Hi list , I am connecting one avaya with asterisk by h323 and when I
call to avaya becomes disconnected, this is my debug
ippbx*CLI> h323 set debug on
H.323 Debugging Enabled
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [1083 at mific:1] Dial("SIP/4097-00000002",
"H323/1083 at 172.16.8.5:1720,40") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Making call to 1083 at 172.16.8.5:1720 without gatekeeper.
Using 172.16.8.56 for outbound call
== New H.323 Connection created.
-- root is calling host 1083 at 172.16.8.5:1720
-- Call token is ip$localhost/19287
-- Call reference is 19287
-- DTMF Payload is 0x4235b48
-- Called 1083 at 172.16.8.5:1720
Setting capabilities to 0xc (ulaw|alaw)
Capabilities in preference order is (ulaw|alaw)
DTMF mode is 8
Allowed Codecs for ip$localhost/19287 (ip$172.16.8.56:39935):
Table:
G.711-uLaw-64k <1>
G.711-ALaw-64k <2>
UserInput/hookflash <3>
UserInput/basicString <4>
Set:
0:
0:
G.711-uLaw-64k <1>
G.711-ALaw-64k <2>
1:
UserInput/hookflash <3>
2:
UserInput/basicString <4>
-- Sending SETUP message
-- Received RELEASE COMPLETE message...
-- ClearCall: Request to clear call with token
ip$localhost/19287, cause EndedByRemoteBusy
-- Sending RELEASE COMPLETE
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
ExternalRTPChannel Destroyed
-- ClearCall: Request to clear call with token
ip$localhost/19287, cause EndedByTransportFail
-- 1083 was busy
== H.323 Connection deleted.
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [1083 at mific:2] Hangup("SIP/4097-00000002",
"") in new stack
== Spawn extension (mific, 1083, 2) exited non-zero on
'SIP/4097-00000002'
I have perfectly compiled h323 in asterisk
core show channeltypes
Type Description Devicestate
Indications Transfer
---------- ----------- -----------
----------- --------
Local Local Proxy Channel Driver yes yes
no
Bridge Bridge Interaction Channel no no
no
H323 The NuFone Network's Open H.323 Channel no yes
no
Console OSS Console Channel Driver no yes
no
USTM UNISTIM Channel Driver no yes
no
Phone Standard Linux Telephony API Driver no yes
no
any idea?
regardss
--
rickygm
http://gnuforever.homelinux.com
Do you have any network devices or VPN tunnels in between the Asterisk and Avaya? The reason I am asking it looks like a potential networking issue. Has this setup ever worked before? -Vladimir On 7/27/2011 1:32 PM, troxlinux wrote:> Hi list , I am connecting one avaya with asterisk by h323 and when I > call to avaya becomes disconnected, this is my debug > > > ippbx*CLI> h323 set debug on > H.323 Debugging Enabled > == Using SIP RTP CoS mark 5 > == Using SIP VRTP CoS mark 6 > -- Executing [1083 at mific:1] Dial("SIP/4097-00000002", > "H323/1083 at 172.16.8.5:1720,40") in new stack > -- Requested transfer capability: 0x00 - SPEECH > -- Making call to 1083 at 172.16.8.5:1720 without gatekeeper. > Using 172.16.8.56 for outbound call > == New H.323 Connection created. > -- root is calling host 1083 at 172.16.8.5:1720 > -- Call token is ip$localhost/19287 > -- Call reference is 19287 > -- DTMF Payload is 0x4235b48 > -- Called 1083 at 172.16.8.5:1720 > Setting capabilities to 0xc (ulaw|alaw) > Capabilities in preference order is (ulaw|alaw) > DTMF mode is 8 > Allowed Codecs for ip$localhost/19287 (ip$172.16.8.56:39935): > Table: > G.711-uLaw-64k <1> > G.711-ALaw-64k <2> > UserInput/hookflash <3> > UserInput/basicString <4> > Set: > 0: > 0: > G.711-uLaw-64k <1> > G.711-ALaw-64k <2> > 1: > UserInput/hookflash <3> > 2: > UserInput/basicString <4> > > -- Sending SETUP message > -- Received RELEASE COMPLETE message... > -- ClearCall: Request to clear call with token > ip$localhost/19287, cause EndedByRemoteBusy > -- Sending RELEASE COMPLETE > ExternalRTPChannel Destroyed > ExternalRTPChannel Destroyed > ExternalRTPChannel Destroyed > ExternalRTPChannel Destroyed > -- ClearCall: Request to clear call with token > ip$localhost/19287, cause EndedByTransportFail > -- 1083 was busy > == H.323 Connection deleted. > == Everyone is busy/congested at this time (1:1/0/0) > -- Executing [1083 at mific:2] Hangup("SIP/4097-00000002", "") in new stack > == Spawn extension (mific, 1083, 2) exited non-zero on 'SIP/4097-00000002' > > > I have perfectly compiled h323 in asterisk > > core show channeltypes > Type Description Devicestate > Indications Transfer > ---------- ----------- ----------- > ----------- -------- > Local Local Proxy Channel Driver yes yes > no > Bridge Bridge Interaction Channel no no > no > H323 The NuFone Network's Open H.323 Channel no yes > no > Console OSS Console Channel Driver no yes > no > USTM UNISTIM Channel Driver no yes > no > Phone Standard Linux Telephony API Driver no yes > no > > > any idea? > > regardss > > > >
2011/7/27 Vladimir Mikhelson <vlad at mikhelson.com>:> Do you have any network devices or VPN tunnels in between the Asterisk > and Avaya? >Hi , the server does not have connections vpn I have and it in the same LAN that avaya> The reason I am asking it looks like a potential networking issue.ok, but I do ping to him to avaya perfectly, without lost of packages --- 172.16.8.5 ping statistics --- 16 packets transmitted, 16 received, 0% packet loss, time 15007ms rtt min/avg/max/mdev = 0.513/1.506/6.393/1.306 ms> > Has this setup ever worked before?I only have it between internal calls of asterisk and works, but h323 no :(> > -Vladimir >regardss -- rickygm http://gnuforever.homelinux.com