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asterisk network connections

4520 is for DUNDI. Obviously your install uses H323 in some flavor.
Mgcp-callagent is for jitterbuffering? And sieve and complex-main I have no
clue (perhaps H323 tag-alongs)

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 17, 2012 1:41 PM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: [asterisk-users] asterisk network connections

Hello,

with the command lsof -i I notice the following network connections of the
asterisk proces :

asterisk 23006 root 12u IPv4 1088961 UDP
*:mgcp-callagent
asterisk 23006 root 13u IPv4 1088964 TCP *:sieve
(LISTEN)
asterisk 23006 root 16u IPv4 1088966 UDP *:iax
asterisk 23006 root 17u IPv4 1088967 UDP
*:commplex-main
asterisk 23006 root 18u IPv4 1088968 UDP *:4520
asterisk 23006 root 20u IPv4 1088969 TCP
*:h323hostcall (LISTEN)
asterisk 23006 root 21u IPv4 1088970 UDP *:sip

Now, iax and sip I find normal…

What is all the rest ??

Headset Options

Hi, Jabra headsets work fine with Polycom.

Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] På vegne af Blake Burgess
Sendt: 7. februar 2012 05:01
Til: asterisk-users@lists.digium.com
Emne: [asterisk-users] Headset Options

Hey,

I’ve heard recently from quite a few customers that there’s cordless handsets around which don’t require a lifter.

Is anyone aware of any of these which will work with the cisco 69xx’s, 79xx’s or any of the current polycom range?

-Blake

dial a queue

No :(

2012/2/6, Danny Nicholas :
> Queue(8888)?
>
> —–Original Message—–
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alejandro
> Cabrera Obed
> Sent: Monday, February 06, 2012 1:40 PM
> To: Asterisk Users Mailing List – Non-Commercial Discussion
> Subject: Re: [asterisk-users] Custom extension: dial a queue
>
> No, Local/queue/8888 don’t work at all :(
>
> 2012/2/6, Danny Nicholas
:
>> Local/queue/8888?
>>
>> —–Original Message—–
>> From: asterisk-users-bounces@lists.digium.com
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
>> Alejandro Cabrera Obed
>> Sent: Monday, February 06, 2012 1:19 PM
>> To: Asterisk Users Mailing List – Non-Commercial Discussion
>> Subject: [asterisk-users] Custom extension: dial a queue
>>
>> Dear, I need to create a custom device extension in order to dial a
>> local queue.
>>
>> Suppose my queue number is 8888, how can fill the Dial field from the
>> custom extension ???
>>
>> Because if I put just 8888 or Local/8888, I don’t succeed.
>>
>> Thanks a lot
>>
>> –
>> _____________________________________________________________________
>> — 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
>>
>>
>> –
>> _____________________________________________________________________
>> — 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
>>
>
>
> –
> Alejandro Cabrera Obed
> aco1967@gmail.com
> www.alejandrocabrera.com.ar
>
> –
> _____________________________________________________________________
> — 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
>
>
> –
> _____________________________________________________________________
> — 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
>

Blind transfers being cancelled by asterisk & hanging up on remote caller

Are we the only 2 people on this list experiencing this issue? (surprised)

Anyone else have any insights?

My hunch is that this is likely some type of FreePBX issue with how it generates the [from-internal-xfer] context.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Mortensen
Sent: Saturday, January 07, 2012 3:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller

Oh crap. I just reread the previous post & realized I’m not alone. Hallelujah! I’ll post back more info soon.

-
Doug Mortensen
Sent via DroidX2 on Verizon Wireless™

asterisk -> AGI (perl) -> sqlplus(oracle)

Yes, I already declared ‘use lib
“/home/asterisk/lib/lib64/perl5/5.8.8/x86_64-linux-thread-multi/”;’ in my
AGI. When I execute the script as a user Asterisk, i.e. perl -wc test.pl in
return I’m getting OK and no error messages and script is running fine when
I try to run in shell.

Even though I already declared the environmental variables in
.bash_profile. At the end I tired every method but still stuck in this
problem.

Date: Thu, 5 Jan 2012 14:07:59 -0800
> From: “Ron Bergin”
> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
> (oracle)
> To: “Asterisk Users Mailing List – Non-Commercial Discussion”
>
> Message-ID:
> <69ecf8ff3230bc206837478422f97aad.squirrel@webmail.i.frys.com>
> Content-Type: text/plain;charset=iso-8859-1
>
> Ahmed Munir wrote:
> > Hi,
> >
> > I installed the modules in asterisk user home directory with read and
> > excitable permissions for asterisk but still my AGI not working.
>
> IMO, it would have been better to install it in it’s normal location.
>
> Is your script using the warnings and strict pragmas?
>
> What error message do you receive when running the script from the command
> line?
>
> Did you add the proper “use lib ‘….'” statement to add the install
> directory to the @INC array?
>
> Ron Bergin
>
> >
> > Please provide me other advise to resolve this issue.
> >
> >
> >> Date: Wed, 4 Jan 2012 11:30:34 -0600
> >> From: “Danny Nicholas”
> >> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
> >> (oracle)
> >> To: “‘Asterisk Users Mailing List – Non-Commercial Discussion'”
> >>
> >> Message-ID: <00ca01cccb06$911e8300$b35b8900$@debsinc.com>
> >> Content-Type: text/plain; charset=”us-ascii”
> >>
> >> The module probably isn’t readable/executeable from Asterisk
> >>
> >>
> >>
> >> From: asterisk-users-bounces@lists.digium.com
> >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ahmed
> >> Munir
> >> Sent: Wednesday, January 04, 2012 10:45 AM
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)
> >>
> >>
> >>
> >> Hi all,
> >>
> >> I’m trying to run an AGI in PERL which uses the module DBD-Oracle.
> >> Currently
> >> my AGI is working fine in my two servers but not in my other four
> >> servers.
> >> When I tried execute an AGI (as a user asterisk) in command line it
> >> works
> >> fine (even I also declare environmental variables in user profile and in
> >> my
> >> AGI), but when I tried to call my AGI (perl) in dial plan, it don’t get
> >> executed.
> >>
> >> Please advise me to resolve this issue.
> >>
> >> –
> >> Regards,
> >>
> >> Ahmed Munir Chohan
> >>
> >>
>
>

registration not authorized – stale nonce

Asterisk destroys SIP dialogs in 32 secs, so increased the UAC registry expiration to 300 secs just in case, but that didn’t help either.

From: mistral9999@hotmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 1 Jan 2012 18:13:07 -0500
Subject: [asterisk-users] registration not authorized – stale nonce

I have a very basic setup where a UAC registers with Asterisk 1.8.7.2 – both on the same subnet, no nat.

The following is the flow of messages:
1. UAC sends the registration request
2. Asterisk responds with 401 Unauthorized with a new nonce
3. UAC sends a new digest with the nonce received from Asterisk
4. Asterisk authorizes UAC and sends OK

This works as expected when there is no load on the Asterisk server.

For testing asterisk under load, I use Sipp (runnig at a server on the same subnet) with a very basic scenario (call is placed, moh played, sipp hangs up after 20 secs). Max calls on the system at one time is 50.

My problem happens when the above UAC tries to register while the Sipp test is running.

Steps 1 and 2 above happen as expected, Asterisk sends a new nonce to the UAC, but at step 3, UAC sends the old digest (old nonce) back to Asterisk. Asterisk doesn’t authorize the UAC with “Correct auth, but based on stale nonce” warning.

UAC registration expiry is 60 seconds. As I mentioned, if the test is not running, everything seems to be OK. If I adjust the max calls to 10 in the test and reduce the load, the registrations go thru.

Any ideas?

Thanks,
Matt