from:"Aksel Celasun"

Re: [asterisk-users] Problems during calls

2011-10-19 Thread Aksel Celasun

Thank you for replying also,

I will as you and Zeeshan suggest, look at the firewall issue first, i have
been suspecting
network issue, because i cannot see anything in the log, so again thanks!

Best regards

Aksel

Fra: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] på vegne av Sammy Govind
[govoi...@gmail.com]
Sendt: 19. oktober 2011 08:48
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

Hi,

Call getting silenced in the middle definitely point to RTP but I think the
redialling part should be considered as well. I think that Phones are loosing
registrations or like Zeeshan mentioned could be getting blocked by firewall -
Asterisk server's firewall as well as any other firewall in front of server
should be inspected for sessions/connections limit etc.

--
Regards,
Sammy

On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun
mailto:ak...@abacus-it.no>> wrote:
Thank you for the reply.

The Asterisk is behind a firewall, but not in a dmz, been thinking of placing
it in a dmz soon, maybe that will solve the problem.
Or else, I will try your guide with wireshark.

Thank you very much.

Best regards

Aksel

Fra:
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>

[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
På vegne av VisionVoIP
Sendt: 18. oktober 2011 16:31

Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

I can only make another guess. If your system is behind a firewall, try adding
'insecure=invite' in your sip.conf's general section.

To troubleshoot such cases, do a tcpdump trace like this:

1. Run tcpdump on your server before making a call. Use command "tcpdump port
5060 -s0 -w dumpfile.pcap".
2. When you notice the silence problem, hangup, and stop the trace using CTRL+C.
3. Copy the dumpfile.pcap to a computer with Wireshark installed.
4. Open this file in Wireshark and follow my blog at
http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/
5. Given that you know some basics of how VoIP works over SIP, the wireshark
graph will tell you if RTP was still flowing when it was silent. It probably
is, but to which IP address.

My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP
address, or stop flowing, or is blocked by the router.

A good solution is to put your Asterisk server in DMZ mode.

There can be many other guesses, but the above is a good start.
--

Zeeshan A Zakaria

PBX - visionvoip.com<http://visionvoip.com>
Blog - ilovetovoip.com<http://ilovetovoip.com>

On 18/10/2011 10:02, Aksel Celasun wrote:
Thank you for replying

My sip.conf is set to no on canreinvite

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Re: [asterisk-users] Problems during calls

2011-10-18 Thread Aksel Celasun

Thank you for the reply.

The Asterisk is behind a firewall, but not in a dmz, been thinking of placing
it in a dmz soon, maybe that will solve the problem.
Or else, I will try your guide with wireshark.

Thank you very much.

Best regards

Aksel

Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av VisionVoIP
Sendt: 18. oktober 2011 16:31
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

I can only make another guess. If your system is behind a firewall, try adding
'insecure=invite' in your sip.conf's general section.

To troubleshoot such cases, do a tcpdump trace like this:

My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP
address, or stop flowing, or is blocked by the router.

A good solution is to put your Asterisk server in DMZ mode.

There can be many other guesses, but the above is a good start.
--

Zeeshan A Zakaria

PBX - visionvoip.com
Blog - ilovetovoip.com

On 18/10/2011 10:02, Aksel Celasun wrote:
Thank you for replying

My sip.conf is set to no on canreinvite

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Re: [asterisk-users] Problems during calls

2011-10-18 Thread Aksel Celasun

Thank you for replying


My sip.conf is set to no on canreinvite



[general]
context=default
allowguest=yes
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
disallow=all
allow=alaw
;allow=ulaw
;allow=gsm
language=en
trustrpid = yes
sendrpid = yes
progressinband=never
useragent=TS200 PBX
promiscredir = no
usereqphone = no
dtmfmode = rfc2833
compactheaders = no
videosupport=no
maxcallbitrate=96
shrinkcallerid=yes
rtptimeout=60
rtpholdtimeout=300
rtpkeepalive=29
rtcachefriends=yes
recordhistory=yes
nat=yes
canreinvite=no
limitonpeers=yes
limitonpeer=yes
allowsubscribe=yes


Maybe there is something with the sip client, qualify=yes?

;Sentralbord
[501]
type=friend
secret=501
host=dynamic
context=phones
mailbox=501@defualt
callerid=Sentralbord Abacus-IT
qualify=yes

Thank you in advance.

Regards

Aksel Celasun


Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av VisionVoIP
Sendt: 18. oktober 2011 15:49
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Problems during calls

I have similar problem at my home extension, but for that I know my phone's 
speaker is defective, and tapping it against the desk or wall fixes the problem.

However in your case probably it is sip configuration (sip.conf or an included 
file), where canreinvite=yes where it should be canreinvite=no, either in 
general section, or in the extension settings.

--

Zeeshan A Zakaria

PBX - visionvoip.com
Blog - ilovetovoip.com

On 18/10/2011 09:35, Aksel Celasun wrote:
Hello dear list.

We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when 
making calls, that the calls become silent.
Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the 
conversation.
When we then hangup, and redial immediately, the calls do not go through, we 
then have to try redial a couple of times, and then It suddenly gets through.
There is nothing in the verbose log in Asterisk -r.

SIP HW is Snom and Different types of Cisco.

Anyone got an idea? Or at lest know how to dig deeper in logs?

Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

L.Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

Se denne månedens gode tilbud fra Abacus IT 
AS<http://www.abacus-it.no/systeml%F8sninger/kampanjer>






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[asterisk-users] Problems during calls

2011-10-18 Thread Aksel Celasun

Hello dear list.

We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when 
making calls, that the calls become silent.
Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the 
conversation.
When we then hangup, and redial immediately, the calls do not go through, we 
then have to try redial a couple of times, and then It suddenly gets through.
There is nothing in the verbose log in Asterisk -r.

SIP HW is Snom and Different types of Cisco.

Anyone got an idea? Or at lest know how to dig deeper in logs?

Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

L.Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no

Se denne månedens gode tilbud fra Abacus IT 
AS


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Re: [asterisk-users] Asterisk 1.6 and multiple parking

2010-06-28 Thread Aksel Celasun

Hello there


You should have a look at features.conf


Regards Aksel

Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 28. juni 2010 21:39
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Asterisk 1.6 and multiple parking

Hi,

One of the big features of 1.6 was described as multi-tenant parking.  
Basically, parking people in different "lots" so the sales dept. could only 
pick up their calls, and tech support theirs and no mix up was possible.

I can only find the original announcement and others asking the same question. 
Is there some sort of sample conf file of how I would get this functionnal on 
the latest 1.6.x?

Regards,

Mike








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[asterisk-users] Unregister and register SIP phones by using num pad on phones?

2010-06-22 Thread Aksel Celasun

Hello dear list.


A couple of years ago, I worked with a Alcatel IP pbx and Alcatel Sip phones, 
and we had the opportunity
to unregister user  by typing *-a number and -* again, ex * 99 *, and then the 
phone number/sip extension was unavailable, and
all of the calls to that extension was redirected to the receptionist.

When the user came back and wanted to register her sip account/extension, the 
user typed in a similar "code" ex * 99 * and internal sip, and voila,
Extension is online again. This was very useful regarding when users changed 
offices and so on, they didn't have to carry their phones, they just 
unregistered and
Later on registered themselves on the other office.

Are there any similar options on Asterisk, or is this more or less HW related?
Currently testing SNOM m300,Cisco spa525, Cisco spa520, and grandstrem gxp 3000.

Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun

Hello, and thank you for your response.

When I push transfer, the buttons with the function "transfer" disappears, and 
then I enter the sip number,
Wait 10 seconds and then it transfers with the MOH in the background, when the 
connection/channel is made,
Then transfer button is revealed again suddenly, and then I can push transfer 
again, and it transfers... =).

I'm gonna try with callback function when no-answer, and the hangup option 
which you mentioned.

Best regards

Aksel

-Opprinnelig melding-
Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Ira
Sendt: 21. juni 2010 19:16
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] features.conf - parkedcalls - transfer

At 12:27 AM 6/21/2010, you wrote:

>Almost 10 seconds, before the transfer to sip 
>200 is made, can I reduce that timer?
>
>And I can't see any button on the Cisco phone 
>which will function like a "direct transfer now", do I have to wait.?

On my Aastra phones, I press Transfer 101 
Transfer.  So you might just try hanging up or pressing transfer again.

Ira 

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Re: [asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun



>>And I can't see any button on the Cisco phone which will function like a 
>>"direct transfer now", do I have to wait...?
Thank you for your reply.

In my Dialplan menu on the SPA525g, I have a field where the input are, and I 
must say, I don't know if this is the right one, but the field contains this:
Dial plan:  (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
What should I edit, if this is the right code

>On the Cisco 79xx series phones, you would modify a config file called 
>dialplan.xml.  I think on the SPA5xx series you can configure this parameter 
>either in the main config file or from the web i>nterface.  You need to look 
>for something like "dialplan" or "dial plans", etc.  This controls the timeout 
>when entering digits.

--
Thanks,
--Warren Selby
http://www.selbytech.com

Best regards

Aksel
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[asterisk-users] features.conf - parkedcalls - transfer

2010-06-21 Thread Aksel Celasun

Hello dear list.


I am having issues on parkedcalls.

I am using a Cisco SPA525G as a test phone, and I have the transfer button 
there when I am in a call,
But when I want to transfer the current call I am in, I push the transfer 
button, and onscreen I se "Enter Number", and if I enter ex sip 200, I have to 
wait
Almost 10 seconds, before the transfer to sip 200 is made, can I reduce that 
timer?
And I can't see any button on the Cisco phone which will function like a 
"direct transfer now", do I have to wait...?

And, secondly, is there a another way to do transfer/send to another sip phone?
Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401 and 
the Sip phone will directly call SIP401?


Default features.conf context.


Thank you.


Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri||'

2010-06-18 Thread Aksel Celasun

Thank you for the info.
As I wrote to Warren GotoIfTime was easy to use and seemed more flexible,
Got it working now! Perfect!

Only one thing left now, and my system is pretty much ready for live testing,
Surely easy for the user list, so it will come in another mail soon, after I 
have done 
Some more research. (how the receptionist can transfer calls to SIP extensions 
internally)


Best regards 

Aksel

-Opprinnelig melding-
Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tilghman Lesher
Sendt: 18. juni 2010 18:01
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' 
tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'

On Friday 18 June 2010 09:49:39 Warren Selby wrote:
> On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun  wrote:
> >  Minor edit on the include => nighttime|12:30-8:00|mon-fri|*|*
> >
> > Correct now.
>
> This isn't how you do time based checks in asterisk.  Lookup the
> application "GotoIfTime".

Actually, it is an old method that still works, but as Warren mentioned, you
should endeavor to switch to using GotoIfTime, as there's a nasty race
condition inherent in using timed includes.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri||'

2010-06-18 Thread Aksel Celasun

Hi again

Thank you Warren, GotoIfTime was  the deal!
And easy to use!
Gr8.


Best regards.


Aksel

Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Warren Selby
Sendt: 18. juni 2010 16:50
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Error trying to add context: Context 'internal' 
tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'

On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun 
mailto:ak...@abacus-it.no>> wrote:
Minor edit on the include => nighttime|12:30-8:00|mon-fri|*|*
Correct now.


This isn't how you do time based checks in asterisk.  Lookup the application 
"GotoIfTime".

--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri||'

2010-06-18 Thread Aksel Celasun

Minor edit on the include => nighttime|12:30-8:00|mon-fri|*|*
Correct now.

Fra: Aksel Celasun
Sendt: 18. juni 2010 14:30
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Error trying to add context: Context 'internal' tries to include 
nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'


Hello again dear list.


Could you please help with this?

Thank you for all support, you are great, and i am now at a  late stage in the 
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I 
have a playback function there.
But CLI reports:

CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes: 
Context 'internal' tries to include nonexistent context 
'nighttime|12:30-8:00|mon-fri|*|*'


Extensions.conf
[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => hovedmeny


[internal]
include => to_SIPtrunk
include => nighttime|12:30-8:00|mon-fri|*|*

exten => _10X,1,NoOp()
exten => _10X,n,Dial(SIP/${EXTEN},10)
exten => _10X,n,Playback(kuntiltestt_)
;exten => _10X,n,Playback(vm-nobodyavail&tt-monkeysintro&tt-monkeys)
exten => _10X,n,Hangup()


exten => 4767209600,1,NoOp();
exten => 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209600,n,Dial(SIP/501,5);
;exten => 4767209600,n,Background(velkommen_abacus&tast123vent_);
;exten => 4767209600,n,WaitExten;
;exten => 4767209600,1,Dial(SIP/200,15);
;exten => 4767209600,1,Goto(submenu,s,1);
exten => 4767209600,n,Playback(kuntiltestt_);
exten => 4767209600,n,Hangup();



[hovedmeny]
exten => 501,1,Answer
exten => 501,n,Wait(2)
exten => 501,n,Playback(velkommen_abacus)
exten => 501,n,Set(Loop=0)
exten => 501,n,While($[${Loop} < 3])
exten => 501,n,Background(tast123vent_)
exten => 501,n,WaitExten(5)
exten => 501,n,Set(Loop=$[${Loop}+1])
exten => 501,n(LoopEnd),EndWhile()
exten => 501,n,Hangup()

exten => 1,1,Playback(tt-weasels)
exten => 1,2,Dial(SIP/200,10,rg)
exten => 1,3,Hangup()

exten => 2,1,Playback(tt-monkeys)
exten => 2,n,Dial(SIP/302,60,rg)
exten => 2,n,Hangup()

exten => 3,1,Dial(SIP/402,60,rg)
exten => 3,n,Hangup
exten => 9,n,Hangup()

exten => i,1,Set(Loop=$[${Loop}+1])
exten => i,n,Goto(LoopEnd)

exten => t,1,Set(Loop=$[${Loop}+1])
exten => t,n,Goto(LoopEnd)


[nighttime]
exten => s,1,Wait(2);
exten => s,n,Playback(tt-somethingwrong);
exten => s,n,Hangup;



[to_SIPtrunk]
exten => _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten => _0, 1, Macro(dial-trunk-sip,${EXTEN});


[incoming]
exten => s,1,Noop();
exten => s,n,Verbose(Call ${EXTEN});
exten => s,n,Dial(SIP/501);
exten => s,n,Hangup();


[macro-dial-trunk-sip]
exten => s,1,Noop(${ARG1},${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=67209600)
exten => s,n,Dial(SIP/phonect_01/${ARG1})
exten => s,n,Hangup
exten => s,n,MacroExit


Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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[asterisk-users] Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri||'

2010-06-18 Thread Aksel Celasun


Hello again dear list.


Could you please help with this?

Thank you for all support, you are great, and i am now at a  late stage in the 
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I 
have a playback function there.
But CLI reports:

CLI
[Jun 18 14:20:22] WARNING[2287]: pbx.c:9542 ast_context_verify_includes: 
Context 'internal' tries to include nonexistent context 
'nighttime|12:30-8:00|mon-fri|*|*'


Extensions.conf
[general]
autofallthrough=yes

[default]

[incoming_calls]

[phones]
include => internal
include => hovedmeny


[internal]
include => to_SIPtrunk
include => nighttime

exten => _10X,1,NoOp()
exten => _10X,n,Dial(SIP/${EXTEN},10)
exten => _10X,n,Playback(kuntiltestt_)
;exten => _10X,n,Playback(vm-nobodyavail&tt-monkeysintro&tt-monkeys)
exten => _10X,n,Hangup()


exten => 4767209600,1,NoOp();
exten => 4767209600,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209600,n,Dial(SIP/501,5);
;exten => 4767209600,n,Background(velkommen_abacus&tast123vent_);
;exten => 4767209600,n,WaitExten;
;exten => 4767209600,1,Dial(SIP/200,15);
;exten => 4767209600,1,Goto(submenu,s,1);
exten => 4767209600,n,Playback(kuntiltestt_);
exten => 4767209600,n,Hangup();



[hovedmeny]
exten => 501,1,Answer
exten => 501,n,Wait(2)
exten => 501,n,Playback(velkommen_abacus)
exten => 501,n,Set(Loop=0)
exten => 501,n,While($[${Loop} < 3])
exten => 501,n,Background(tast123vent_)
exten => 501,n,WaitExten(5)
exten => 501,n,Set(Loop=$[${Loop}+1])
exten => 501,n(LoopEnd),EndWhile()
exten => 501,n,Hangup()

exten => 1,1,Playback(tt-weasels)
exten => 1,2,Dial(SIP/200,10,rg)
exten => 1,3,Hangup()

exten => 2,1,Playback(tt-monkeys)
exten => 2,n,Dial(SIP/302,60,rg)
exten => 2,n,Hangup()

exten => 3,1,Dial(SIP/402,60,rg)
exten => 3,n,Hangup
exten => 9,n,Hangup()

exten => i,1,Set(Loop=$[${Loop}+1])
exten => i,n,Goto(LoopEnd)

exten => t,1,Set(Loop=$[${Loop}+1])
exten => t,n,Goto(LoopEnd)


[nighttime]
exten => s,1,Wait(2);
exten => s,n,Playback(tt-somethingwrong);
exten => s,n,Hangup;



[to_SIPtrunk]
exten => _[2-9]XXX, 1, Macro(dial-trunk-sip,${EXTEN});
exten => _0, 1, Macro(dial-trunk-sip,${EXTEN});


[incoming]
exten => s,1,Noop();
exten => s,n,Verbose(Call ${EXTEN});
exten => s,n,Dial(SIP/501);
exten => s,n,Hangup();


[macro-dial-trunk-sip]
exten => s,1,Noop(${ARG1},${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=67209600)
exten => s,n,Dial(SIP/phonect_01/${ARG1})
exten => s,n,Hangup
exten => s,n,MacroExit


Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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Re: [asterisk-users] Automatic attendant - Error in CLI.

2010-06-18 Thread Aksel Celasun

Ah, I missed the "comma", thank you, and thank you Tzafrir Cohen!


Best regards

Aksel

-Opprinnelig melding-
Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Doug Lytle
Sendt: 18. juni 2010 11:41
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Automatic attendant - Error in CLI.

Aksel Celasun wrote:
>
> This should be:
> exten =>  501,n(LoopEnd),EndWhile
>
> I don't understand, i do have the same thing you wrote above.
>

The difference between yours and his is that you had a n,(LoopEnd) and 
it should be n(LoopEnd)

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] Automatic attendant - Error in CLI.

2010-06-18 Thread Aksel Celasun



> Extensions.conf
> [mainmenu]
> exten => 501,1,Answer
> exten => 501,n,Wait(2)
> exten => 501,n,Playback(velkommen_abacus)
> exten => 501,n,Set(Loop=0)
> exten => 501,n,While($[${Loop} < 3])
> exten => 501,n,Background(tast123vent_)
> exten => 501,n,WaitExten(5)
> exten => 501,n,Set(Loop=$[${Loop}+1])
> exten => 501,n,(LoopEnd),EndWhile

This should be:
exten => 501,n(LoopEnd),EndWhile

I don't understand, i do have the same thing you wrote above.

 
> Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
> Verbosity is at least 3
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Executing [...@phones:1] Answer("SIP/301-0248", "") in new stack
> -- Executing [...@phones:2] Wait("SIP/301-0248", "2") in new stack
> -- Executing [...@phones:3] Playback("SIP/301-0248", 
> "velkommen_abacus") in new stack
> --  Playing 'velkommen_abacus.slin' (language 'en')
> -- Executing [...@phones:4] Set("SIP/301-0248", "Loop=0") in new stack
> -- Executing [...@phones:5] While("SIP/301-0248", "1") in new stack
> -- Executing [...@phones:6] BackGround("SIP/301-0248", 
> "tast123vent_") in new stack
> --  Playing 'tast123vent_.slin' (language 'en')
> -- Executing [...@phones:7] WaitExten("SIP/301-0248", "5") in new 
> stack
> -- Timeout on SIP/301-0248, continuing...
> -- Executing [...@phones:8] Set("SIP/301-0248", "Loop=1") in new stack
> [Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No 
> application '' for extension (phones, 501, 9)

You put '(LoopEnd)' in the place for the application. Hence empty
application with 'LoopEnd' as its input.

>   == Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'

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

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[asterisk-users] Automatic attendant - Error in CLI.

2010-06-18 Thread Aksel Celasun

Hello dear list.


I am currently working on a Automatic attendant, and the core things work, but 
I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The setup here is when 301 calls 501, and 301 doesn't enter an extension, it 
will go in loop, 3 times, and then hangup...Can't get that working.


Could someone please help me?

Extensions.conf
[mainmenu]
exten => 501,1,Answer
exten => 501,n,Wait(2)
exten => 501,n,Playback(velkommen_abacus)
exten => 501,n,Set(Loop=0)
exten => 501,n,While($[${Loop} < 3])
exten => 501,n,Background(tast123vent_)
exten => 501,n,WaitExten(5)
exten => 501,n,Set(Loop=$[${Loop}+1])
exten => 501,n,(LoopEnd),EndWhile
exten => 501,n,Hangup()

exten => 1,1,Playback(tt-weasels)
exten => 1,2,Dial(SIP/200,10,rg)
exten => 1,3,Hangup()

exten => 2,1,Playback(tt-monkeys)
exten => 2,n,Dial(SIP/302,60,rg)
exten => 2,n,Hangup()

exten => 3,1,Dial(SIP/402,60,rg)
exten => 3,n,Hangup
exten => 9,n,Hangup()

exten => i,1,Set(Loop=$[${Loop}+1])
exten => i,n,Goto(LoopEnd)

exten => t,1,Set(Loop=$[${Loop}+1])
exten => t,n,Goto(LoopEnd)


CLI Output

Connected to Asterisk 1.6.2.6 currently running on asterisk (pid = 2467)
Verbosity is at least 3
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Answer("SIP/301-0248", "") in new stack
-- Executing [...@phones:2] Wait("SIP/301-0248", "2") in new stack
-- Executing [...@phones:3] Playback("SIP/301-0248", 
"velkommen_abacus") in new stack
--  Playing 'velkommen_abacus.slin' (language 'en')
-- Executing [...@phones:4] Set("SIP/301-0248", "Loop=0") in new stack
-- Executing [...@phones:5] While("SIP/301-0248", "1") in new stack
-- Executing [...@phones:6] BackGround("SIP/301-0248", "tast123vent_") 
in new stack
--  Playing 'tast123vent_.slin' (language 'en')
-- Executing [...@phones:7] WaitExten("SIP/301-0248", "5") in new stack
-- Timeout on SIP/301-0248, continuing...
-- Executing [...@phones:8] Set("SIP/301-0248", "Loop=1") in new stack
[Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No 
application '' for extension (phones, 501, 9)
  == Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'
asterisk*CLI>

sip.conf regarding sip 501

[501]
type=friend
secret=XX
host=dynamic
context=phones
mailbox=...@default
callerid=Sentralbord
qualify=yes



Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner

Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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[asterisk-users] error message in CLI regarding SET Timeout

2010-06-17 Thread Aksel Celasun

Hello!


Does anybody know why i get the following error in CLI regarding the timeout 
option, when I dial sip ext 501?
I get the message playing in the background, but the cli output confuses me.
Running asterisk 1.6 on centos.

== Using SIP RTP CoS mark 5
-- Executing [...@phones:1] Answer("SIP/301-0203", "") in new stack
-- Executing [...@phones:2] Set("SIP/301-0203", "TIMEOUT(5)=timeout") 
in new stack
[Jun 17 10:35:31] ERROR[30876]: func_timeout.c:184 timeout_write: Unknown 
timeout type specified.
-- Executing [...@phones:3] Set("SIP/301-0203", "Timeout(response)=30") 
in new stack
[Jun 17 10:35:31] ERROR[30876]: pbx.c:3386 ast_func_write: Function Timeout not 
registered
-- Executing [...@phones:4] BackGround("SIP/301-0203", 
"velkommen_abacus&tast123vent_") in new stack



extensions.conf snipped.

exten => 501,1,Answer
exten => 501,n,Set(Timeout(5)=timeout)
exten => 501,n,Set(Timeout(30)=response)
exten => 501,n,Background(velkommen_abacus&tast123vent_)


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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Re: [asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun

Thank You Tarek!

That was the case, and i saw now i had a typo in the extension further down, 
but, you solved it.
Now I faced a couple of other problems, alle the announcements and MOH didn’t 
play, the settings are default.
Maybe i'll figure it out.

Thank you


Regards 

Aksel


Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Tarek Sawah
Sendt: 14. juni 2010 15:00
Til: Asterisk Users
Emne: Re: [asterisk-users] Call queues - issues, can't make it work.

when you add an agent to a queue the agent should log in
try adding
member=SIP/301
member=SIP/302
instead of agent directives.
this will ring both phones.. from your output it doesn't seem to be ringing the 
agents at all.

-- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 
562 2308



From: ak...@abacus-it.no
To: asterisk-users@lists.digium.com
Date: Mon, 14 Jun 2010 13:41:20 +0200
Subject: [asterisk-users] Call queues - issues, can't make it work.
Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue “teknisk”
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings…and rings…until it hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

[asterisk-users] Call queues - issues, can't make it work.

2010-06-14 Thread Aksel Celasun

Hello there


I have been struggling with queues, because i think this is the right module 
for our business.
My main goal, is when we receive external calls, the receptionist should be 
able to transfer the call to us
Technicians, and I am trying to add 2 extensions to a queue name [teknisk]
Extension 301 and 302.

I have a test setup now which I thought should look like this:
When a external call come to my external number (67209611) this will ring for 5 
seconds, and then transferred to queue "teknisk"
And I thought that internal phonex/extensions 301 and 302 would ring.

But, when I ring the external number, it just rings...and rings...until it 
hang-ups.

CLI output shows that the commands are running, but maybe the wrong way, are 
the queue command routed to my sip provider?

Info: 67209611 is my public phone number.
90015103 is my cell phone number
301 and 302 are internal extensions in technician department, which I am trying 
to route the queue to with the ringall argument.
This happens:
Reloading MGCP
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [4767209...@internal:1] 
NoOp("SIP/odin.service.ipallover.net-00d1", "") in new stack
-- Executing [4767209...@internal:2] 
Verbose("SIP/odin.service.ipallover.net-00d1", "Callerid num 90015103") in 
new stack
Callerid num 90015103
-- Executing [4767209...@internal:3] 
Dial("SIP/odin.service.ipallover.net-00d1", "SIP/301,5") in new stack
  == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
-- Called 301
-- SIP/301-00d2 is ringing
-- Nobody picked up in 5000 ms
-- Executing [4767209...@internal:4] 
Queue("SIP/odin.service.ipallover.net-00d1", "teknisk") in new stack
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
--  Playing 'queue-youarenext.gsm' 
(language 'en')
-- Told SIP/odin.service.ipallover.net-00d1 in teknisk their queue 
position (which was 1)
--  Playing 'queue-thankyou.gsm' 
(language 'en')
-- Started music on hold, class 'default', on channel 
'SIP/odin.service.ipallover.net-00d1'
-- Stopped music on hold on SIP/odin.service.ipallover.net-00d1
  == Spawn extension (internal, 4767209611, 4) exited non-zero on 
'SIP/odin.service.ipallover.net-00d1'

asterisk*CLI>

---
Agents.conf is default and  i have two extensions/agents
agent => 301,301
agent => 302,302


--
[r...@asterisk asterisk]# more queues.conf

[teknisk]
music = default
announce = queue-callswaiting.gsm
strategy = ringall
timeout = 15
retry = 0
maxlen = 0
announce-frequency = 120
announce-holdtime = yes

member => Agent/301
member => Agent/302

-
Sip.conf
[301]
type=friend
secret=xx
host=dynamic
context=phones
mailbox=...@default
qualify=yes
callgroup=teknisk
-
extensions.conf snipped

;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301);   ;Added 06.Mai.10-Aksel




Could someone please help me in the right direction here?


Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner

Tor Aksel Celasun
Mobilnummer 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>

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Re: [asterisk-users] Reloading queue members (realtime DB)

2010-06-01 Thread Aksel Celasun

Try "queue show"
And "queue show rules" and mail the output back.


Regards

Aksel

Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 31. mai 2010 21:35
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Re: [asterisk-users] Reloading queue members (realtime DB)

I did, "show queues" doesn't show the membernames, but the interface (which is 
normal if the membername is NULL in the table, but it isn't).

Mike

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun
Sent: Monday, May 31, 2010 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Reloading queue members (realtime DB)

Hello there.


Have you tried "reload" in CLI?


Greeting
Aksel

Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 31. mai 2010 21:00
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Reloading queue members (realtime DB)

Hi,

Asterisk 1.4.31 here.  I have a queue with members defined, and those member 
have member names "member 1", "member 2", etc.  They are in a realtime DB.

When I modify those member names (column membername) the changes aren't 
reflected in the queue status ("show queues" from cli. They aren't reflected 
when a new call comes in, or when I reload the dialplan.

What do I need to do for the changes to be shown in the CLI, short of 
restarting Asterisk?

Regards,

Mike
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Re: [asterisk-users] Reloading queue members (realtime DB)

2010-05-31 Thread Aksel Celasun

Hello there.


Have you tried "reload" in CLI?


Greeting
Aksel

Fra: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 31. mai 2010 21:00
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Reloading queue members (realtime DB)

Hi,

Asterisk 1.4.31 here.  I have a queue with members defined, and those member 
have member names "member 1", "member 2", etc.  They are in a realtime DB.

When I modify those member names (column membername) the changes aren't 
reflected in the queue status ("show queues" from cli. They aren't reflected 
when a new call comes in, or when I reload the dialplan.

What do I need to do for the changes to be shown in the CLI, short of 
restarting Asterisk?

Regards,

Mike
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Reloading queue members (realtime DB)

Re: [asterisk-users] Reloading queue members (realtime DB)

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