gmane.comp.telephony.pbx.asterisk.user

Re: hangup not detected?

Justin Killen — 2012-05-22T15:53:05

Okay, the next time it gets in this state I'll gather that information.

Justin Killen
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From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, May 21, 2012 1:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup not detected?

On Fri, May 18, 2012 at 12:00 PM, Justin Killen <jkillen< at >allamericanasphalt.com<mailto:jkillen< at >allamericanasphalt.com>> wrote:
I have and automated call-in dispatch system where hundreds of people call in daily for 2-3 minutes each.  The extension is set up to get their information, then text-to-speech the dispatch information (via odbc).  It then loops 5 times then ends the call.  These calls are being handled by an 8 port analog digium card.

Sometimes though, I see calls via 'core show channel dahdi/1-1' that have a time of > 16 hours.  I'm not sure if this is a result of dahdi missing the hangup, ODBC timing out, or TTS fai

Re: Asterisk AMI SIP channel detect phone

JIMMY GATHAGE — 2012-05-22T14:58:14

Thanks Kevin and Yaroslav,

Sorry was out of town.

Sorry I forgot to mention that Iam using an VOIP GSM gateway to connect to PSTN.

Kevin,

I am have decided to use Sangoma CPA. Do you know of any other options
that are easier to integrate with?

Yaroslav,
Yes I had set the header ASYNC to yes.

Thanks for the help.

Thanks.

regards,


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Re: SET SIP_CODEC and Video issues

Kevin P. Fleming — 2012-05-22T14:00:08

Unfortunately you can't do what you want using SIP_CODEC; if you set 
that variable, the formats (both audio and video) allowed on the channel 
are reset to whatever you specify, and that variable can only hold one 
format name.

It seems odd though that you want to change audio codecs based on 
bandwidth tests, but still allow video. The video stream is going to 
consume vastly more bandwidth than the audio stream.

Re: sip show peers

Mitul Limbani — 2012-05-22T12:53:54

yeah, put qualify=2000 to ensure that you shall get the latency perfectly.

Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul< at >enterux.in
DID: +91-22-61447605
Cell: +91-9820332422




On Tue, May 22, 2012 at 5:50 PM, Faisal Hanif <faisal< at >vopium.com> wrote:

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Re: gr-303

Kevin P. Fleming — 2012-05-22T12:49:05


Yes, chan_dahdi has support for connecting GR-303 channel banks to 
Asterisk via T1 spans. It's pretty rare that someone tries to use it, 
though, and as you say, there is little documentation.

There is no support in chan_dahdi to make Asterisk behave *as* a GR-303 
channel bank.

Re: sip show peers

Faisal Hanif — 2012-05-22T12:20:54

If I understand correct you need to increase qualify value.

Regards,

Faisal Hanif
-----Original Message-----
From: asterisk-users-bounces< at >lists.digium.com
[mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, May 22, 2012 5:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip show peers

I have a process that runs on a server and does a simple 'asterisk -rx "sup
show peers' > /tmp/peers"
and then looks for any "(Unspecified)" items and reports them as having lost
connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while 1 of my boxes reports "(Unspecified)". I am trying to
find out why.

How can I make the remote boxes have a shorter heart beat to checking more
frequently with the server so as not to go "(Unspecified)". By the time I
log in and check its already back connected again.

Any other thoughts?

Thanks,

Jerry

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sip show peers

Jerry Geis — 2012-05-22T12:02:18

I have a process that runs on a server and does a simple 'asterisk -rx 
"sup show peers' > /tmp/peers"
and then looks for any "(Unspecified)" items and reports them as having 
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also 
running 1.4.43.
Once in a great while 1 of my boxes reports "(Unspecified)". I am trying 
to find out why.

How can I make the remote boxes have a shorter heart beat to checking 
more frequently
with the server so as not to go "(Unspecified)". By the time I log in 
and check its already
back connected again.

Any other thoughts?

Thanks,

Jerry

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gr-303

Don Dawson — 2012-05-22T04:43:01

Does asterisk support gr-303?
Seems to be undocumented if so.


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Re: Asterisk and the media path

Jared Geiger — 2012-05-21T22:08:16

A2billing usually stays in the media path due to the dialstring
parameters that it uses to cut a call off when the balance would reach
$0. To get Asterisk to step out of the media path, I had to change
dialcommand_param_sipiax_friend and dialcommand_param to |60|S(14400)
which lets all calls go to 14400 seconds. The default uses the L
parameter. You need to use the S parameter instead. However the S
parameter doesn't like very large numbers in Asterisk 1.4 so I've just
hard set mine.

~Jared

On Mon, May 21, 2012 at 5:18 PM, Kevin P. Fleming <kpfleming< at >digium.com> wrote:

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Re: Asterisk and the media path

Kevin P. Fleming — 2012-05-21T21:18:38

Yes, it has to be set on both peers involved in the bridged call.

Re: Asterisk and the media path

David Wessell — 2012-05-21T20:45:31

More specific on sip.conf

In sip.conf I have a trunk specified for the SIP provider, and a trunk
specified for the PBX itself.

Do  I need to specify directmedia=yes on both sides?

Thanks for your help. I'll be testing it in a few minutes.

Thanks
David

On Mon, May 21, 2012 at 2:18 PM, Kevin P. Fleming <kpfleming< at >digium.com> wrote:

Re: hangup not detected?

Warren Selby — 2012-05-21T20:21:32

On Fri, May 18, 2012 at 12:00 PM, Justin Killen <
jkillen< at >allamericanasphalt.com> wrote:

Can you post the CLI output of a call that gets "hung"?  I'd like to see
where it's hanging on.

Also, as a work-around to attempt to solve the symptom and not the
underlying issue, you could maybe setup a cron job that runs once every ten
minutes that checks for stale calls using AMI, and then hangs up any calls
up that are over 10 minutes long?  Using the AMI Hangup command?

Wrong SIP to SIP SIP Cause mapping

alexandre Moutot — 2012-05-21T19:09:18

Hello,


I'm using asterisk v1.8 with a standard scenario, A Sip call from A to B through asterisk :

A --SIP--> ASTERISK --SIP--> B

The asterisk extension is :
exten => _X.,1,Dial(SIP/B/${EXTEN},600)
exten => _X.,n,Hangup()

When B send a 404 back to the asterisk, the asterisk sends a 503 to A. It is the same with 403 and some others erroc code.
I think it should send back to A the same error code.

I have done tests with some versions:
- 1.8.11.x : wrong sip cause mapping
- 1.8.13.0rc1 : wrong sip cause mapping
- 1.10.3 : wrong sip cause mapping
- 1.8.8.0 : works good

Do i do something wrong or should i open a bug ?


MOUTOT A.

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Re: Asterisk and the media path

Kevin P. Fleming — 2012-05-21T18:18:57

You say 'in sip.conf' multiple times, but that's far too vague to mean 
anything. sip.conf is a configuration file used to define SIP peers (and 
users), each with their own settings. There isn't any place you can set 
'directmedia' on and have it affect all your defined peers. Each peer 
that should have directmedia enabled must have it set.


No.


It should happen quite quickly after the call is answered.

Re: Asterisk and the media path

David Wessell — 2012-05-21T17:54:02

So I need directmedia set in sip.conf on the LCR trunk.

1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
2) Do I need anything on the pbx side that we are hoping to transfer media to?
3) How long into the call before the media is transferred over?

Thanks
David

On Mon, May 21, 2012 at 1:18 PM, Kevin P. Fleming <kpfleming< at >digium.com> wrote:

Re: Realtime peers and trunks coming from the sameIP

Ricardo Carvalho — 2012-05-21T17:23:50

Thanks Sammy, I think I'll stop using SIP realtime.

Regards,
Ricardo.



On Mon, May 21, 2012 at 5:14 AM, SamyGo <govoiper< at >gmail.com> wrote:

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Re: Asterisk and the media path

Kevin P. Fleming — 2012-05-21T17:18:50

Sure, but you used the *old* name for the option on the system running a 
*newer* version of Asterisk. That's why I was confused, I suspected you 
might have thought that 'directmedia' and 'canreinvite' were somehow 
different. Since both of your systems are 1.6.2.x or later, you can use 
'directmedia' on all of them.

Re: Asterisk and the media path

David Wessell — 2012-05-21T16:46:41

Hi Kevin,

Thank you. Here's the requested information.

1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
3) I did directmedia on the trunk and canreinvite on the pbx since
they were different versions.

Thansk
David

On Mon, May 21, 2012 at 11:22 AM, Kevin P. Fleming <kpfleming< at >digium.com> wrote:

Re: Recommendations on FXS Bank

Eric Wieling — 2012-05-21T15:58:58

We use Adtran Total Access boxes to convert PSTN to SIP.    Xorcom has some PSTN/SIP USB boxes which people seem to love.

-----Original Message-----
From: asterisk-users-bounces< at >lists.digium.com [mailto:asterisk-users-bounces< at >lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, May 21, 2012 11:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Recommendations on FXS Bank

Gradstream has a 24 port solution: GXW4024

Sangoma has a new solution for up to 50 ports: Vega5000

On Mon, 2012-05-21 at 06:04 +0000, Klaverstyn, David C wrote:

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Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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Re: Recommendations on FXS Bank

Carlos Chavez — 2012-05-21T15:54:11

Gradstream has a 24 port solution: GXW4024

Sangoma has a new solution for up to 50 ports: Vega5000

On Mon, 2012-05-21 at 06:04 +0000, Klaverstyn, David C wrote:

Re: asterisk voicemail

Bogdan — 2012-05-21T15:22:36

Hello,

That did it!!! I still have a "thank you" at the end but I assume that 
is because of the default language set to English. Thank you very much 
for the help!

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