[asterisk-users] safe_asterisk ?

[virendra bhati]

Hi List,

What do it mean safe_asterisk ? I read too much about it but how it's works
as Daemon process?
When We install asterisk with the help of .tar file then safe_asterisk is
install or not ? If yes then how can we work with that ?
I am too much confusing..



-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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[asterisk-users] how to call a ring group via the dial plan language in asterisk?

[Edward de Jong]

When you are dialing a regular extension you might do something like this:

exten => Dial(123)

that would presumably dial extension 123.

but when one is using freepbx to admin the asterisk, and building a custom 
piece of dial plan code, how do you access a ring group?

exten => Dial (Local/601@)

how does one know what context to specify? and is the local keyboard needed? is 
Local case sensitive?


help is appreciated.



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Re: [asterisk-users] How to use password file with AuthenticateApplication

[Warren Selby]

On Tue, Nov 22, 2011 at 11:02 PM, virendra bhati  wrote:

> Hi Warren,
>
> As per your suggestion I revert back the things. In such case nothing is
> working. So it's completely wrong case.
>
> Can someone tell me how Authenticate check password from plan text file ?
> If we know who it's work then we can implements the logic on it.
>

I'm not sure, one thought would be to try without the "a" option in the
Read()?  Other than that, I'd suggest maybe opening a ticket on the issue
tracker.

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--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] Sending more than one call the agent while he is already in a call !! ringinuse=no/yes

[Raj Mathur (राज माथुर)]

On Wednesday 23 Nov 2011, bilal ghayyad wrote:
> Asterisk version is 1.8.4.2
> 
> Just I need to know if the below is a  normal behaviour of asterisk
> or I have something wrong in the settings:
> 
> I am surprised how the queue is sending calls (and not only one call,
> but a lot of calls) to the agent and the agent already has a call?!!
> 
> I tried ringinuse=no but same thing.
> 
> Does this happen because when I login to the queue then I use the
> Phone Interface (for example, SIP/username) and if I used the login
> to be as agent, then I will not see this problem? Really I spent a
> lot of time trying to resolve it without any success.

You need to set call-limit and busylevel in the peer's SIP 
configuration.  Works for our agents with call-limit = 50 (some 
arbitrary figure) and busylevel = 1.

ringinuse = no is required in any case.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Sending more than one call the agent while he is already in a call !! ringinuse=no/yes

[Sammy Govind]

Hello,

You need to limit the calls on that agent's sip declaration to 1. I guess
the parameter is call-limit=1

Regards,
Sammy

On Wed, Nov 23, 2011 at 7:46 AM, bilal ghayyad  wrote:

> Hi All;
>
> Asterisk version is 1.8.4.2
>
> Just I need to know if the below is a  normal behaviour of asterisk or I
> have something wrong in the settings:
>
> I am surprised how the queue is sending calls (and not only one call, but
> a lot of calls) to the agent and the agent already has a call?!!
>
> I tried ringinuse=no but same thing.
>
> Does this happen because when I login to the queue then I use the Phone
> Interface (for example, SIP/username) and if I used the login to be as
> agent, then I will not see this problem? Really I spent a lot of time
> trying to resolve it without any success.
>
> The phones I am using is Cisco with SIP image (also I tried Polycom and
> the problem is the same), but just to give all the information, I need to
> say that the contains the following (I am mentioning this to know if it is
> effecting and causing this problem):
>
> 
>   2
>
>3
>
> Please, I need a help.
>
> Regards
> Bilal
>
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Re: [asterisk-users] How to use password file with AuthenticateApplication

[virendra bhati]

Hi Warren,

As per your suggestion I revert back the things. In such case nothing is
working. So it's completely wrong case.

Can someone tell me how Authenticate check password from plan text file ?
If we know who it's work then we can implements the logic on it.


On Tue, Nov 22, 2011 at 9:47 PM, Warren Selby  wrote:

> On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati wrote:
>
>> Hi,
>>
>> After deleting all space no improvements.
>>
>
> Try reversing the account code and password hash, like this:
>
> 81dc9bdb52d04dc20036dbd8313ed055:Virendra
> 9996535e07258a7bbfd8b132435c5962:Vijay
> 7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
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>



-- 



-
Thanks and regards

 Virendra Bhati
+91-9172341457
Software Engineer
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[asterisk-users] Sending more than one call the agent while he is already in a call !! ringinuse=no/yes

[bilal ghayyad]

Hi All;

Asterisk version is 1.8.4.2

Just I need to know if the below is a  normal behaviour of asterisk or I have 
something wrong in the settings:

I am surprised how the queue is sending calls (and not only one call, but a lot 
of calls) to the agent and the agent already has a call?!!

I tried ringinuse=no but same thing.

Does this happen because when I login to the queue then I use the Phone 
Interface (for example, SIP/username) and if I used the login to be as agent, 
then I will not see this problem? Really I spent a lot of time trying to 
resolve it without any success.

The phones I am using is Cisco with SIP image (also I tried Polycom and the 
problem is the same), but just to give all the information, I need to say that 
the contains the following (I am mentioning this to know if it is effecting and 
causing this problem):


   2

3

Please, I need a help.

Regards
Bilal

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[asterisk-users] Receive sms via bluetooth with chan_mobile and an Android phone?

[Sebastian Arcus]

  
  
Hi all,
  
  I've been using a Nokia series 40 phone to receive GSM incoming
  calls OK into Asterisk for a few years now. According to the
  documentation of chan_mobile, it seems that (at least some) Nokia
  mobiles with S60 operating systems should be able to also receive
  (and even send) sms via bluetooth using chan_mobile.
  
  Is anybody here familiar with the implementations of bluetooth
  stacks and what is required to receive and send SMS via bluetooth?
  More precisely, the question is - do any of the Android devices
  (or versions of Android) have the required bluetooth stack to
  perform sms over bluetooth (at least in theory)? I already have an
  Android phone - and would like to put it to good use for incoming
  sms - if it has what it takes.
  
  Looking at the Wikipedia page for Bluetooth profiles
  (http://en.wikipedia.org/wiki/Bluetooth_profile), I'm trying to
  figure out which profile is used by chan_mobile to retrieve and
  send sms (and implicitly, which profile would need to be supported
  by the phone in order to enable this). Would it be the SAP/SIM
  profile, or MAP profile - or another one?
  
  Then I could use this information to work out if any Android
  version implements the required profile.
  
  Many thanks in advance,
  
  Sebastian

  


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[asterisk-users] Follow me unreachable message default

[Todd Routhier]

Hey folks, been out of the loop for a while and need to make a few changes
to my Ast box.

I have been digging around trying to find an answer on the list archives,
the wiki, google etc. but not joy.

I have using Asterisk 1.4.8 and have recently added the FollowMe feature.

Basically, I have what I need working but when nobody answers on any of the
follow me numbers the caller hears the "The party you are calling is
unreachable" message, then goes to my vmail.

This is OK but I don't like the unreachable message and I want that
skipped/removed from my config.

I read in the docs that you are suppose to use the "n" option to enable
this which I have NOT done anywhere in followme.conf or when I call the
followme app from my dial plan. For some reason it's defaulting to this
option. Is there a way to turn the playing of the unreachable message off,
short of changing the message it plays to something more desirable?

n - Playback the unreachable status message if we've run out of steps to
reach the

Thanks,
 Todd
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Re: [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall

[Danny Nicholas]

If you are using 1 Queue command, you're probably out of luck.  If you are
using 2 Queue commands, you can put a Wait(.1) between them.  Can you post
the dialplan snippet?



 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas
Mortensen
Sent: Tuesday, November 22, 2011 4:34 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] How to program a 100ms delay between the ringing
of queued calls w/ ringall

 

Hello,

 

Does anyone have any idea of how I can program a 100ms delay in between the
ringing of 2 subsequent calls in a queue configured with a ringall strategy?
In other words, our queue ringing strategy rings all queue agents with the
first caller in line in the queue. We only permit 1 ringing call at a time,
but it does ring to all phones. After an agent answers it, asterisk then
rings through the 2nd caller in the queue to all phones/agents again (other
than the "busy" agent who is on the phone with the 1st caller).

 

I want to force a 100ms delay in between the time that the 1st call is
answered, and the 2nd call starts ringing.

 

Any suggestions of how I can do this? I assume I'll need to modify the
asterisk dial plan, as I didn't see any queue parameters that would provide
this feature for me.

 

I do appreciate your help!

 

Sincerely,

-

Doug Mortensen

Network Consultant

Impala Networks Inc

CCNA, MCSA, Security+, A+

Linux+, Network+, Server+

.

www.impalanetworks.com

P: (505) 327-7300

F: (505) 327-7545

.

 

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[asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall

[Douglas Mortensen]

Hello,

Does anyone have any idea of how I can program a 100ms delay in between the 
ringing of 2 subsequent calls in a queue configured with a ringall strategy? In 
other words, our queue ringing strategy rings all queue agents with the first 
caller in line in the queue. We only permit 1 ringing call at a time, but it 
does ring to all phones. After an agent answers it, asterisk then rings through 
the 2nd caller in the queue to all phones/agents again (other than the "busy" 
agent who is on the phone with the 1st caller).

I want to force a 100ms delay in between the time that the 1st call is 
answered, and the 2nd call starts ringing.

Any suggestions of how I can do this? I assume I'll need to modify the asterisk 
dial plan, as I didn't see any queue parameters that would provide this feature 
for me.

I do appreciate your help!

Sincerely,
-
Doug Mortensen
Network Consultant
Impala Networks Inc
CCNA, MCSA, Security+, A+
Linux+, Network+, Server+
.
www.impalanetworks.com
P: (505) 327-7300
F: (505) 327-7545
.

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Re: [asterisk-users] AEX410P drops DTMF digits

[Danny Nicholas]

Just to update you, I got Asterisk 10 up and running by modifying main/db.c to 
open astdb2 instead of astdb and modified safe_asterisk to drop the astdb2 
table on each restart.  All of my testing on 10.0 has not repeated the dropped 
digit or app_read failing problems.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, November 22, 2011 10:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AEX410P drops DTMF digits

On Tue, Nov 22, 2011 at 09:25:45AM -0600, Danny Nicholas wrote:
> Server is  SUSE Linux Enterprise Server 10 for AMD64(TM) & Intel(R) 
> EM64T

>  │  i  │sqlite  │3.2.8   │3.2.8  │Embeddable SQL Database

That's sqlite3, indeed.

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

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Re: [asterisk-users] AEX410P drops DTMF digits

[Danny Nicholas]

What would cause this?
[Nov 22 11:34:08] WARNING[12709]: db.c:131 init_stmt: Couldn't prepare 
statement 'CREATE TABLE IF NOT EXISTS astdb(key VARCHAR(256), value 
VARCHAR(256), PRIMARY KEY(key))': near "NOT": syntax error
[Nov 22 11:34:08] WARNING[12709]: db.c:176 db_create_astdb: Couldn't create 
astdb table: near "NOT": syntax error


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, November 22, 2011 10:25 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AEX410P drops DTMF digits

On Tue, Nov 22, 2011 at 09:25:45AM -0600, Danny Nicholas wrote:
> Server is  SUSE Linux Enterprise Server 10 for AMD64(TM) & Intel(R) 
> EM64T

>  │  i  │sqlite  │3.2.8   │3.2.8  │Embeddable SQL Database

That's sqlite3, indeed.

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

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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

Jonas,

I did see the traces and I agree they accurately show what you described.  What 
traces never do is say WHY it's happening.  What I suggest is that you go into 
the cli as root with:

asterisk -r

issue the following commands:

 core set debug 9
 core set verbose 5
 quit

and then monitor the logs found in /var/log/asterisk (that's where they are 
usually anyway)

you will most likely find the cause in either the debug, full or messages file, 
again assuming you have those files enabled.  If you don't have them enabled, 
please contact me,
either directly or via the list and I'll help you get them enabled and assist 
you in interpreting them.

I suggest that you NOT post them to the list as they could be quite large and a 
risk of accidentally revealing something you might not want to reveal.


On 11/22/2011 08:51 AM, Jonas Kellens wrote:
> On 11/22/2011 05:42 PM, Alex Vishnev wrote:
>> I doubt it. Unknown headers should be ignored by UAS. is it possible to post 
>> the trace?
>> On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:
>
> What trace do you need ? Have you read my original post ? Asterisk SIP debug 
> trace is posted in my original post.
>
>
> Kind regards,
> Jonas.
>
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Alex Vishnev]

it is strange that Aastra acks 401, sends another invite but does not increase 
CSeq. Is that the same behavior with others?
On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote:

> On 11/22/2011 05:42 PM, Alex Vishnev wrote:
>> I doubt it. Unknown headers should be ignored by UAS. is it possible to post 
>> the trace?
>> On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:
> 
> What trace do you need ? Have you read my original post ? Asterisk SIP debug 
> trace is posted in my original post.
> 
> 
> Kind regards,
> Jonas.
> 
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Jonas Kellens]

On 11/22/2011 05:42 PM, Alex Vishnev wrote:
I doubt it. Unknown headers should be ignored by UAS. is it possible 
to post the trace?

On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:


What trace do you need ? Have you read my original post ? Asterisk SIP 
debug trace is posted in my original post.



Kind regards,
Jonas.

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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Alex Vishnev]

I doubt it. Unknown headers should be ignored by UAS. is it possible to post 
the trace?
On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:

> On 11/22/2011 05:31 PM, Alex Vishnev wrote:
>> 
>> Your registration should have also have the flow
>> 
>> PEER ASTERISK
>> REGISTER--->
>> <--401
>> REGISTER(nonce) ->
>> <200OK
>> 
>> Then the server controls the life of the registration and 200 Expires Header 
>> gives you this timeout. If the invite is sent within that window, then 
>> Asterisk should not challenge anymore. If Invite is challenged and the peer 
>> responds with the correctly calculated NONCE, domain and other Auth params, 
>> then something is wrong with your Authentication. I suggest trapping the 
>> traffic with Ethereal or any other packet capture programs and examining 
>> that carefully from the start of the session (i.e. register) to the invite. 
>> I would also check where the 401 is coming from (i.e. IP address).
>> 
>> Hope that helps
>> 
>> Alex
> 
> 
> I've already captured with Wireshark, but what to do with it if I don't know 
> what I'm looking for ??
> 
> Registration goes without problem, but every INVITE is answered with a 
> 401-Unauthorized.
> 
> Like I already said : there is no problem with Avaya, Panasonic and 
> Alcatel-Lucent.
> The only difference I see between an INVITE from Avaya and the INVITE from 
> Aastra PBX is the presence of the SIP-header : "P-Behind-Gsi: 192.168.6.1".
> 
> Could this header mess up Asterisk ?
> 
> Jonas.
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Jonas Kellens]

On 11/22/2011 05:31 PM, Alex Vishnev wrote:

Your registration should have also have the flow

PEER ASTERISK
REGISTER--->
<--401
REGISTER(nonce) ->
<200OK

Then the server controls the life of the registration and 200 Expires 
Header gives you this timeout. If the invite is sent within that 
window, then Asterisk should not challenge anymore. If Invite is 
challenged and the peer responds with the correctly calculated NONCE, 
domain and other Auth params, then something is wrong with your 
Authentication. I suggest trapping the traffic with Ethereal or any 
other packet capture programs and examining that carefully from the 
start of the session (i.e. register) to the invite. I would also check 
where the 401 is coming from (i.e. IP address).


Hope that helps

Alex



I've already captured with Wireshark, but what to do with it if I don't 
know what I'm looking for ??


Registration goes without problem, but every INVITE is answered with a 
401-Unauthorized.


Like I already said : there is no problem with Avaya, Panasonic and 
Alcatel-Lucent.
The only difference I see between an INVITE from Avaya and the INVITE 
from Aastra PBX is the presence of the SIP-header : "P-Behind-Gsi: 
192.168.6.1".


Could this header mess up Asterisk ?

Jonas.
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Alex Vishnev]

Your registration should have also have the flow

PEER ASTERISK
REGISTER--->
<--401
REGISTER(nonce) ->
<200OK

Then the server controls the life of the registration and 200 Expires Header 
gives you this timeout. If the invite is sent within that window, then Asterisk 
should not challenge anymore. If Invite is challenged and the peer responds 
with the correctly calculated NONCE, domain and other Auth params, then 
something is wrong with your Authentication. I suggest trapping the traffic 
with Ethereal or any other packet capture programs and examining that carefully 
from the start of the session (i.e. register) to the invite. I would also check 
where the 401 is coming from (i.e. IP address).

Hope that helps

Alex
On Nov 22, 2011, at 11:23 AM, Jonas Kellens wrote:

> On 11/22/2011 04:37 PM, Bruce Ferrell wrote:
>> 
>> 
>> 
>> On 11/22/2011 07:29 AM, Jonas Kellens wrote:
>>> 
>>> On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
 
 
 Jonas,
 
 May I suggest that you present us your sip.conf entry for this peer, 
 properly redacted, of course.  That might help more.  What I do for 
 "gateways" at known addresses is to put an entry like this into the 
 sip.conf entry:
 
 
 [peer]
 type=peer
 defaultip=192.168.40.123
 insecure=invite,port
 context=some_context
 
>>> 
>>> 
>>> This is the peer definition in sip.conf :
>>> 
>>> [SIPPEERusername]
>>> type=friend
>>> host=dynamic
>>> defaultuser=SIPPEERusername
>>> secret=guessthis
>>> context=from-PEERTRUNK
>>> nat=yes
>>> dtmfmode=rfc2833
>>> canreinvite=no
>>> disallow=all
>>> allow=alaw
>>> allow=gsm
>>> 
>>> 
>>> Hope you can help me out with this extra information.
>>> 
>>> 
>>> Kind regards,
>>> 
>>> Jonas.
>> From what I see in your entry, you are requiring registration from the peer. 
>>  The next thing i would check is to see if the registration has succeeded.  
>> If it doesn't succeed, you will see the results you presented.  I see you 
>> have the peer set as a dynamic host, and if the IP address of the device 
>> does in fact change then registration is appropriate.
> 
> Registration of the SIP PEER is no problem. The PEER registers with a correct 
> REGISTER statement and Asterisk sends a 200 OK.
> 
> So the PEER is registered and then wants to make a call (INVITE) but for some 
> reason this INVITE is being refused with 401-Unauthorized.
> 
> The first 401-Unauthorized is normal, because the SIP PEER needs to send a 
> second INVITE with a challenge (nonce). But after this INVITE with challenge, 
> Asterisk still sends a 401 and that's strange !!
> 
> Jonas.
> 
> 
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Re: [asterisk-users] AEX410P drops DTMF digits

[Tzafrir Cohen]

On Tue, Nov 22, 2011 at 09:25:45AM -0600, Danny Nicholas wrote:
> Server is  SUSE Linux Enterprise Server 10 for AMD64(TM) & Intel(R) EM64T

>  │  i  │sqlite  │3.2.8   │3.2.8  │Embeddable SQL Database

That's sqlite3, indeed.

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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Jonas Kellens]

On 11/22/2011 04:37 PM, Bruce Ferrell wrote:



On 11/22/2011 07:29 AM, Jonas Kellens wrote:

On 11/22/2011 04:25 PM, Bruce Ferrell wrote:


Jonas,

May I suggest that you present us your sip.conf entry for this peer, 
properly redacted, of course.  That might help more.  What I do for 
"gateways" at known addresses is to put an entry like this into the 
sip.conf entry:



[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context




This is the peer definition in sip.conf :

[SIPPEERusername]
type=friend
host=dynamic
defaultuser=SIPPEERusername
secret=guessthis
context=from-PEERTRUNK
nat=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=alaw
allow=gsm


Hope you can help me out with this extra information.


Kind regards,

Jonas.
From what I see in your entry, you are requiring registration from the 
peer.  The next thing i would check is to see if the registration has 
succeeded.  If it doesn't succeed, you will see the results you 
presented.  I see you have the peer set as a dynamic host, and if the 
IP address of the device does in fact change then registration is 
appropriate.


Registration of the SIP PEER is no problem. The PEER registers with a 
correct REGISTER statement and Asterisk sends a 200 OK.


So the PEER is registered and then wants to make a call (INVITE) but for 
some reason this INVITE is being refused with 401-Unauthorized.


The first 401-Unauthorized is normal, because the SIP PEER needs to send 
a second INVITE with a challenge (nonce). But after this INVITE with 
challenge, Asterisk still sends a 401 and that's strange !!


Jonas.


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Re: [asterisk-users] How to use password file with AuthenticateApplication

[Warren Selby]

On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati  wrote:

> Hi,
>
> After deleting all space no improvements.
>

Try reversing the account code and password hash, like this:

81dc9bdb52d04dc20036dbd8313ed055:Virendra
9996535e07258a7bbfd8b132435c5962:Vijay
7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati

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http://www.SelbyTech.com 
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Re: [asterisk-users] check if devices reachable in queue

[Dale Noll]

On 11/22/2011 08:59 AM, Matt Hamilton wrote:

Thanks Dale, you pointed me in the right direction.


Happy to be of assistance.
I hope your project goes well.

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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

On 11/22/2011 07:29 AM, Jonas Kellens wrote:
> On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
>>
>> Jonas,
>>
>> May I suggest that you present us your sip.conf entry for this peer, 
>> properly redacted, of course.  That might help more.  What I do for 
>> "gateways" at known addresses is to put
>> an entry like this into the sip.conf entry:
>>
>>
>> [peer]
>> type=peer
>> defaultip=192.168.40.123
>> insecure=invite,port
>> context=some_context
>>
>
>
> This is the peer definition in sip.conf :
>
> [SIPPEERusername]
> type=friend
> host=dynamic
> defaultuser=SIPPEERusername
> secret=guessthis
> context=from-PEERTRUNK
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> disallow=all
> allow=alaw
> allow=gsm
>
>
> Hope you can help me out with this extra information.
>
>
> Kind regards,
>
> Jonas.
>From what I see in your entry, you are requiring registration from the peer.  
>The next thing i would check is to see if the registration has succeeded.  If 
>it doesn't succeed, you
will see the results you presented.  I see you have the peer set as a dynamic 
host, and if the IP address of the device does in fact change then registration 
is appropriate.
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Alex Vishnev]

do you see the register messages? if your device is not registered, INVITE 
would be challenged. You should check to see if register messages are being 
properly acknowledge with 200OK. 
On Nov 22, 2011, at 10:29 AM, Jonas Kellens wrote:

> On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
>> 
>> 
>> Jonas,
>> 
>> May I suggest that you present us your sip.conf entry for this peer, 
>> properly redacted, of course.  That might help more.  What I do for 
>> "gateways" at known addresses is to put an entry like this into the sip.conf 
>> entry:
>> 
>> 
>> [peer]
>> type=peer
>> defaultip=192.168.40.123
>> insecure=invite,port
>> context=some_context
>> 
> 
> 
> This is the peer definition in sip.conf :
> 
> [SIPPEERusername]
> type=friend
> host=dynamic
> defaultuser=SIPPEERusername
> secret=guessthis
> context=from-PEERTRUNK
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> disallow=all
> allow=alaw
> allow=gsm
> 
> 
> Hope you can help me out with this extra information.
> 
> 
> Kind regards,
> 
> Jonas.
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> _
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Jonas Kellens]

On 11/22/2011 04:25 PM, Bruce Ferrell wrote:


Jonas,

May I suggest that you present us your sip.conf entry for this peer, 
properly redacted, of course.  That might help more.  What I do for 
"gateways" at known addresses is to put an entry like this into the 
sip.conf entry:



[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context




This is the peer definition in sip.conf :

[SIPPEERusername]
type=friend
host=dynamic
defaultuser=SIPPEERusername
secret=guessthis
context=from-PEERTRUNK
nat=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=alaw
allow=gsm


Hope you can help me out with this extra information.


Kind regards,

Jonas.
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Re: [asterisk-users] AEX410P drops DTMF digits

[Danny Nicholas]

Server is  SUSE Linux Enterprise Server 10 for AMD64(TM) & Intel(R) EM64T

Here is the SQLITE info
│ │Name│Avail. Vers.│Inst. Vers.│Summary
│Size│
 │ │mono-data-sqlite│1.1.13.8│   │Database connectivity for
Mono   │73.0 K│ │
 │  i  │perl-DBD-SQLite │1.11│1.11   │The DBD::SQLite is a self
contained RDBMS in a DBI driver│92.5 K│ │
 │ │php5-sqlite │5.1.2   │   │PHP5 Extension Module
│   354.6 K│ │
 │ │qt-sql-sqlite   │4.1.0   │   │Qt 4 sqlite plugin
│41.0 K│ │
 │  i  │sqlite  │3.2.8   │3.2.8  │Embeddable SQL Database
Engine   │   381.7 K│ │
 │  i  │sqlite-32bit│3.2.8   │3.2.8  │Embeddable SQL Database
Engine   │   327.0 K│ │
 │ │sqlite-devel│3.2.8   │   │Embeddable SQL Database
Engine   │ 1.5 M│ │
 │  i  │sqlite2 │2.8.17  │2.8.17 │Embeddable SQL Database
Engine   │   322.5 K│ │
 │ │sqlite2-32bit   │2.8.17  │   │Embeddable SQL Database
Engine   │   280.2 K│ │
 │  i  │sqlite2-devel   │2.8.17  │2.8.17 │Embeddable SQL Database
Engine   │ 1.1 M│ │
L---

 Filter: Search Res
Required Disk Space: 0 B
 

---┐
 │sqlite - Embeddable SQL Database Engine
│
 │Version: 3.2.8-15.2 Installed: 3.2.8-15.2 Size: 381.7 K Media No.: 1
│
 │License: distributable, Other License(s), see package
│
 │Package Group: Productivity/Databases/Servers
│
 │Provides: libsqlite3.so.0()(64bit), sqlite == 3.2.8-15.2
│
 │Authors:


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, November 22, 2011 9:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AEX410P drops DTMF digits

On Tue, Nov 22, 2011 at 08:14:14AM -0600, Danny Nicholas wrote:
> Suse info -
> Linux x 2.6.16.27-0.9-smp #1 SMP Tue Feb 13 09:35:18 UTC 2007 
> x86_64
> x86_64 x86_64 GNU/Linux

What is the version of the SUSE distribution?

> 
> This version of SUSE does not have SQLite3 in it's repository, only 
> sqlite and sqlite2

Any chance sqlite is sqlite3? What is the version number of that package?

-- 
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

Jonas,

May I suggest that you present us your sip.conf entry for this peer, properly 
redacted, of course.  That might help more.  What I do for "gateways" at known 
addresses is to put an
entry like this into the sip.conf entry:


[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context





On 11/22/2011 06:40 AM, Jonas Kellens wrote:
> Hello list,
>
> this is the communication between an Aastra 5000 PBX and Asterisk, where the 
> Aastra makes a call to Asterisk. For some reason, Asterisk responds with 
> 401-Unauthorized and I don't
> know why.
>
> Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with 
> this Aastra.
>
>
> A1.A1.A1.A1 = IP-address Asterisk PBX
> AS.AS.AS.AS = IP-address Aastra PBX
>
> Aastra PBX makes a call to the number 3221112233...
>
> This is all the sip debug trace gathered with asterisk :
>
>
> <--- SIP read from UDP:AS.AS.AS.AS:61490 --->
> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
> From: ;tag=310158BD
> To: 
> Call-ID: 0201CEFEA742
> CSeq: 1 INVITE
> Contact: 
> Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", 
> nonce="67105ac4", uri="sip:3221112233@A1.A1.A1.A1:5060", response="60be856773
> f86450fc9ddbaf7a568505", algorithm=MD5
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
> Max-Forwards: 70
> Privacy: none
> P-Asserted-Identity: 
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Type: application/sdp
> Content-Length:195
>
> v=0
> o=- 0 0 IN IP4 sip.domain.tld
> s=-
> i=(o=IN IP4 10.1.2.35)
> c=IN IP4 AS.AS.AS.AS
> t=0 0
> m=audio 62654 RTP/AVP 8 0
> a=rtcp:65115
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
> <->
>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 
> 11 lines) ---
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using SIP 
> RTP TOS bits 184
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using SIP 
> RTP CoS mark 5
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to 
> AS.AS.AS.AS : 61490 (no NAT)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE 
> request as basis request - 0201CEFEA742
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 
> 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> <--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 
> AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490
> From: ;tag=310158BD
> To: ;tag=as68f71fe5
> Call-ID: 0201CEFEA742
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9"
> Content-Length: 0
>
> <>
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling 
> destruction of SIP dialog '0201CEFEA742' in 32000 ms (Method: INVITE)
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> <--- SIP read from UDP:AS.AS.AS.AS:61490 --->
> ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
> From: ;tag=310158BD
> To: ;tag=as68f71fe5
> Call-ID: 0201CEFEA742
> CSeq: 1 ACK
> Contact: 
> Max-Forwards: 70
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Length: 0
>
>
> <->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 
> 0 lines) ---
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
> <--- SIP read from UDP:AS.AS.AS.AS:61490 --->
> INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
> Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport
> From: ;tag=33015DBD
> To: 
> Call-ID: 0201CCFEA242
> CSeq: 1 INVITE
> Contact: 
> Proxy-Authorization: Digest username="SIPPEERusername", realm="domain.tld", 
> nonce="46ef24d9", uri="sip:3221112233@A1.A1.A1.A1:5060", 
> response="14ecbfc7df24b49926151284c123ea11",
> algorithm=MD5
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
> Max-Forwards: 70
> Privacy: none
> P-Asserted-Identity: 
> User-Agent: A5000 R52-H2C0205
> P-Behind-Gsi: 192.168.6.1
> Content-Type: application/sdp
> Content-Length:195
>
> v=0
> o=- 0 0 IN IP4 sip.domain.tld
> s=-
> i=(o=IN IP4 10.1.2.35)
> c=IN IP4 AS.AS.AS.AS
> t=0 0
> m=audio 62654 RTP/AVP 8 0
> a=rtcp:65115
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
>
> <->
> [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 
> 11 lines) ---
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using SIP 
> RTP TOS bits 184
> [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15

Re: [asterisk-users] sip show peers

[Danny Nicholas]

Re-compile channels/chan_sip.c because this is what is limiting you
/*! \brief  _sip_show_peers: Execute sip show peers command */
static int _sip_show_peers(int fd, int *total, struct mansession *s, const
struct message *m, int argc, const char *argv[])
{
regex_t regexbuf;
int havepattern = FALSE;

#define FORMAT2 "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n"
#define FORMAT  "%-25.25s  %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n"

char name[256];
int total_peers = 0;
int peers_mon_online = 0;
int peers_mon_offline = 0;
int peers_unmon_offline = 0;
int peers_unmon_online = 0;
const char *id;
char idtext[256] = "";
int realtimepeers;

realtimepeers = ast_check_realtime("sippeers");

if (s) {/* Manager - get ActionID */
id = astman_get_header(m,"ActionID");
if (!ast_strlen_zero(id))
snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n",
id);
}

switch (argc) {
case 5:
if (!strcasecmp(argv[3], "like")) {
if (regcomp(®exbuf, argv[4], REG_EXTENDED |
REG_NOSUB))
return RESULT_SHOWUSAGE;
havepattern = TRUE;
} else
return RESULT_SHOWUSAGE;
case 3:
break;
default:
return RESULT_SHOWUSAGE;
}

if (!s) /* Normal list */
ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Nat",
"ACL", "Port", "Status", (realtimepeers ? "Realtime" : ""));
the 25.25s definition of FORMAT and FORMAT2 means you get 25 characters to
display.   You should be able to change the 25.25 to something like 50.50 (I
tried 45.45 and it worked for me). 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, November 22, 2011 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers

  Is there a way with the command (1.4.42) for sip show peers to see the
FULL "Name/Username" field???

I have long names and mine are being truncated.

Thanks

Jerry

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Re: [asterisk-users] Resell VoIP Servcies

[Jai Rangi]

I am sorry. Meant to send to biz list. Thank you for correcting me.


On Tue, Nov 22, 2011 at 5:57 AM, Patrick Lists <
asterisk-l...@puzzled.xs4all.nl> wrote:

> On 11/22/2011 08:14 AM, Jai Rangi wrote:
> [removed commercial offer]
>
> You posted to the wrong list. The correct list for commercial & business
> related discussion is asterisk-biz. Please do not spam the asterisk-users
> list again with your commercial offers.
>
> Regards,
> Patrick
>
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[asterisk-users] (no subject)

[Charles Wang]

http://aiscjmi.com/modules/mod_wdbanners/time.php?html143

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Re: [asterisk-users] sip show peers

[eherr]

I believe it is set by a character length for formatting the output.

What are you trying to accomplish? Are you just viewing it in the CLI or are 
you writing monitoring scripts?

Can you switch names so that they are unique in the beginning?

--E

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, November 22, 2011 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers

  Is there a way with the command (1.4.42) for sip show peers to
see the FULL "Name/Username" field???

I have long names and mine are being truncated.

Thanks

Jerry

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Re: [asterisk-users] AEX410P drops DTMF digits

[Tzafrir Cohen]

On Tue, Nov 22, 2011 at 08:14:14AM -0600, Danny Nicholas wrote:
> Suse info -
> Linux x 2.6.16.27-0.9-smp #1 SMP Tue Feb 13 09:35:18 UTC 2007 x86_64
> x86_64 x86_64 GNU/Linux

What is the version of the SUSE distribution?

> 
> This version of SUSE does not have SQLite3 in it's repository, only sqlite
> and sqlite2

Any chance sqlite is sqlite3? What is the version number of that
package?

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[asterisk-users] sip show peers

[Jerry Geis]

 Is there a way with the command (1.4.42) for sip show peers to
see the FULL "Name/Username" field???

I have long names and mine are being truncated.

Thanks

Jerry

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Re: [asterisk-users] check if devices reachable in queue

[Matt Hamilton]

Thanks Dale, you pointed me in the right direction.





> You are calling the Dial() application here.  If you are using
queues, you should use the Queue() application.  




I'm using Local channels with the queue:



- queues.conf

[support]

member => Local/1001@handle-queue





---extensions.conf-


[incoming]

Queue(support)



[handle-queue]

; some preprocessing here

same => n,Dial()





I found out that since I'm using the Local channel as the queue member, 
the Queue() doesn't know the state the call is in. It monitors the state
 of the Local channel, and not the device. However, it seems to be 
possible to give the queue the actual device to monitor and associate 
that with the Local channel by modifiying the member in the queues.conf:





; queues.conf

[support]

member => Local/1001@[handle-queue],,,SIP/1001





I won't be able to test this until this evening, but it seems like it's going 
to work...


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[asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Jonas Kellens]

Hello list,

this is the communication between an Aastra 5000 PBX and Asterisk, where 
the Aastra makes a call to Asterisk. For some reason, Asterisk responds 
with 401-Unauthorized and I don't know why.


Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT 
with this Aastra.



A1.A1.A1.A1 = IP-address Asterisk PBX
AS.AS.AS.AS = IP-address Aastra PBX

Aastra PBX makes a call to the number 3221112233...

This is all the sip debug trace gathered with asterisk :


<--- SIP read from UDP:AS.AS.AS.AS:61490 --->
INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
From: ;tag=310158BD
To: 
Call-ID: 0201CEFEA742
CSeq: 1 INVITE
Contact: 
Proxy-Authorization: Digest username="SIPPEERusername", 
realm="domain.tld", nonce="67105ac4", 
uri="sip:3221112233@A1.A1.A1.A1:5060", response="60be856773

f86450fc9ddbaf7a568505", algorithm=MD5
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
Max-Forwards: 70
Privacy: none
P-Asserted-Identity: 
User-Agent: A5000 R52-H2C0205
P-Behind-Gsi: 192.168.6.1
Content-Type: application/sdp
Content-Length:195

v=0
o=- 0 0 IN IP4 sip.domain.tld
s=-
i=(o=IN IP4 10.1.2.35)
c=IN IP4 AS.AS.AS.AS
t=0 0
m=audio 62654 RTP/AVP 8 0
a=rtcp:65115
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

<->

[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 
headers 11 lines) ---
[Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using 
SIP RTP TOS bits 184
[Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using 
SIP RTP CoS mark 5
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to 
AS.AS.AS.AS : 61490 (no NAT)
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using 
INVITE request as basis request - 0201CEFEA742
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 
'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490

[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
<--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490

From: ;tag=310158BD
To: ;tag=as68f71fe5
Call-ID: 0201CEFEA742
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="domain.tld", nonce="46ef24d9"
Content-Length: 0

<>
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling 
destruction of SIP dialog '0201CEFEA742' in 32000 ms (Method: INVITE)

[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
<--- SIP read from UDP:AS.AS.AS.AS:61490 --->
ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
From: ;tag=310158BD
To: ;tag=as68f71fe5
Call-ID: 0201CEFEA742
CSeq: 1 ACK
Contact: 
Max-Forwards: 70
User-Agent: A5000 R52-H2C0205
P-Behind-Gsi: 192.168.6.1
Content-Length: 0


<->
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 
headers 0 lines) ---

[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
<--- SIP read from UDP:AS.AS.AS.AS:61490 --->
INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport
From: ;tag=33015DBD
To: 
Call-ID: 0201CCFEA242
CSeq: 1 INVITE
Contact: 
Proxy-Authorization: Digest username="SIPPEERusername", 
realm="domain.tld", nonce="46ef24d9", 
uri="sip:3221112233@A1.A1.A1.A1:5060", 
response="14ecbfc7df24b49926151284c123ea11", algorithm=MD5

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
Max-Forwards: 70
Privacy: none
P-Asserted-Identity: 
User-Agent: A5000 R52-H2C0205
P-Behind-Gsi: 192.168.6.1
Content-Type: application/sdp
Content-Length:195

v=0
o=- 0 0 IN IP4 sip.domain.tld
s=-
i=(o=IN IP4 10.1.2.35)
c=IN IP4 AS.AS.AS.AS
t=0 0
m=audio 62654 RTP/AVP 8 0
a=rtcp:65115
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20


<->
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 
headers 11 lines) ---
[Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using 
SIP RTP TOS bits 184
[Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using 
SIP RTP CoS mark 5
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to 
AS.AS.AS.AS : 61490 (no NAT)
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using 
INVITE request as basis request - 0201CCFEA242
[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 
'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490

[Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
<--- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;received

Re: [asterisk-users] difference between playback and background?

[Kingsley Tart]

Alternatively, if you don't have that extension defined anywhere,
Asterisk will jump to the i extension, where you can then read the
actual entered digits from the INVALID_EXTEN variable and jump back to
the main part of the dialplan.

Note that if they enter digits that *could* match a defined extension,
Asterisk won't necessarily jump to the i extension until more digits are
entered which may or may not cause no explicitly defined extensions to
be matched.

Cheers,
Kingsley.

On Mon, 2011-11-21 at 12:32 -0800, Steve Edwards wrote:
> On Mon, 21 Nov 2011, Danny Nicholas wrote:
> 
> > Option 2
> > Use WaitExten with Background
> > [getnum]
> > Exten => start,1,background(prompt)
> > Exten => start,n,waitexten(2)
> > Exten => ,1,noop(user pressed )
> > Exten => I,1,playback(invalid)
> >
> > For option 2 you have to define each valid 4 digit entry in the context.
> 
> Or, (since the OP seems a bit newbish), read up on extension pattern 
> matching.
> 
-- 
Cheers,
Kingsley.


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Re: [asterisk-users] AEX410P drops DTMF digits

[Danny Nicholas]

Suse info -
Linux x 2.6.16.27-0.9-smp #1 SMP Tue Feb 13 09:35:18 UTC 2007 x86_64
x86_64 x86_64 GNU/Linux

This version of SUSE does not have SQLite3 in it's repository, only sqlite
and sqlite2

The ports are all FXO
 
Going to try the dahdi_monitor this morning.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, November 22, 2011 3:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AEX410P drops DTMF digits

On Mon, Nov 21, 2011 at 04:17:59PM -0600, Danny Nicholas wrote:
> Hello again list,
> 
>   I'm running a 1.4.42 install on SUSE 
> with an AEX410P card.  The DAHDI release is 2.4.0 because the machine 
> won't properly install 2.5

Care to elaborate? Build issue? Install issue?

What version of SUSE?

> and also won't install Asterisk 10.0 because I can't get a good
> SQLite3 library to install.  

Package from your repository?

> Whenever I enter DTMF very quickly or very slowly, app_read des on me.  
> Has anyone experienced similar joy using DAHDI drivers?  I've piddled 
> with channel.c and app_read.c trying to tame this beast but it seems 
> to have the better of me.

What port is it? FXS or FXO?

If you want to record the audio at the DAHDI level:

  dahdi_monitor 1 -r dtmf.wav

(and press ctrl-c to stop recording)

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Re: [asterisk-users] Resell VoIP Servcies

[Patrick Lists]

On 11/22/2011 08:14 AM, Jai Rangi wrote:
[removed commercial offer]

You posted to the wrong list. The correct list for commercial & business 
related discussion is asterisk-biz. Please do not spam the 
asterisk-users list again with your commercial offers.


Regards,
Patrick

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[asterisk-users] Limit monthly calls by context

[Hans Goossen]

Many thanks to all for your suggestions. I followed adam's advise and created 
an AGI. It works brilliantly!

Hans Goossen
Investigación & Desarrollo
Planet S.A.
http://www.pla.net.py


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Re: [asterisk-users] check if devices reachable in queue

[Dale Noll]

On 11/21/2011 09:16 PM, Matt Hamilton wrote:




>Have you tried, instead of pre-processing the caller before calling 
Queue(), checking the ${QUEUESTATUS} variable.


Even when the phones are UNREACHABLE, QUEUE is still trying until it 
times out -  ${QUEUESTATUS} = TIMEOUT


I get the following for all the members of the queue, in a loop, until 
it times out.


Executing [1001@handle-queue:3] Dial("Local/1001@handle-queue-6d01;2", 
"SIP/1001") in new stack
You are calling the Dial() application here.  If you are using queues, 
you should use the Queue() application.
Dial() does not interact with the queue or the device state for the 
queue members, it just attempts to make the call.
If the queue members are not available, the Queue() app will immediately 
return to the next dialplan step.
If the members are available, the call will be placed into the queue.  
If then caller waits for TIMEOUT time, then dialplan will continue at 
the next step.
Either way, the ${QUEUESTATUS} variable will contain the reason it 
continued.


Dale


--
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 Lyta Alexander - Babylon 5

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Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

[David Cunningham]

Kingsley,

We have the same - the daemon forks child processes to handle individual
calls.

We need the fastAGI to continue so it can take some further action
recording details of the call. This could be done using the 'h' extension,
but it would be nice to avoid this method for simplicity sake. It does
appear that some people can continue after the Dial and we can't for some
reason.


On 22 November 2011 21:21, Kingsley Tart  wrote:

> When something makes a socket connection to your fastAGI daemon, does
> your daemon fork a child process to deal with that connection, or handle
> it in the main process?
>
> I've set ours up to fork a child process and detach itself from the
> parent socket. When it ends, the child exits (which is what we want) and
> the parent stays running (which is also what we want).
>
> Is there any particular reason you want your fastAGI instance to persist
> for the duration of the call?
>
> Cheers,
> Kingsley.
>
> On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote:
> > The strange thing is that we are using fast AGI, and for some reason
> > the AGI always exits when the caller hangs up - even when I set HUP to
> > IGNORE. If I set HUP to a subroutine that just logs a message, that
> > message is never logged.
> >
> > Thanks for all the help.
> >
> >
> > On 22 November 2011 05:23, Kingsley Tart 
> > wrote:
> > Yeah fastAGI is great, I've been using it for a while for
> > performance
> > reasons but yes I guess it would solve problems like this too.
> >
> > Cheers,
> > Kingsley.
> >
> > On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
> > > Just offhand, I think you should utilize the FastAGI
> > protocol, since it
> > > doesn't seem to live or die based on when the call hangs up.
> > Otherwise,
> > > the
> > >   $SIG{'HUP'} = 'IGNORE';
> > > Statement will "separate" the process so it doesn't die on a
> > hangup.
> > >
> > > -Original Message-
> > > From: asterisk-users-boun...@lists.digium.com
> > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
> > Of Kingsley Tart
> > > Sent: Monday, November 21, 2011 7:54 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] Continue AGI after Dial()
> > following caller
> > > hang up?
> > >
> > > Yeah I think I slightly misread your original question,
> > which I realised
> > > when I saw Thorsten's reply. I initially thought you just
> > wanted to avoid
> > > going into the h extension.
> > >
> > > I'm not doing any AGI stuff here that hangs around while the
> > call does stuff
> > > - the AGI process just runs quickly then quits, returning
> > control back to
> > > the dialplan. I had incorrectly assumed you were doing the
> > same.
> > >
> > > Cheers,
> > > Kingsley.
> > >
> > > On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
> > > > Kingsley,
> > > >
> > > > Thanks for the reply, but I am looking to continue within
> > the same AGI
> > > > process and I believe that method would require starting a
> > new AGI.
> > > >
> > > >
> > > > On 21 November 2011 22:22, Kingsley Tart
> > 
> > > > wrote:
> > > > We do that with the "F" option in Dial().
> > > >
> > > >
> > > > >From http://www.voip-info.org/wiki/view/Asterisk
> > +cmd+Dial :
> > > >
> > > > F(context^exten^pri): When the caller hangs up,
> > transfer the
> > > > called
> > > > party to the specified context and extension and
> > continue
> > > > execution.
> > > >
> > > >
> > > > Cheers,
> > > > Kingsley.
> > > >
> > > > On Mon, 2011-11-21 at 17:38 +1100, David
> > Cunningham wrote:
> > > > > Hello,
> > > > >
> > > > > We would like to continue a Perl AGI after a
> > Dial() it has
> > > > done
> > > > > completes following caller hangup. We would like
> > to do this
> > > > in the
> > > > > same AGI, and not using a new AGI from the 'h'
> > extension. It
> > > > works
> > > > > fine when the called party hangs up and the 'g'
> > option is
> > > > used, but
> > > > > not for caller hangup.
> > > > >
> > > > > Is this possible?
> > > > >
> > > > > If not a confirmation that this is the cas

Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

[Kingsley Tart]

When something makes a socket connection to your fastAGI daemon, does
your daemon fork a child process to deal with that connection, or handle
it in the main process?

I've set ours up to fork a child process and detach itself from the
parent socket. When it ends, the child exits (which is what we want) and
the parent stays running (which is also what we want).

Is there any particular reason you want your fastAGI instance to persist
for the duration of the call?

Cheers,
Kingsley.

On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote:
> The strange thing is that we are using fast AGI, and for some reason
> the AGI always exits when the caller hangs up - even when I set HUP to
> IGNORE. If I set HUP to a subroutine that just logs a message, that
> message is never logged.
> 
> Thanks for all the help.
> 
> 
> On 22 November 2011 05:23, Kingsley Tart 
> wrote:
> Yeah fastAGI is great, I've been using it for a while for
> performance
> reasons but yes I guess it would solve problems like this too.
> 
> Cheers,
> Kingsley.
> 
> On Mon, 2011-11-21 at 08:34 -0600, Danny Nicholas wrote:
> > Just offhand, I think you should utilize the FastAGI
> protocol, since it
> > doesn't seem to live or die based on when the call hangs up.
> Otherwise,
> > the
> >   $SIG{'HUP'} = 'IGNORE';
> > Statement will "separate" the process so it doesn't die on a
> hangup.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
> Of Kingsley Tart
> > Sent: Monday, November 21, 2011 7:54 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Continue AGI after Dial()
> following caller
> > hang up?
> >
> > Yeah I think I slightly misread your original question,
> which I realised
> > when I saw Thorsten's reply. I initially thought you just
> wanted to avoid
> > going into the h extension.
> >
> > I'm not doing any AGI stuff here that hangs around while the
> call does stuff
> > - the AGI process just runs quickly then quits, returning
> control back to
> > the dialplan. I had incorrectly assumed you were doing the
> same.
> >
> > Cheers,
> > Kingsley.
> >
> > On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote:
> > > Kingsley,
> > >
> > > Thanks for the reply, but I am looking to continue within
> the same AGI
> > > process and I believe that method would require starting a
> new AGI.
> > >
> > >
> > > On 21 November 2011 22:22, Kingsley Tart
> 
> > > wrote:
> > > We do that with the "F" option in Dial().
> > >
> > >
> > > >From http://www.voip-info.org/wiki/view/Asterisk
> +cmd+Dial :
> > >
> > > F(context^exten^pri): When the caller hangs up,
> transfer the
> > > called
> > > party to the specified context and extension and
> continue
> > > execution.
> > >
> > >
> > > Cheers,
> > > Kingsley.
> > >
> > > On Mon, 2011-11-21 at 17:38 +1100, David
> Cunningham wrote:
> > > > Hello,
> > > >
> > > > We would like to continue a Perl AGI after a
> Dial() it has
> > > done
> > > > completes following caller hangup. We would like
> to do this
> > > in the
> > > > same AGI, and not using a new AGI from the 'h'
> extension. It
> > > works
> > > > fine when the called party hangs up and the 'g'
> option is
> > > used, but
> > > > not for caller hangup.
> > > >
> > > > Is this possible?
> > > >
> > > > If not a confirmation that this is the case
> would be very
> > > helpful.
> > > >
> > > > Thanks for any advice!
> > > >
> > > > --
> > > > David Cunningham, Voisonics
> > > > http://voisonics.com/
> > > > US toll-free: +1 888 842 2720
> > > > UK: +44 (0) 20 3298 1642
> > > > Australia: +61 (0) 2 8063 9019
> > > >
> > >
> > > > --
> > > >
> > >
> >
> _
> > >   

Re: [asterisk-users] AEX410P drops DTMF digits

[Tzafrir Cohen]

On Mon, Nov 21, 2011 at 04:17:59PM -0600, Danny Nicholas wrote:
> Hello again list,
> 
>   I'm running a 1.4.42 install on SUSE with an
> AEX410P card.  The DAHDI release is 2.4.0 because the machine won't properly
> install 2.5

Care to elaborate? Build issue? Install issue?

What version of SUSE?

> and also won't install Asterisk 10.0 because I can't get a good
> SQLite3 library to install.  

Package from your repository?

> Whenever I enter DTMF very quickly or very
> slowly, app_read des on me.  Has anyone experienced similar joy using DAHDI
> drivers?  I've piddled with channel.c and app_read.c trying to tame this
> beast but it seems to have the better of me.

What port is it? FXS or FXO?

If you want to record the audio at the DAHDI level:

  dahdi_monitor 1 -r dtmf.wav

(and press ctrl-c to stop recording)

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

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Re: [asterisk-users] Deleting AstDB family at start

[Tzafrir Cohen]

On Sun, Nov 20, 2011 at 02:25:13PM -0500, Matt Hamilton wrote:
> 
> Is it possible to delete the keys belonging to a family in AstDB at
> Asterisk startup? I would like to repopulate it from another source
> each time Asterisk is restarted. 

Why not use global variables instead?

-- 
   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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users