 
hoists - and they tend to drown out the ringers.  I'm using Clarity WR100s now. 
 They're analog devices, attached to Linksys PAP2T ATAs as part of a call group 
to get a loud (advertised as 95dB) ring out there, but it still could be 
louder.  Maybe a light-up option would be better.

The old phone system here had some huge loudspeakers that someone had wired 
right into the speakers of the old digital phones.  I haven't figured out yet 
if they need a different voltage, or even if they still work; they were not 
responding when I replaced the attached phones.

Justin C. Sherrill - American Rock Salt
p: 585-991-6825 f: 585-991-6926 c: 585-298-6826



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Re: [asterisk-users] Really, really loud ringers

[Andrew Latham]

On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill
justin.sherr...@americanrocksalt.com wrote:
 Anyone have some recommended equipment for alerting people to calls in a 
 noisy environment?

 I have Polycom IP550 phones set up in some really noisy environments - our 
 mine hoists - and they tend to drown out the ringers.  I'm using Clarity 
 WR100s now.  They're analog devices, attached to Linksys PAP2T ATAs as part 
 of a call group to get a loud (advertised as 95dB) ring out there, but it 
 still could be louder.  Maybe a light-up option would be better.

 The old phone system here had some huge loudspeakers that someone had wired 
 right into the speakers of the old digital phones.  I haven't figured out yet 
 if they need a different voltage, or even if they still work; they were not 
 responding when I replaced the attached phones.

 Justin C. Sherrill - American Rock Salt
 p: 585-991-6825 f: 585-991-6926 c: 585-298-6826

Look for ADA devices.  The Disabilities Act has encouraged some nice
products. And it allows for you to get ISDN service anywhere in the
country...

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

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[asterisk-users] Voicemail Configuration

[John Marvin]

Hi,

I can't figure out a way of achieving what I want to do with the 
voicemail feature. I thought I'd ask here to see if there are any 
creative solutions that I have not considered.


What I want to do is have a message that says Press 1 for Dick, or 2 
for Jane. Then, depending on which number is pressed, have the caller 
sent to the appropriate voicemail box. I know how to do that without any 
problem.


I want to keep repeating that message after a timeout, i.e. not send the 
caller to a default voicemail box if nothing is pressed. I can handle 
that also.


However, I want to record what is said during that time and send it to 
a third voicemail box once the caller hangs up without having pressed 1 
or 2. I want this ability in order to handle robot callers. I'm not 
interested in most robot calls, but sometimes they contain useful 
information (school closures, your order is ready ...  etc.). The 
problem is if I just timeout and send the caller to a 3rd mailbox I will 
usually have lost the beginning of the message, and I don't want a short 
timeout because I want someone to be able to listen to the prompt and 
make a proper choice (if they're not a robot).


Any ideas? Is this possible?

Thanks,

John


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[asterisk-users] Call ends when using SendDTMF(*)

[m j]

I'm not sure why but my call is being ended when I SendDTMF(*).


I'm using agi to originate a call and set the context,extension,priority to 
test,1,1 respectively. I've got the following in my extensions.conf:


[test]
exten = 1,1,Answer();
  same =n,Wait(5);
  same =n,Verbose(1, Sending *);
  same =n,SendDTMF(*,500);
  same =n,Verbose(1, Sent *);
  same =n,Wait(5);
  same =n,Hangup();



I've set the following in features.conf:

[featuremap]
...
disconnect = ***0


I've also set the following in agents.conf:

[agents]
endcall=no
enddtmf=***


Am I missing something? I'm totally lost.

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Re: [asterisk-users] Voicemail Configuration

[Roger Burton West]

On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote:
However, I want to record what is said during that time and send it
to a third voicemail box once the caller hangs up without having
pressed 1 or 2.

You could use Monitor to record the whole call, then use an AGI to do
something with it on hangup if the other conditions haven't been
satisfied...?

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Re: [asterisk-users] Voicemail Configuration

[John Marvin]

On 5/9/2011 3:08 PM, Roger Burton West wrote:


You could use Monitor to record the whole call, then use an AGI to do
something with it on hangup if the other conditions haven't been
satisfied...?



I understand how to do the first part, and I at least understand that I 
could do something fancy with the AGI capability. But what I don't know 
is how I can take the recording and insert it into a voicemail box such 
that it can be retrieved through the normal VoiceMailMain mechanism.


Would the asterisk voicemail app dynamically notice something new being 
dropped into the voicemail mbox directory? Would it only be noticed once 
Asterisk is restarted? Most importantly, would it send out the notifies 
to the phone associated with that voicemail box? I can probably fake 
the last part if necessary, but making the voicemail retrievable through 
the normal voicemail mechanism is what I really need to achieve.


Thanks,

John


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Re: [asterisk-users] [SOT] Virtualising Asterisk

[Jan Bakuwel]

Hi Phil,

Happily running with the following here:

dom0: Debian Lenny Xen 3.2-1 2.6.26-2-xen-amd64
domU: Asterisk 1.4 Debian Lenny 2.6.26-2-xen-amd64
domU: Asterisk 1.6 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-aware
kernel)
domU: Asterisk 1.8 Debian Squeeze 2.6.32-5-amd64 (which is a Xen-aware
kernel)

All kernels are stock kernels from the Debian repositories. I'm using
dahdi-dummy for conferencing.

I haven't been running very long with this (new) set up but so far all
looks good. You obviously need to make sure that you have sufficient
resources for your domU's. You might consider CPU pinning, possibly
dedicated discs depending on your situation.

cheers,
Jan



On 08/05/11 03:24, --[ UxBoD ]-- wrote:
 I know a lot has changed over the past couple of years, and even
 monthly, and that Asterisk running within a virtualised environment is
 very happy indeed. If one would only be using SIP/IAX would Xen/KVM be
 the best solution ? / or perhaps VServer/LXC maybe advantageous due to
 binary hashing.  Your thoughts would be very welcome.
 -- 
 Thanks, Phil


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Re: [asterisk-users] asterisk syntax highlighting for gedit

[Matt Riddell]

On 10/05/11 2:32 AM, Naomi Rosenberg wrote:

Hi,

Apologies if this is a duplicate - been having mail server issues and I don't 
think I managed to send it when I tried this morning.

It seems there is no .conf syntax highlighting script available for gedit. I'm 
thinking of putting one together myself, but don't want to reinvent the wheel.

So I'm just enquiring if anyone knows of one that already exists that i've 
missed.


There have been some editor highlighting stuff submitted to the list in 
the past but as far as I'm aware, nothing for gedit.  If you do put 
something together post it here and I'll post it to the Daily Asterisk News.


--
Cheers,

Matt Riddell
___

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/cc.php (Call Centre Solutions)

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Re: [asterisk-users] Rates Importer Tool

[Markus]

Hi,

 new to the list. Wondering if anyone has / knows of, a good rate importer
 tool that can be used to standardize and normalize the ratesheets / rate
 decks etc. obtained from various carriers so they can be analysed and
 imported into a DB or be saved as a CSV or something?

I'm using a2billing (http://www.a2billing.org), a free of charge and
complete call shop web-based PHP application for Asterisk. Very buggy
overall but I couldn't find anything better (which is free of charge) yet.
Anyway, it gets the job done. I'm uploading the rate sheets to a Linux box
respectively download them directly to the box and then use a shell script
for each provider's rate sheet to properly order to fit into the a2billing
format, a la:

wget http://www.provider.com/rates/premium.csv
cat premium.csv | grep \1\,\1\  temp.csv
cat temp.csv | cut -d , -f 3  tempcode.csv
cat temp.csv | cut -d , -f 1  tempdest.csv
cat temp.csv | cut -d , -f 6  temprate.csv
paste -d , tempcode.csv tempdest.csv temprate.csv temprate.csv | tail
-n+2 | sed 's/^\/\00/g'  Provider.PREMIUM.$DATE.csv
unix2dos Provider.PREMIUM.$DATE.csv
scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
rm temp.csv tempcode.csv tempdest.csv temprate.csv premium.csv

This fetches and orders the rate sheet properly and uploads it to my home.
Then I just log into a2billing and upload the rate sheet there, done with
a few clicks. But you could also create a new ratecard directly in MySQL
and store the rates there directly if you want to. a2b stores all rates in
a MySQL DB. You can then choose least cost routing between different
providers etc.

Also, when a provider only supplies XLS instead of CSV, I use a script
like the following, utilizing xlhtml:

xlhtml -csv -xp:0 Provider.xls | cut -d , -f 1-3  temp.csv
cat temp.csv | cut -d , -f 3  temp2.csv
paste -d , temp.csv temp2.csv | tail -n+3 | sed 's/^/00/g' 
Provider.PREMIUM.$DATE.csv
unix2dos Provider.PREMIUM.$DATE.csv
scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
rm temp.csv
rm temp2.csv

Regards.





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Re: [asterisk-users] Rates Importer Tool

[A E [Gmail]]

On Mon, May 9, 2011 at 7:58 PM, Markus unive...@truemetal.org wrote:

 Hi,

  new to the list. Wondering if anyone has / knows of, a good rate importer
  tool that can be used to standardize and normalize the ratesheets / rate
  decks etc. obtained from various carriers so they can be analysed and
  imported into a DB or be saved as a CSV or something?

 I'm using a2billing (http://www.a2billing.org), a free of charge and
 complete call shop web-based PHP application for Asterisk. Very buggy
 overall but I couldn't find anything better (which is free of charge) yet.
 Anyway, it gets the job done. I'm uploading the rate sheets to a Linux box
 respectively download them directly to the box and then use a shell script
 for each provider's rate sheet to properly order to fit into the a2billing
 format, a la:

 wget http://www.provider.com/rates/premium.csv
 cat premium.csv | grep \1\,\1\  temp.csv
 cat temp.csv | cut -d , -f 3  tempcode.csv
 cat temp.csv | cut -d , -f 1  tempdest.csv
 cat temp.csv | cut -d , -f 6  temprate.csv
 paste -d , tempcode.csv tempdest.csv temprate.csv temprate.csv | tail
 -n+2 | sed 's/^\/\00/g'  Provider.PREMIUM.$DATE.csv
 unix2dos Provider.PREMIUM.$DATE.csv
 scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
 rm temp.csv tempcode.csv tempdest.csv temprate.csv premium.csv

 This fetches and orders the rate sheet properly and uploads it to my home.
 Then I just log into a2billing and upload the rate sheet there, done with
 a few clicks. But you could also create a new ratecard directly in MySQL
 and store the rates there directly if you want to. a2b stores all rates in
 a MySQL DB. You can then choose least cost routing between different
 providers etc.

 Also, when a provider only supplies XLS instead of CSV, I use a script
 like the following, utilizing xlhtml:

 xlhtml -csv -xp:0 Provider.xls | cut -d , -f 1-3  temp.csv
 cat temp.csv | cut -d , -f 3  temp2.csv
 paste -d , temp.csv temp2.csv | tail -n+3 | sed 's/^/00/g' 
 Provider.PREMIUM.$DATE.csv
 unix2dos Provider.PREMIUM.$DATE.csv
 scp -P 450 Provider.PREMIUM.$DATE.csv u...@athome.dyndns.org:samba/_rates/
 rm temp.csv
 rm temp2.csv

 Regards.

 Hello Markus,
thanks for sharing. I am looking into A2Billing myself at the moment. Don't
really need most of the functionality in it, but will check out its rates
import tool although I'm not sure it can handle rate updates but seems like
something to check out. Although like I'd said in my OP, this is mostly for
the business people to be able to visualize the rates and analyse them them
more than anything else and judging from the extra hacking involved in
getting these rates to be ready to be imported into A2Billing even seems too
complicated for the business people be able to do on their own, and I don't
want to have to sit and normalize it for them every time there's a rate
update. But will look more into this. Thanks again for putting up your
script and trying to help out :)
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Re: [asterisk-users] QueueCallerAbandon is not triggering after 1.8.3.3...

[Louis Carreiro]

Has anyone else noticed this?

v/r,
Me



On Fri, May 6, 2011 at 12:11 PM, Louis Carreiro carreir...@gmail.comwrote:

 Has anyone else noticed that QueueCallerAbandon is not showing up in the
 AMI after the 1.8.3.3? Am I missing something? I'm getting what seems like
 everything else but QueueCallerAbandon.

 v/r,
 Me


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