[asterisk-users] Asterisk and H.323

[Aaron Anderson]

Thanks for the response. I have been able to now receive calls over 
h.323 using sjphone through the built in ooh323 channel driver.  It 
seems to work ok for a bit but then asterisk seems to stop accepting 
connections and the server needs to be rebooted.

On a slightly different note, I have 3 sip trunks set up in asterisk as 
Trunk1, Trunk2, and Trunk3 however asterisk only ever uses trunk3.  This 
makes it impossible to receive or make more than one call at a time.

An odd situation, but something I need to resolve.  Also, when receiving 
calls from the stateside client, it seems that the call picks up, and 
then hangs up after about 3 seconds of connectivity:

  -- Executing GotoIf("OOH323/2128506-5b99", "0?16") in new stack
  -- Executing Dial("OOH323/2128506-5b99", "SIP/trunk3/09068601194") in 
new stack
 -- Called trunk3/09068601194
 -- SIP/trunk3-7e99 is making progress passing it to OOH323/2128506-5b99
 -- SIP/trunk3-7e99 is ringing
  -- SIP/trunk3-7e99 is making progress passing it to OOH323/2128506-5b99
 -- SIP/trunk3-7e99 answered OOH323/2128506-5b99
  -- Attempting native bridge of OOH323/2128506-5b99 and SIP/trunk3-7e99
= Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 
'OOH323/2128506-5b99' in macro 'dialout-trunk'
== Spawn extension (from-internal, 909068601194, 1) exited non-zero on 
'OOH323/2128506-5b99'
  -- Executing Macro("OOH323/2128506-5b99", "hangupcall") in new stack
  -- Executing ResetCDR("OOH323/2128506-5b99", "w") in new stack
 -- Executing NoCDR("OOH323/2128506-5b99", "") in new stack
  -- Executing Wait("OOH323/2128506-5b99", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 
'OOH323/2128506-5b99' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 
'OOH323/2128506-5b99'

Any ideas?  It works fine from a soft phone.  Could it be something to 
do with the codec they are using?

Thanks in advance
Aaron

Cesc wrote:
> On 7/24/06, Aaron Anderson <webmaster at psphacks.net> wrote:
>>
>>  I have been messing with both all day.  I think what might be 
>> tripping me
>> up is the extensions.conf.
>>
> i do think so too :)
>
>
>>  I was able to receive an incoming connection from the client, but 
>> all the
>> system returned was a busy signal.  This call was to a known good 
>> number (my
>> phone) so I'm not sure what's wrong.
>>
> if you got the incoming connection but then got a busy tone, it is
> because the "peer info" in your asterisk conf is not good ...
> try the "standard" demo setup in asterisk ... there you can call
> extension 600 (i think) and you get an echo test ...
> later, extend this to call whoever you want ... but you'll need to
> enter the "location" info somehow (i never did it ... ). I used
> sjphone for testing ... try it ... you can add an extension to dial
> directly to the IP address of the sjphone. So, you dial the MMM
> extension on your phone, get to asterisk and then asterisk dials the
> IP address of sjphone ... nice and easy for testing.
>
>>  Will gnugk do a "translation" from h.323 to sip so I don't have to 
>> make any
>> major modifications?  Is there an example gnugk.ini and h323.conf 
>> file I can
>> look at to get this all running?
>
> gnugk has no clue of sip.
> gnugk is a gatekeeper for h323 ... you can do stuff with the dialled
> numbers ... forwarding the call to various [asterisk] h323 gateways
> and the like ... but the conversion of h323 to sip is done in
> asterisk.
>
>
> Cesc
>
>
>>
>>  Thank you in advance
>>  Aaron
>>
>>
>>  harrygaillac-sip at yahoo.fr wrote:
>>
>>  Hello,
>>
>> Try both chan_oh323 and gnugk .
>>
>> Harry
>> --- Aaron Anderson <webmaster at psphacks.net> a écrit :
>>
>>
>>
>>  I have been scouring the net the last couple of days
>> looking for some
>> kind of tutorial or walkthrough on setting up a
>> h.323 channel in asterisk.
>>
>> What I need to do is basically this:
>>
>> I have a client who wants to be able to connect to
>> me via h.323 and make
>> a local phone call (local to me, he is in a
>> different country). The
>> call is an automated process and no callee
>> interaction is required. My
>> client simply wants to be able to call a user and
>> give them a
>> verification number and then hang up. He's using
>> some in-house software
>> so unfortunately, h.323 is his only option.
>>
>> Can someone point me to a doc or perhaps give me a
>> simple breakdown of
>> what I need to add to asterisk in order to be able
>> to do this? I am on
>> a tight deadline and my searches have not revealed
>> the information I am
>> looking for. I have built chan_h323 and it is
>> loaded but I'm not sure
>> how to set it up beyond that.
>>
>> Any help would be much appreciated.
>>
>> Thank you
>> Aaron
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com
>> -- 
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>
>>
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> ___________________________________________________________________________ 
>>
>> Découvrez un nouveau moyen de poser toutes vos questions quelque soit le
>> sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos
>> opinions et vos expériences. http://fr.answers.yahoo.com
>> _______________________________________________ --Bandwidth
>> and Colocation provided by Easynews.com -- asterisk-users mailing 
>> list To
>> UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> -----------------------------------------------------------------
>> This message was scanned by Sunny-Net SpamVirus Detector Gateway If 
>> this is
>> not spam but was Maked with SPAM in the subject line please forward this
>> e-mail to ham at bay1.sunny-net.ne.jp. if this is spam but was not 
>> marked spam
>> please forward the e-mail to spam at bay1.sunny-net.ne.jp. Thankyou
>> ------------------------------------------------------------
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> -----------------------------------------------------------------
> This message was scanned by Sunny-Net SpamVirus Detector Gateway
> If this is not spam but was Maked with  SPAM in the subject line 
> please forward this e-mail to ham at bay1.sunny-net.ne.jp.
> if this is spam but was not marked spam please forward the
> e-mail to spam at bay1.sunny-net.ne.jp.  Thankyou
> ------------------------------------------------------------
>
>