Re: [Asterisk-Users] oh-323 / Cisco AS5300 problem

Matias G. Fri, 03 Jun 2005 12:13:58 -0700

Leandro,

I don�t have the licenses on porpuose, I shouldn't need'em cause the phone I'm using has only g729 enabled, the oh323 has only g729a enabled and the Cisco (over which I have no control at all) should be able to manage g729... furthermore reading the trace from the oh323 (debug level 5) I never found any reference to ulaw... just tenths of references to rtp audio being sent in g729 and then suddenly a:

51:45.239 ClearCallT...d:b092b248 H323 Clearing connection ip$localhost/32126 reason=EndedByLocalUser

on the second case, the phone is (99% sure) not busy (it's my desk phone) and if I call using g729 I get it to ring the phone and when I pick up I get the error, if I call using ulaw or alaw it doesn�t even ring I get the busy msg and that's it... but what is strange is that it can't be wrong rules in the 53xx cause when using g729 everything works fine till I answer the call... so the dialing logic is ok, of course I dial the same number when in g729 and g711...

any more clues?

thanks again.

----- Original Message -----

From: Leandro Tenorio

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Sent: Friday, June 03, 2005 3:32 PM

Subject: RE: [Asterisk-Users] oh-323 / Cisco AS5300 problem

The first error is probably because you don�t have licenses for 729 and you are trancoding the audio.

The second is well dialed and you get from the 5300 a busy message, the reason could be user busy (as the message saids), wrong dial peer config, wrong dialing rules in 53xx.

LTenorio

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matias G.
Sent: Friday, June 03, 2005 1:02 PM
To: [email protected]
Subject: [Asterisk-Users] oh-323 / Cisco AS5300 problem

Hi i'm trying to connect to the PSTN in the following way

sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN

I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3

Asterisk-Oh323 0.7.2 pre1

Open H323 v1.13.5

pwlib v1.6.6

and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and has a private ip but there's a stun server working and the connection works fine.

If i try to dial out from the sip ATA through the GK (gnugk) configuring g729 in the sip account and the oh323.conf I get an error about trying to use ulaw:

   -- Executing Dial("SIP/D0103-3a43", "OH323/80152392522|50")
    -- H.323 call to 80152392522 with codec(s) g729
    -- Called 80152392522
    -- OH323/80152392522-ef38 is ringing
Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find
a path from ulaw to g729
Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find
a path from ulaw to g729
OH323/80152392522-ef38: Format changed to ulaw (native g729).
Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find
a path from g729 to ulaw
Jun 3 10:09:05 WARNING[27744]: app_dial.c:583 wait_for_answer: Unable
to forward voice
    -- Hungup 'OH323/80152392522-ef38'
== No one is available to answer at this time (1:0/0/0)
    -- H.323 call 'ip$localhost/32125' cleared, reason 1 (Cleared by
local user)

but if i try g711 ulaw on both the ata and the oh323 i get the following error

   -- Executing Dial("SIP/D0103-d16b", "OH323/00541152392522|50")
    -- H.323 call to 00541152392522 with codec(s) ulaw
    -- Called 00541152392522
    -- H.323 call 'ip$localhost/26583' cleared, reason 24 (Call ended
with Q.931 cause)
    -- OH323/00541152392522-d872 is busy
    -- Hungup 'OH323/00541152392522-d872'
== Everyone is busy/congested at this time (1:1/0/0)

I found a message about this error from January but there is no follow up

any clues?

thanks a lot in advance.

Matias

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] oh-323 / Cisco AS5300 problem

Reply via email to